One document matched: draft-ietf-mmusic-rfc2326bis-38.xml
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<front>
<title abbrev="Real Time Streaming Protocol 2.0 (RTSP)">Real Time
Streaming Protocol 2.0 (RTSP)</title>
<author fullname="Henning Schulzrinne" initials="H." surname="Schulzrinne">
<organization>Columbia University</organization>
<address>
<postal>
<street>1214 Amsterdam Avenue</street>
<city>New York</city>
<region>NY</region>
<code>10027</code>
<country>USA</country>
</postal>
<email>schulzrinne@cs.columbia.edu</email>
</address>
</author>
<author fullname="Anup Rao" initials="A." surname="Rao">
<organization>Cisco</organization>
<address>
<postal>
<street/>
<city/>
<region/>
<code/>
<country>USA</country>
</postal>
<email>anrao@cisco.com</email>
</address>
</author>
<author fullname="Rob Lanphier" initials="R." surname="Lanphier">
<organization/>
<address>
<postal>
<street/>
<city>Seattle</city>
<region>WA</region>
<code/>
<country>USA</country>
</postal>
<email>robla@robla.net</email>
</address>
</author>
<author fullname="Magnus Westerlund" initials="M." surname="Westerlund">
<organization>Ericsson AB</organization>
<address>
<postal>
<street>Färögatan 6</street>
<city>STOCKHOLM</city>
<region/>
<code>SE-164 80</code>
<country>SWEDEN</country>
</postal>
<email>magnus.westerlund@ericsson.com</email>
</address>
</author>
<author fullname="Martin Stiemerling" initials="M."
surname="Stiemerling (Ed.)">
<organization abbrev="NEC">NEC Laboratories Europe, NEC Europe
Ltd.</organization>
<address>
<postal>
<street>Kurfuersten-Anlage 36</street>
<city>Heidelberg</city>
<code>69115</code>
<country>Germany</country>
</postal>
<phone>+49 (0) 6221 4342 113</phone>
<email>martin.stiemerling@neclab.eu</email>
<uri>http://ietf.stiemerling.org</uri>
</address>
</author>
<date year="2013"/>
<area>Real-time Applications and Infrastructure Area</area>
<workgroup>MMUSIC Working Group</workgroup>
<keyword>I-D</keyword>
<keyword>INTERNET-DRAFT</keyword>
<keyword>mmusic, RTSP, RTSP/2.0, real-time streaming protocol</keyword>
<abstract>
<t>This memorandum defines RTSP version 2.0 which obsoletes RTSP version
1.0 defined in RFC 2326.</t>
<t>The Real Time Streaming Protocol, or RTSP, is an application-level
protocol for setup and control of the delivery of data with real-time
properties. RTSP provides an extensible framework to enable controlled,
on-demand delivery of real-time data, such as audio and video. Sources
of data can include both live data feeds and stored clips. This protocol
is intended to control multiple data delivery sessions, provide a means
for choosing delivery channels such as UDP, multicast UDP and TCP, and
provide a means for choosing delivery mechanisms based upon RTP (RFC
3550).</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t>This memo defines version 2.0 of the Real Time Streaming Protocol
(RTSP 2.0). RTSP 2.0 is an application-level protocol for setup and
control over the delivery of data with real-time properties, typically
streaming media. Streaming media is, for instance, video on demand or
audio live streaming. Put simply, RTSP acts as a "network remote
control" for multimedia servers, similar to the remote control for a DVD
player.</t>
<t>The protocol operates between RTSP 2.0 clients and servers, but also
supports the usage of proxies placed between clients and servers.
Clients can request information about streaming media from servers by
asking for a description of the media or use media description provided
externally. The media delivery protocol is used to establish the media
streams described by the media description. Clients can then request to
play out the media, pause it, or stop it completely, as known from DVD
players remote control or media players. The requested media can consist
of multiple audio and video streams that are delivered as
time-synchronized streams from servers to clients.</t>
<t>RTSP 2.0 is a replacement of <xref target="RFC2326">RTSP 1.0</xref>
and obsoletes that specification. This protocol is based on RTSP 1.0 but
is not backwards compatible other than in the basic version negotiation
mechanism. The changes are documented in <xref target="sec_changes"/>.
There are many reasons why RTSP 2.0 can't be backwards compatible with
RTSP 1.0 but some of the main ones are:<list style="symbols">
<t>Most headers that needed to be extensible did not define the
allowed syntax, preventing safe deployment of extensions;</t>
<t>The changed behavior of the PLAY method when received in Play
state;</t>
<t>Changed behavior of the extensibility model and its
mechanism;</t>
<t>The change of syntax for some headers.</t>
</list></t>
<t>In summary, there are so many small details that changing version
became necessary to enable clarification and consistent behavior.</t>
<t>This document is structured as follows. It begins with an overview of
the protocol operations and its functions in an informal way. Then a set
of definitions of terms used and document conventions is introduced. It
is followed by the actual RTSP 2.0 core protocol specification. The
appendixes describe and define some functionalities that are not part of
the core RTSP specification, but which are still important to enable
some usages. Among them, the RTP usage is defined in <xref
target="sec_mediatran"/>, the Session Description Protocol (SDP) usage
with RTSP is defined in <xref target="sec_sdpusage"/>, and the
text/parameters file format <xref target="sec_text-parameters"/>, are
three normative specification appendixes. Others include a number of
informational parts discussing the changes, use cases, different
considerations or motivations.</t>
</section>
<section title="Protocol Overview">
<t>This section provides an informative overview of the different
mechanisms in the RTSP 2.0 protocol, to give the reader a high level
understanding before getting into all the different details. In case of
conflict with this description and the later sections, the later
sections take precedence. For more information about use cases
considered for RTSP see <xref target="sec_usecases"/>.</t>
<t>RTSP 2.0 is a bi-directional request and response protocol that first
establishes a context including content resources (the media) and then
controls the delivery of these content resources from the provider to
the consumer. RTSP has three fundamental parts: Session Establishment,
Media Delivery Control, and an extensibility model described below. The
protocol is based on some assumptions about existing functionality to
provide a complete solution for client controlled real-time media
delivery.</t>
<t>RTSP uses text-based messages, requests and responses, that may
contain a binary message body. An RTSP request starts with a method line
that identifies the method, the protocol and version and the resource to
act on. The resource is identified by a URI and the hostname part of the
URI is used by RTSP client to resolve the IPv4 or IPv6 address of the
RTSP server. Following the method line are a number of RTSP headers.
This part is ended by two consecutive carriage return line feed (CRLF)
character pairs. The message body if present follows the two CRLF and
the body's length is described by a message header. RTSP responses are
similar, but start with a response line with the protocol and version,
followed by a status code and a reason phrase. RTSP messages are sent
over a reliable transport protocol between the client and server. RTSP
2.0 requires clients and servers to implement TCP, and TLS over TCP, as
mandatory transports for RTSP messages.</t>
<section title="Presentation Description">
<t>RTSP exists to provide access to multi-media presentations and
content, but tries to be agnostic about the media type or the actual
media delivery protocol that is used. To enable a client to implement
a complete system, an RTSP-external mechanism for describing the
presentation and the delivery protocol(s) is used. RTSP assumes that
this description is either delivered completely out of band or as a
data object in the response to a client's request using the <xref
target="sec_DESCRIBE">DESCRIBE method</xref>.</t>
<t>Parameters that commonly have to be included in the Presentation
Description are the following:<list style="symbols">
<t>Number of media streams;</t>
<t>The resource identifier for each media stream/resource that is
to be controlled by RTSP;</t>
<t>The protocol that each media stream is to be delivered
over;</t>
<t>Transport protocol parameters that are not negotiated or vary
with each client;</t>
<t>Media encoding information enabling a client to correctly
decode the media upon reception;</t>
<t>An aggregate control resource identifier.</t>
</list></t>
<t>RTSP uses its own URI schemes ("rtsp" and "rtsps") to reference
media resources and aggregates under common control (See <xref
target="sec_url"/>).</t>
<t>This specification describes in <xref target="sec_sdpusage"/> how
one uses <xref target="RFC4566">SDP</xref> for Presentation
Description</t>
</section>
<section title="Session Establishment">
<t>The RTSP client can request the establishment of an RTSP session
after having used the presentation description to determine which
media streams are available, and also which media delivery protocol is
used and their particular resource identifiers. The RTSP session is a
common context between the client and the server that consists of one
or more media resources that are to be under common media delivery
control.</t>
<t>The client creates an RTSP session by sending a request using the
<xref target="sec_SETUP">SETUP method</xref> to the server. In the
SETUP request the client also includes all the transport parameters
necessary to enable the media delivery protocol to function in the
<xref target="sec_Transport">"Transport" header</xref>. This includes
parameters that are pre-established by the presentation description
but necessary for any middlebox to correctly handle the media delivery
protocols. The Transport header in a request may contain multiple
alternatives for media delivery in a prioritized list, which the
server can select from. These alternatives are typically based on
information in the presentation description.</t>
<t>The server determines if the media resource is available upon
receiving a SETUP request and if any of the transport parameter
specifications are acceptable. If that is successful, an RTSP session
context is created and the relevant parameters and state is stored. An
identifier is created for the RTSP session and included in the
response in the <xref target="sec_Session">Session header</xref>. The
SETUP response includes a Transport header that specifies which of the
alternatives has been selected and relevant parameters.</t>
<t>A SETUP request that references an existing RTSP session but
identifies a new media resource is a request to add that media
resource under common control with the already present media resources
in an aggregated session. A client can expect this to work for all
media resources under RTSP control within a multi-media content.
However, aggregating resources from different content are likely to be
refused by the server. The RTSP session as aggregate is referenced by
the aggregate control URI, even if the RTSP session only contains a
single media.</t>
<t>To avoid an extra round trip in the session establishment of
aggregated RTSP sessions, RTSP 2.0 supports pipelined requests; i.e.,
the client can send multiple requests back-to-back without waiting
first for the completion of any of them. The client uses a
client-selected identifier in the <xref
target="sec_Pipelined-Requests">Pipelined-Requests header</xref> to
instruct the server to bind multiple requests together as if they
included the session identifier.</t>
<t>The SETUP response also provides additional information about the
established sessions in a couple of different headers. The <xref
target="sec_Media-Properties">Media-Properties header</xref> includes
a number of properties that apply for the aggregate that is valuable
when doing media delivery control and configuring user interface. The
<xref target="sec_Accept-Ranges">Accept-Ranges header</xref> informs
the client about which range formats that the server supports with
these media resources. The <xref target="sec_Media-Range">Media-Range
header</xref> informs the client about the time range of the media
currently available.</t>
</section>
<section title="Media Delivery Control">
<t>After having established an RTSP session, the client can start
controlling the media delivery. The basic operations are Start by
using the <xref target="sec_PLAY">PLAY method</xref> and Halt by using
the <xref target="sec_PAUSE">PAUSE method</xref>. PLAY also allows for
choosing the starting media position from which the server should
deliver the media. The positioning is done by using the <xref
target="sec_Range">Range header</xref> that supports several different
time formats: <xref target="sec_npt">Normal Play Time (NPT)</xref>,
<xref target="sec_smpte"> Society of Motion Picture and Television
Engineers (SMPTE) Timestamps</xref> and <xref
target="sec_clock">absolute time</xref>. The Range header does further
allow the client to specify a position where delivery should end, thus
allowing a specific interval to be delivered.</t>
<t>The support for positioning/searching within a content depends on
the content's media properties. Content exists in a number of
different types, such as: on-demand, live, and live with simultaneous
recording. Even within these categories there are differences in how
the content is generated and distributed, which affect how it can be
accessed for playback. The properties applicable for the RTSP session
are provided by the server in the SETUP response using the <xref
target="sec_Media-Properties">Media-Properties header</xref>. These
are expressed using one or several independent attributes. A first
attribute is Random Access, which expresses if positioning can be
done, and with what granularity. Another aspect is whether the content
will change during the lifetime of the session. While on-demand
content will be provided in full from the beginning, a live stream
being recorded results in the length of the accessible content growing
as the session goes on. There also exists content that is dynamically
built by another protocol than RTSP and thus also changes in steps
during the session, but maybe not continuously. Furthermore, when
content is recorded, there are cases where not the complete content is
maintained, but, for example, only the last hour. All these properties
result in the need for mechanisms that will be discussed below.</t>
<t>When the client accesses on-demand content that allows random
access, the client can issue the PLAY request for any point in the
content between the start and the end. The server will deliver media
from the closest random access point prior to the requested point and
indicate that in its PLAY response. If the client issues a PAUSE, the
delivery will be halted and the point at which the server stopped will
be reported back in the response. The client can later resume by
sending a PLAY request without a range header. When the server is
about to complete the PLAY request by delivering the end of the
content or the requested range, the server will send a <xref
target="sec_PLAY_NOTIFY">PLAY_NOTIFY request</xref> indicating
this.</t>
<t>When playing live content with no extra functions, such as
recording, the client will receive the live media from the server
after having sent a PLAY request. Seeking in such content is not
possible as the server does not store it, but only forwards it from
the source of the session. Thus delivery continues until the client
sends a PAUSE request, tears down the session, or the content
ends.</t>
<t>For live sessions that are being recorded the client will need to
keep track of how the recording progresses. Upon session establishment
the client will learn the current duration of the recording from the
Media-Range header. As the recording is ongoing the content grows in
direct relation to the passed time. Therefore, each server's response
to a PLAY request will contain the current Media-Range header. The
server should also regularly send approximately every 5 minutes the
current media range in a PLAY_NOTIFY request (<xref
target="sec_Media-Properties-Update-Reason"/>). If the live
transmission ends, the server must send a PLAY_NOTIFY request with the
updated Media-Properties indicating that the content stopped being a
recorded live session and instead became on-demand content; the
request also contains the final media range. While the live delivery
continues the client can request to play the current live point by
using the NPT timescale symbol "now", or it can request a specific
point in the available content by an explicit range request for that
point. If the requested point is outside of the available interval the
server will adjust the position to the closest available point, i.e.,
either at the beginning or the end.</t>
<t>A special case of recording is that where the recording is not
retained longer than a specific time period, thus as the live delivery
continues the client can access any media within a moving window that
covers, for example, "now" to "now" minus 1 hour. A client that pauses
on a specific point within the content may not be able to retrieve the
content anymore. If the client waits too long before resuming the
pause point, the content may no longer be available. In this case the
pause point will be adjusted to the closest point in the available
media.</t>
</section>
<section title="Session Parameter Manipulations">
<t>A session may have additional state or functionality that affects
how the server or client treats the session, content, how it
functions, or feedback on how well the session works. Such extensions
are not defined in this specification, but may be done in various
extensions. RTSP has two methods for retrieving and setting parameter
values on either the client or the server: <xref
target="sec_GET_PARAMETER">GET_PARAMETER</xref> and <xref
target="sec_SET_PARAMETER">SET_PARAMETER</xref>. These methods carry
the parameters in a message body of the appropriate format. One can
also use headers to query state with the GET_PARAMETER method. As an
example, clients needing to know the current media-range for a
time-progressing session can use the GET_PARAMETER method and include
the media-range. Furthermore, synchronization information can be
requested by using a combination of <xref
target="sec_RTP-Info">RTP-Info</xref> and <xref
target="sec_Range">Range</xref>.</t>
<t>RTSP 2.0 does not have a strong mechanism for providing negotiation
of the headers, or parameters and their formats, that can be used.
However, responses will indicate request-headers or parameters that
are not supported. A priori determination of what features are
available needs to be done through out-of-band mechanisms, like the
session description, or through the usage of <xref
target="sec_feature_tags">feature tags</xref>.</t>
</section>
<section title="Media Delivery">
<t>The delivery of media to the RTSP client is done with a protocol
outside of RTSP and this protocol is determined during the session
establishment. This document specifies how media is delivered with
<xref target="RFC3550">RTP</xref> over <xref
target="RFC0768">UDP</xref>, <xref target="RFC0793">TCP</xref> or the
RTSP connection. Additional protocols may be specified in the future
based on demand.</t>
<t>The usage of RTP as media delivery protocol requires some
additional information to function well. The PLAY response contains
information to enable reliable and timely delivery of how a client
should synchronize different sources in the different RTP sessions. It
also provides a mapping between RTP timestamps and the content time
scale. When the server wants to notify the client about the completion
of the media delivery, it sends a PLAY_NOTIFY request to the client.
The PLAY_NOTIFY request includes information about the stream end,
including the last RTP sequence number for each stream, thus enabling
the client to empty the buffer smoothly.</t>
<section title="Media Delivery Manipulations">
<t>The basic playback functionality of RTSP enables delivery of a
range of requested content to the client at the pace intended by the
content's creator. However, RTSP can also manipulate the delivery to
the client in two ways.</t>
<t><list style="hanging">
<t hangText="Scale:">The ratio of media content time delivered
per unit playback time.</t>
<t hangText="Speed:">The ratio of playback time delivered per
unit of wallclock time.</t>
</list>Both affect the media delivery per time unit. However, they
manipulate two independent time scales and the effects are possible
to combine.</t>
<t><xref target="sec_Scale">Scale</xref> is used for fast forward or
slow motion control as it changes the amount of content timescale
that should be played back per time unit. Scale > 1.0, means fast
forward, e.g., Scale=2.0 results in that 2 seconds of content is
played back every second of playback. Scale = 1.0 is the default
value that is used if no Scale is specified, i.e., playback at the
content's original rate. Scale values between 0 and 1.0 is providing
for slow motion. Scale can be negative to allow for reverse playback
in either regular pace (Scale = -1.0) or fast backwards (Scale <
-1.0) or slow motion backwards (-1.0 < Scale < 0). Scale = 0
is equal to pause and is not allowed.</t>
<t>In most cases the realization of scale means server side
manipulation of the media to ensure that the client can actually
play it back. The nature of these media manipulations and when they
are needed is highly media-type dependent. Let's consider an example
with two common media types audio and video.</t>
<t>It is very difficult to modify the playback rate of audio. A
maximum of 10-30% is possible by changing the pitch-rate of speech.
Music goes out of tune if one tries to manipulate the playback rate
by resampling it. This is a well known problem and audio is commonly
muted or played back in short segments with skips to keep up with
the current playback point.</t>
<t>For video it is possible to manipulate the frame rate, although
the rendering capabilities are often limited to certain frame rates.
Also the allowed bitrates in decoding, the structure used in the
encoding and the dependency between frames and other capabilities of
the rendering device limits the possible manipulations. Therefore,
the basic fast forward capabilities often are implemented by
selecting certain subsets of frames.</t>
<t>Due to the media restrictions, the possible scale values are
commonly restricted to the set of realizable scale ratios. To enable
the clients to select from the possible scale values, RTSP can
signal the supported Scale ratios for the content. To support
aggregated or dynamic content, where this may change during the
ongoing session and dependent on the location within the content, a
mechanism for updating the media properties and the scale factor
currently in use, exists.</t>
<t><xref target="sec_Speed">Speed</xref> affects how much of the
playback timeline is delivered in a given wallclock period. The
default is Speed = 1 which means to deliver at the same rate the
media is consumed. Speed > 1 means that the receiver will get
content faster than it regularly would consume it. Speed < 1
means that delivery is slower than the regular media rate. Speed
values of 0 or lower have no meaning and are not allowed. This
mechanism enables two general functionalities. One is client side
scale operations, i.e., the client receives all the frames and makes
the adjustment to the playback locally. The second is delivery
control for buffering of media. By specifying a speed over 1.0 the
client can build up the amount of playback time it has present in
its buffers to a level that is sufficient for its needs.</t>
<t>A naive implementation of Speed would only affect the
transmission schedule of the media and has a clear impact on the
needed bandwidth. This would result in the data rate being
proportional to the speed factor. Speed = 1.5, i.e., 50% faster than
normal delivery, would result in a 50% increase in the data
transport rate. If that can be supported or not depends solely on
the underlying network path. Scale may also have some impact on the
required bandwidth due to the manipulation of the content in the new
playback schedule. An example is fast forward where only the
independently decodable intra frames are included in the media
stream. This usage of solely intra frames increases the data rate
significantly compared to a normal sequence with the same number of
frames, where most frames are encoded using prediction.</t>
<t>This potential increase of the data rate needs to be handled by
the media sender. The client has requested that the media will be
delivered in a specific way, which should be honored. However, the
media sender cannot ignore if the network path between the sender
and the receiver can't handle the resulting media stream. In that
case the media stream needs to be adapted to fit the available
resources of the path. This can result in a reduced media
quality.</t>
<t>The need for bitrate adaptation becomes especially problematic in
connection with the Speed semantics. If the goal is to fill up the
buffer, the client may not want to do that at the cost of reduced
quality. If the client wants to make local playout changes then it
may actually require that the requested speed be honored. To resolve
this issue, Speed uses a range so that both cases can be supported.
The server is requested to use the highest possible speed value
within the range which is compatible with the available bandwidth.
As long as the server can maintain a speed value within the range it
shall not change the media quality, but instead modify the actual
delivery rate in response to available bandwidth and reflect this in
the Speed value in the response. However, if this is not possible,
the server should instead modify the media quality to respect the
lowest speed value and the available bandwidth.</t>
<t>This functionality enables the local scaling implementation to
use a tight range, or even a range where the lower bound equals the
upper bound, to identify that it requires the server to deliver the
requested amount of media time per delivery time independent of how
much it needs to adapt the media quality to fit within the available
path bandwidth. For buffer filling, it is suitable to use a range
with a reasonable span and with a lower bound at the nominal media
rate 1.0, such as 1.0 - 2.5. If the client wants to reduce the
buffer, it can specify an upper bound that is below 1.0 to force the
server to deliver slower than the nominal media rate.</t>
</section>
</section>
<section title="Session Maintenance and Termination">
<t>The session context that has been established is kept alive by
having the client show liveness. This is done in two main ways:<list
style="symbols">
<t>Media transport protocol keep-alive. RTP Control Protocol
(RTCP) may be used when using RTP.</t>
<t>Any RTSP request referencing the session context.</t>
</list></t>
<t><xref target="sec_liveness"/> discusses the methods for showing
liveness in more depth. If the client fails to show liveness for more
than the established session timeout value (normally 60 seconds), the
server may terminate the context. Other values may be selected by the
server through the inclusion of the timeout parameter in the session
header.</t>
<t>The session context is normally terminated by the client sending a
<xref target="sec_TEARDOWN">TEARDOWN request</xref> to the server
referencing the aggregated control URI. An individual media resource
can be removed from a session context by a TEARDOWN request
referencing that particular media resource. If all media resources are
removed from a session context, the session context is terminated.</t>
<t>A client may keep the session alive indefinitely if allowed by the
server; however, it is recommended to release the session context when
an extended period of time without media delivery activity has passed.
The client can re-establish the session context if required later.
What constitutes an extended period of time is dependent on the server
and its usage. It is recommended that the client terminates the
session before ten times the session timeout value has passed. A
server may terminate the session after one session timeout period
without any client activity beyond keep-alive. When a server
terminates the session context, it does that by sending a TEARDOWN
request indicating the reason.</t>
<t>A server can also request that the client tear down the session and
re-establish it at an alternative server, as may be needed for
maintenance. This is done by using the <xref
target="sec_REDIRECT">REDIRECT method</xref>. The <xref
target="sec_Terminate-Reason">Terminate-Reason header</xref> is used
to indicate when and why. The Location header indicates where it
should connect if there is an alternative server available. When the
deadline expires, the server simply stops providing the service. To
achieve a clean closure, the client needs to initiate session
termination prior to the deadline. In case the server has no other
server to redirect to, and wants to close the session for maintenance,
it shall use the TEARDOWN method with a Terminate-Reason header.</t>
</section>
<section anchor="sec_extend-rtsp" title="Extending RTSP">
<t>RTSP is quite a versatile protocol which supports extensions in
many different directions. Even this core specification contains
several blocks of functionality that are optional to implement. The
use case and need for the protocol deployment should determine what
parts are implemented. Allowing for extensions makes it possible for
RTSP to reach out to additional use cases. However, extensions will
affect the interoperability of the protocol and therefore it is
important that they can be added in a structured way.</t>
<t>The client can learn the capability of a server by using the <xref
target="sec_OPTIONS">OPTIONS method</xref> and the <xref
target="sec_Supported">Supported header</xref>. It can also try and
possibly fail using new methods, or require that particular features
are supported using the <xref target="sec_Require">Require</xref> or
<xref target="sec_Proxy-Require">Proxy-Require</xref> header.</t>
<t>The RTSP protocol in itself can be extended in three ways, listed
here in increasing order of the magnitude of changes supported: <list
style="symbols">
<t>Existing methods can be extended with new parameters, for
example, headers, as long as these parameters can be safely
ignored by the recipient. If the client needs negative
acknowledgment when a method extension is not supported, a tag
corresponding to the extension may be added in the field of the
Require or Proxy-Require headers.</t>
<t>New methods can be added. If the recipient of the message does
not understand the request, it must respond with error code 501
(Not Implemented) so that the sender can avoid using this method
again. A client may also use the OPTIONS method to inquire about
methods supported by the server. The server must list the methods
it supports using the Public response-header.</t>
<t>A new version of the protocol can be defined, allowing almost
all aspects (except the position of the protocol version number)
to change. A new version of the protocol must be registered
through an IETF standards track document.</t>
</list></t>
<t>The basic capability discovery mechanism can be used to both
discover support for a certain feature and to ensure that a feature is
available when performing a request. For a detailed explanation of
this see <xref target="sec_capability"/>.</t>
<t>New media delivery protocols may be added and negotiated at session
establishment, in addition to extensions to the core protocol. Certain
types of protocol manipulations can be done through parameter formats
using SET_PARAMETER and GET_PARAMETER.</t>
</section>
</section>
<section title="Document Conventions">
<t/>
<section anchor="sec_notational_conventions"
title="Notational Conventions">
<t>Since a few of the definitions are identical to HTTP/1.1, this
specification only points to the section where they are defined rather
than copying it. For brevity, [HX.Y] is to be taken to refer to
Section X.Y of the current HTTP/1.1 specification (<xref
target="RFC2616"/>).</t>
<t>All the mechanisms specified in this document are described in both
prose and the Augmented Backus-Naur form (ABNF) described in detail in
<xref target="RFC5234"/>.</t>
<t>Indented paragraphs are used to provide informative background and
motivation. This is intended to give readers who were not involved
with the formulation of the specification an understanding of why
things are the way they are in RTSP.</t>
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
<xref target="RFC2119"/>.</t>
<t>The word, "unspecified" is used to indicate functionality or
features that are not defined in this specification. Such
functionality cannot be used in a standardized manner without further
definition in an extension specification to RTSP.</t>
</section>
<section anchor="sec_Terminology" title="Terminology">
<t><list style="hanging">
<t hangText="Aggregate control:">The concept of controlling
multiple streams using a single timeline, generally maintained by
the server. A client, for example, uses aggregate control when it
issues a single play or pause message to simultaneously control
both the audio and video in a movie. A session which is under
aggregate control is referred to as an aggregated session.</t>
<t hangText="Aggregate control URI:">The URI used in an RTSP
request to refer to and control an aggregated session. It
normally, but not always, corresponds to the presentation URI
specified in the session description. See <xref
target="sec_SETUP"/> for more information.</t>
<t hangText="Client:">The client requests media service from the
media server.</t>
<t hangText="Connection:">A transport layer virtual circuit
established between two programs for the purpose of
communication.</t>
<t hangText="Container file:">A file which may contain multiple
media streams which often constitutes a presentation when played
together. The concept of a container file is not embedded in the
protocol. However, RTSP servers may offer aggregate control on the
media streams within these files.</t>
<t hangText="Continuous media:">Data where there is a timing
relationship between source and sink; that is, the sink needs to
reproduce the timing relationship that existed at the source. The
most common examples of continuous media are audio and motion
video. Continuous media can be real-time (interactive or
conversational), where there is a "tight" timing relationship
between source and sink, or streaming where the relationship is
less strict.</t>
<t hangText="Feature-tag:">A tag representing a certain set of
functionality, i.e., a feature.</t>
<t hangText="IRI:">Internationalized Resource Identifier, is the
same as a URI, with the exception that it allows characters from
the whole Universal Character Set (Unicode/ISO 10646), rather than
the US-ASCII only. See <xref target="RFC3987"/> for more
information.</t>
<t hangText="Live:">Normally used to describe a presentation or
session with media coming from an ongoing event. This generally
results in the session having an unbound or only loosely defined
duration, and sometimes no seek operations are possible.</t>
<t hangText="Media initialization:">Datatype/codec specific
initialization. This includes such things as clock rates, color
tables, etc. Any transport-independent information which is
required by a client for playback of a media stream occurs in the
media initialization phase of stream setup.</t>
<t hangText="Media parameter:">Parameter specific to a media type
that may be changed before or during stream delivery.</t>
<t hangText="Media server:">The server providing media delivery
services for one or more media streams. Different media streams
within a presentation may originate from different media servers.
A media server may reside on the same host or on a different host
from which the presentation is invoked.</t>
<t hangText="(Media) stream:">A single media instance, e.g., an
audio stream or a video stream as well as a single whiteboard or
shared application group. When using RTP, a stream consists of all
RTP and RTCP packets created by a source within an RTP
session.</t>
<t hangText="Message:">The basic unit of RTSP communication,
consisting of a structured sequence of octets matching the syntax
defined in <xref target="sec_syntax"/> and transmitted over a
connection-based transport. A message is either a Request or a
Response.</t>
<t hangText="Message Body:">The information transferred as the
payload of a message (Request or response). A message body
consists of meta-information in the form of message-body headers
and content in the form of a message-body, as described in <xref
target="sec_entity"/>.</t>
<t hangText="Non-Aggregated Control:">Control of a single media
stream.</t>
<t hangText="Presentation:">A set of one or more streams presented
to the client as a complete media feed and described by a
presentation description as defined below. Presentations with more
than one media stream are often handled in RTSP under aggregate
control.</t>
<t hangText="Presentation description:">A presentation description
contains information about one or more media streams within a
presentation, such as the set of encodings, network addresses and
information about the content. Other IETF protocols such as SDP
(<xref target="RFC4566"/>) use the term "session" for a
presentation. The presentation description may take several
different formats, including but not limited to the session
description protocol format, SDP.</t>
<t hangText="Response:">An RTSP response to a Request. One type of
RTSP message. If an HTTP response is meant, it is indicated
explicitly.</t>
<t hangText="Request:">An RTSP request. One type of RTSP message.
If an HTTP request is meant, it is indicated explicitly.</t>
<t hangText="Request-URI:">The URI used in a request to indicate
the resource on which the request is to be performed.</t>
<t hangText="RTSP agent:">Refers to either an RTSP client, an RTSP
server, or an RTSP proxy. In this specification, there are many
capabilities that are common to these three entities such as the
capability to send requests or receive responses. This term will
be used when describing functionality that is applicable to all
three of these entities.</t>
<t hangText="RTSP session:">A stateful abstraction upon which the
main control methods of RTSP operate. An RTSP session is a common
context; it is created and maintained on client's request and can
be destroyed by either the client or server. It is established by
an RTSP server upon the completion of a successful SETUP request
(when a 200 OK response is sent) and is labeled with a session
identifier at that time. The session exists until timed out by the
server or explicitly removed by a TEARDOWN request. An RTSP
session is a stateful entity; an RTSP server maintains an explicit
session state machine (see <xref target="sec_machine"/>) where
most state transitions are triggered by client requests. The
existence of a session implies the existence of state about the
session's media streams and their respective transport mechanisms.
A given session can have one or more media streams associated with
it. An RTSP server uses the session to aggregate control over
multiple media streams.</t>
<t hangText="Origin Server:">The server on which a given resource
resides.</t>
<t hangText="Transport initialization:">The negotiation of
transport information (e.g., port numbers, transport protocols)
between the client and the server.</t>
<t hangText="URI:">Universal Resource Identifier, see <xref
target="RFC3986"/>. The URIs used in RTSP are generally URLs as
they give a location for the resource. As URLs are a subset of
URIs, they will be referred to as URIs to cover also the cases
when an RTSP URI would not be an URL.</t>
<t hangText="URL:">Universal Resource Locator, is a URI which
identifies the resource through its primary access mechanism,
rather than identifying the resource by name or by some other
attribute(s) of that resource.</t>
</list></t>
</section>
</section>
<section anchor="sec_parameters" title="Protocol Parameters">
<section title="RTSP Version">
<t>This specification defines version 2.0 of RTSP.</t>
<t>RTSP uses a "<major>.<minor>" numbering scheme to
indicate versions of the protocol. The protocol versioning policy is
intended to allow the sender to indicate the format of a message and
its capacity for understanding further RTSP communication, rather than
the features obtained via that communication. No change is made to the
version number for the addition of message components which do not
affect communication behavior or which only add to extensible field
values.</t>
<t>The <minor> number is incremented when the changes made to
the protocol add features which do not change the general message
parsing algorithm, but which may add to the message semantics and
imply additional capabilities of the sender. The <major> number
is incremented when the format of a message within the protocol is
changed. The version of an RTSP message is indicated by an
RTSP-Version field in the first line of the message. Note that the
major and minor numbers MUST be treated as separate integers and that
each MAY be incremented higher than a single digit. Thus, RTSP/2.4 is
a lower version than RTSP/2.13, which in turn is lower than RTSP/12.3.
Leading zeros SHALL NOT be sent and MUST be ignored by recipients.</t>
</section>
<section anchor="sec_url" title="RTSP IRI and URI">
<t>RTSP 2.0 defines and registers or updates three URI schemes "rtsp",
"rtsps" and "rtspu". The usage of the last, "rtspu", is unspecified in
RTSP 2.0, and is defined here to register the URI scheme that was
defined in RTSP 1.0. The "rtspu" scheme indicates unspecified
transport of the RTSP messages over unreliable transport (UDP in RTSP
1.0). An RTSP server MUST respond with an error code indicating the
"rtspu" scheme is not implemented (501) to a request that carries a
"rtspu" URI scheme.</t>
<t>The details of the syntax of "rtsp" and "rtsps" URIs has been
changed from RTSP 1.0. These changes are:<list style="symbols">
<t>Support for IPV6 literal in host part and future IP literals
through RFC 3986 defined mechanism.</t>
<t>A new relative format to use in the RTSP protocol elements that
is not required to start with "/".</t>
</list>Neither should have any significant impact on
interoperability. If one is required to use IPv6 literals to reach an
RTSP server, then that RTSP server must be IPv6 capable, and RTSP 1.0
is not a fully IPv6 capable protocol. If an RTSP 1.0 client attempts
to process the URI it will not match the allowed syntax and be
considered invalid and processing will be stopped. This is clearly a
failure to reach the resource, however it is not a signification issue
as RTSP 2.0 support was needed anyway in both server and client. Thus
failure will only occur in a later step when there is a RTSP version
mismatch between client and server. The second change will only occur
inside RTSP message headers, as the request URI must be an absolute
URI. Thus such usages will only occur after an agent has accepted and
started processing RTSP 2.0 messages, and an RTSP 1.0 only agent will
not be required to parse such types of relative URIs.</t>
<t>This specification also defines the format of the RTSP IRI <xref
target="RFC3987"/> that can be used as RTSP resource identifiers and
locators, in web pages, user interfaces, on paper, etc. However, the
RTSP request message format only allows usage of the absolute URI
format. The RTSP IRI format MUST use the rules and transformation for
IRIs to URIs, as defined in <xref target="RFC3987"/>. This allows a
URI that matches the RTSP 2.0 specification, and so is suitable for
use in a request, to be created from an RTSP IRI.</t>
<t>The RTSP IRI and URI are both syntax restricted compared to the
generic syntax defined in <xref target="RFC3986"/> and <xref
target="RFC3987"/>: <list hangIndent="3" style="symbols">
<t>An absolute URI requires the authority part; i.e., a host
identity MUST be provided.</t>
<t>Parameters in the path element are prefixed with the reserved
separator ";".</t>
</list> The RTSP URI and IRI are case sensitive, with the exception
of those parts that <xref target="RFC3986"/> and <xref
target="RFC3987"/> define as case-insensitive; for example, the scheme
and host part.</t>
<t>The fragment identifier is used as defined in sections 3.5 and 4.3
of <xref target="RFC3986"/>, i.e., the fragment is to be stripped from
the IRI by the requester and not included in the request URI. The user
agent needs to interpret the value of the fragment based on the media
type the request relates to; i.e., the media type indicated in
Content-Type header in the response to DESCRIBE.</t>
<t>The syntax of any URI query string is unspecified and responder
(usually the server) specific. The query is, from the requester's
perspective, an opaque string and needs to be handled as such. Please
note that relative URI with queries are difficult to handle due to the
RFC 3986 relative URI handling rules. Any change of the path element
using a relative URI results in the stripping of the query, which
means the relative part needs to contain the query.</t>
<t>The URI scheme "rtsp" requires that commands are issued via a
reliable protocol (within the Internet, TCP), while the scheme "rtsps"
identifies a reliable transport using secure transport (TLS <xref
target="RFC5246"/>, see (<xref target="sec_security-framework"/>).</t>
<t>For the scheme "rtsp", if no port number is provided in the
authority part of the URI, the port number 554 MUST be used. For the
scheme "rtsps", if no port number is provided in the authority part of
the URI port number, the TCP port 322 MUST be used.</t>
<t>A presentation or a stream is identified by a textual media
identifier, using the character set and escape conventions of URIs
<xref target="RFC3986"/>. URIs may refer to a stream or an aggregate
of streams; i.e., a presentation. Accordingly, requests described in
(<xref target="sec_methods"/>) can apply to either the whole
presentation or an individual stream within the presentation. Note
that some request methods can only be applied to streams, not
presentations, and vice versa.</t>
<t>For example, the RTSP URI: <list style="hanging">
<t>rtsp://media.example.com:554/twister/audiotrack</t>
</list> may identify the audio stream within the presentation
"twister", which can be controlled via RTSP requests issued over a TCP
connection to port 554 of host media.example.com.</t>
<t>Also, the RTSP URI: <list style="hanging">
<t>rtsp://media.example.com:554/twister</t>
</list> identifies the presentation "twister", which may be composed
of audio and video streams, but could also be something else like a
random media redirector.</t>
<t><list style="hanging">
<t>This does not imply a standard way to reference streams in
URIs. The presentation description defines the hierarchical
relationships in the presentation and the URIs for the individual
streams. A presentation description may name a stream "a.mov" and
the whole presentation "b.mov".</t>
</list></t>
<t>The path components of the RTSP URI are opaque to the client and do
not imply any particular file system structure for the server.</t>
<t><list style="hanging">
<t>This decoupling also allows presentation descriptions to be
used with non-RTSP media control protocols simply by replacing the
scheme in the URI.</t>
</list></t>
</section>
<section anchor="sec_session-id" title="Session Identifiers">
<t>Session identifiers are strings of length 8-128 characters. A
session identifier MUST be chosen cryptographically random (see <xref
target="RFC4086"/>). It is RECOMMENDED that it contains 128 bits of
entropy, i.e., approximately 22 characters from a high quality
generator (see <xref target="sec_security"/>). However, note that the
session identifier does not provide any security against session
hijacking unless it is kept confidential by the client, server and
trusted proxies.</t>
</section>
<section anchor="sec_Media-Time-Formats" title="Media Time Formats">
<t>RTSP currently supports three different media time formats defined
below. Additional time formats may be specified in the future. These
time formats can be used with the <xref target="sec_Range">Range
header</xref> to request playback and specify at which media position
protocol requests actually will or have taken place. They are also
used in description of the media's properties using the <xref
target="sec_Media-Range">Media-Range header</xref>. The unqualified
format identifier is used on its own in <xref
target="sec_Accept-Ranges">Accept-Ranges header</xref> to declare
supported time formats and also in the <xref target="sec_Range">Range
header</xref> to request the time format used in the response.</t>
<section anchor="sec_smpte" title="SMPTE Relative Timestamps">
<t>A Society of Motion Picture and Television Engineers (SMPTE)
relative timestamp expresses time relative to the start of the clip.
Relative timestamps are expressed as <xref target="SMPTE_TC">SMPTE
time codes</xref> for frame-level access accuracy. The time code has
the format <list style="hanging">
<t>hours:minutes:seconds:frames.subframes,</t>
</list> with the origin at the start of the clip. The default
SMPTE format is "SMPTE 30 drop" format, with frame rate is 29.97
frames per second. Other SMPTE codes MAY be supported (such as
"SMPTE 25") through the use of "smpte-type". For SMPTE 30, the
"frames" field in the time value can assume the values 0 through 29.
The difference between 30 and 29.97 frames per second is handled by
dropping the first two frame indices (values 00 and 01) of every
minute, except every tenth minute. If the frame and the subframe
values are zero, they may be omitted. Subframes are measured in
one-hundredth of a frame.</t>
<t>Examples:</t>
<figure>
<artwork><![CDATA[
smpte=10:12:33:20-
smpte=10:07:33-
smpte=10:07:00-10:07:33:05.01
smpte-25=10:07:00-10:07:33:05.01
]]></artwork>
</figure>
</section>
<section anchor="sec_npt" title="Normal Play Time">
<t>Normal play time (NPT) indicates the stream absolute position
relative to the beginning of the presentation. The timestamp
consists of two parts: the mandatory first part may be expressed in
either seconds or hours, minutes, and seconds. The optional second
part consists of a decimal point and decimal figures and indicates
fractions of a second.</t>
<t>The beginning of a presentation corresponds to 0.0 seconds.
Negative values are not defined.</t>
<t>The special constant "now" is defined as the current instant of a
live event. It MAY only be used for live events, and MUST NOT be
used for on-demand (i.e., non-live) content.</t>
<t>NPT is defined as in DSM-CC <xref target="ISO.13818-6.1995"/>:
"Intuitively, NPT is the clock the viewer associates with a program.
It is often digitally displayed on a VCR. NPT advances normally when
in normal play mode (scale = 1), advances at a faster rate when in
fast scan forward (high positive scale ratio), decrements when in
scan reverse (negative scale ratio) and is fixed in pause mode. NPT
is (logically) equivalent to SMPTE time codes."</t>
<t>Examples:</t>
<figure>
<artwork><![CDATA[
npt=123.45-125
npt=12:05:35.3-
npt=now-
]]></artwork>
</figure>
<t>The syntax is based on ISO 8601 <xref target="ISO.8601.2000"/>.
Two different notations are allowed. The npt-hhmmss notation are
using a ISO 8601 extended complete representation of the time of the
day format (Section 5.3.1.1 of <xref target="ISO.8601.2000"/>) using
colon (":") as separators between hours, minutes and seconds
(hh:mm:ss). With the exception that it expresses duration since
presentation start rather than time since midnight and the hour
counter is not limited to 0-24 hours, instead up to nineteen (19)
digits of hours are allowed. ISO 8601 time format requires all
digits to be used for each format, and all format required needs to
be included, e.g. if one use a hh:mm:ss format, then that requires
two digits for hours, two digits for minutes and two digits for
second, a time value such as 7 minutes and 0 seconds, is expressed
as 00:07:00. The npt-sec notation is expressing the duration since
presentation start in seconds, using one to nineteen (19) digits.
Both notations allows decimal fractions of seconds as specified in
Section 5.3.1.3 of <xref target="ISO.8601.2000"/> with the
limitation of at maximum of 9 digits and only allowing "." (full
stop) as separator. Due to RTSP 1.0 and the fact that the highest
values are expanded beyond two digits, all implementations MUST
allow the highest value to be single digit and SHALL NOT send
leading zeros for hours in the npt-hhmmss notation and leading zeros
for seconds in the npt-sec notation. The hours and the seconds in
the npt-hhmmss notation SHALL be sent using leading zeros, but
receivers SHALL accept values without leading zeros.</t>
<t>The npt-sec notation is optimized for automatic generation, the
npt-hhmmss notation for consumption by human readers. The "now"
constant allows clients to request to receive the live feed rather
than the stored or time-delayed version. This is needed since
neither absolute time nor zero time are appropriate for this
case.</t>
</section>
<section anchor="sec_clock" title="Absolute Time">
<t>Absolute time is expressed following a specific types of ISO 8601
<xref target="ISO.8601.2000"/> based timestamps. The date is
complete representation calendar date in basic format (YYYYMMDD)
without separators (per Section 5.2.1.1 of <xref
target="ISO.8601.2000"/>). The time of day is provided in the
complete representation basic format (hhmmss) as specified in
Section 5.3.1.1 of <xref target="ISO.8601.2000"/>, allowing decimal
fractions of seconds following Section 5.3.1.3 requiring "." (full
stop) as decimal separator and limiting the number of digits to no
more than nine (9). The time expressed MUST be using UTC (GMT), i.e.
no timezone offsets allowed. The full date and time specification is
the eight digit date followed by a "T" followed by the six digits
time value, optionally followed by a full stop followed by one to
nine fractions of a second and ended by "Z", e.g.
YYYYMMDDThhmmss.ssZ.</t>
<t><list style="empty">
<t>The reason for this time format rather than using <xref
target="RFC3339">"Date and Time on the Internet:
Timestamps"</xref> are historic and using the format specified
in RTSP 1.0. The motivations raised in RFC 3339 applies to why a
selection from ISO 8601 was done, but a different and even more
restrictive selection was applied in this case.</t>
</list>Example for clock format range request for a starting time
of November 8, 1996 at 14h 37 min and 20 and a quarter seconds UTC
playing for 10 min and 5 seconds, a Media-Properties header's
"Time-Limited UTC property for 24th of December 2014 at 15 hours and
00 mins, and a Terminate-Readon headers "time" property for 18th of
June 2013 at 16 hours, 12 minutes and 56 seconds:</t>
<figure>
<artwork><![CDATA[
clock=19961108T143720.25Z-19961108T144725.25Z
Time-Limited=20141224T1500Z
time=20130618T161256Z]]></artwork>
</figure>
</section>
</section>
<section anchor="sec_feature_tags" title="Feature-Tags">
<t>Feature-tags are unique identifiers used to designate features in
RTSP. These tags are used in Require (<xref target="sec_Require"/>),
Proxy-Require (<xref target="sec_Proxy-Require"/>), Proxy-Supported
(<xref target="sec_Proxy-Supported"/>), Supported (<xref
target="sec_Supported"/>) and Unsupported (<xref
target="sec_Unsupported"/>) header fields.</t>
<t>A feature-tag definition MUST indicate which combination of
clients, servers or proxies it applies to.</t>
<t>The creator of a new RTSP feature-tag should either prefix the
feature-tag with a reverse domain name (e.g.,
"com.example.mynewfeature" is an apt name for a feature whose inventor
can be reached at "example.com"), or register the new feature-tag with
the Internet Assigned Numbers Authority (IANA) (see IANA <xref
target="sec_IANA"/>).</t>
<t>The usage of feature-tags is further described in <xref
target="sec_capability"/> that deals with capability handling.</t>
</section>
<section anchor="sec_message-tags" title="Message Body Tags">
<t>Message body tags are opaque strings that are used to compare two
message bodies from the same resource, for example in caches or to
optimize setup after a redirect. Message body tags can be carried in
the MTag header (see <xref target="sec_MTag"/>) or in SDP (see <xref
target="sec_sdp-mtag"/>). MTag is similar to ETag in HTTP/1.1 (see
Section 3.11 of <xref target="RFC2068"/>).</t>
<t>A message body tag MUST be unique across all versions of all
message bodies associated with a particular resource. A given message
body tag value MAY be used for message bodies obtained by requests on
different URIs. The use of the same message body tag value in
conjunction with message bodies obtained by requests on different URIs
does not imply the equivalence of those message bodies</t>
<t>Message body tags are used in RTSP to make some methods
conditional. The methods are made conditional through the inclusion of
headers; see "<xref target="sec_If-Match">If-Match"</xref> and "<xref
target="sec_If-None-Match">If-None-Match"</xref>. Note that RTSP
message body tags apply to the complete presentation; i.e., both the
presentation description and the individual media streams. Thus
message body tags can be used to verify at setup time after a redirect
that the same session description applies to the media at the new
location using the If-Match header.</t>
</section>
<section anchor="sec_Media-Properties-Intro" title="Media Properties">
<t>When an RTSP server handles media, it is important to consider the
different properties a media instance for delivery and playback can
have. This specification considers the media properties listed below
in its protocol operations. They are derived from the differences
between a number of supported usages. <list style="hanging">
<t hangText="On-demand:">Media that has a fixed (given) duration
that doesn't change during the life time of the RTSP session and
is known at the time of the creation of the session. It is
expected that the content of the media will not change, even if
the representation, i.e., encoding, quality, etc, may change.
Generally one can seek, i.e., request any range, within the
media.</t>
<t hangText="Dynamic On-demand:">This is a variation of the
on-demand case where external methods are used to manipulate the
actual content of the media setup for the RTSP session. The main
example is a content defined by a playlist.</t>
<t hangText="Live:">Live media represents a progressing content
stream (such as broadcast TV) where the duration may or may not be
known. It is not seekable, only the content presently being
delivered can be accessed.</t>
<t hangText="Live with Recording:">A Live stream that is combined
with a server-side capability to store and retain the content of
the live session, and allow for random access delivery within the
part of the already recorded content. The actual behavior of the
media stream is very much dependent on the retention policy for
the media stream; either the server will be able to capture the
complete media stream, or it will have a limitation in how much
will be retained. The media range will dynamically change as the
session progress. For servers with a limited amount of storage
available for recording, there will typically be a sliding window
that moves forward while new data is made available and older data
is discarded.</t>
</list></t>
<t>To cover the above usages, the following media properties with
appropriate values are specified:</t>
<section anchor="sec_random_access_seek"
title="Random Access and Seeking">
<t>Random Access is the ability to specify and get media delivered
starting from any time instant within the content, an operation
called seeking. The Media-Properties header will indicate the
general capability for a media resource to perform random
access:</t>
<t><list style="hanging">
<t hangText="Random-Access:">The media is seekable to any out of
a large number of points within the media. Due to media encoding
limitations, a particular point may not be reachable, but
seeking to a point close by is enabled. A floating point number
of seconds may be provided to express the worst case distance
between random access points.</t>
<t hangText="Beginning-Only:">Seeking is only possible to the
beginning of the content.</t>
<t hangText="No-seeking:">Seeking is not possible at all.</t>
</list></t>
<t>If random access is possible, as indicated by the
Media-Properties header, the actual behavior policy when seeking can
be controlled using the <xref target="sec_Seek-Style">Seek-Style
header</xref>.</t>
</section>
<section title="Retention">
<t>Media may have different retention policies in place that affect
the operation on media. The following different media retention
policies are envisioned and taken into consideration where
applicable:</t>
<t><list style="hanging">
<t hangText="Unlimited:">The media will not be removed as long
as the RTSP session is in existence.</t>
<t hangText="Time-Limited:">The media will not be removed before
the given wallclock time. After that time it may or may not be
available any more.</t>
<t hangText="Time-Duration:">Each individual unit of the media
will be retained for the specified duration.</t>
</list></t>
<t/>
</section>
<section title="Content Modifications">
<t>There is also the question of how the content may change over
time for a given media resource:</t>
<t><list style="hanging">
<t hangText="Immutable:">The content of the media will not
change, even if the representation, i.e., encoding, quality,
etc., may change.</t>
<t hangText="Dynamic:">Between explicit updates the media
content will not change, but the content may change due to
external methods or triggers, such as playlists.</t>
<t hangText="Time-Progressing:">As time progresses new content
will become available. If the content also is retained it will
become longer as everything between the start point and the
point currently being made available can be accessed. If the
media server uses a sliding window policy for retention, the
start point will also change as time progresses.</t>
</list></t>
<t/>
</section>
<section title="Supported Scale Factors">
<t>Content often supports only a limited set or range of scales when
delivering the media.. To enable the client to know what values or
ranges of scale operations that the whole content or the current
position supports, a media properties attribute for this is defined
which contains a list with the values and/or ranges that are
supported. The attribute is named "Scales". It may be updated at any
point in the content due to content consisting of spliced pieces or
content being dynamically updated by out-of-band mechanisms.</t>
</section>
<section title="Mapping to the Attributes">
<t>This section shows examples of how one would map the above usages
to the properties and their values.</t>
<t><list style="hanging">
<t hangText="On-demand:">Random Access: Random-Access=5.0,
Content Modifications: Immutable, Retention: Unlimited or
Time-Limited.</t>
<t hangText="Dynamic On-demand:">Random Access:
Random-Access=3.0, Content Modifications: Dynamic, Retention:
Unlimited or Time-Limited.</t>
<t hangText="Live:">Random Access: No-seeking, Content
Modifications: Time-Progressing, Retention:
Time-Duration=0.0</t>
<t hangText="Live with Recording:">Random Access:
Random-Access=3.0, Content Modifications: Time-Progressing,
Retention: Time-Duration=7200.0</t>
</list></t>
</section>
</section>
</section>
<section anchor="sec_message" title="RTSP Message">
<t>RTSP is a text-based protocol and uses the ISO 10646 character set in
UTF-8 encoding RFC 3629 <xref target="RFC3629"/>. Lines MUST be
terminated by CRLF.</t>
<t><list style="hanging">
<t>Text-based protocols make it easier to add optional parameters in
a self-describing manner. Since the number of parameters and the
frequency of commands is low, processing efficiency is not a
concern. Text-based protocols, if done carefully, also allow easy
implementation of research prototypes in scripting languages such as
TCL, Visual Basic and Perl.</t>
</list></t>
<t>The ISO 10646 character set avoids character set switching, but is
invisible to the application as long as US-ASCII is being used. This is
also the encoding used for <xref target="RFC3550">RTCP</xref>.</t>
<t>Requests contain methods, the object the method is operating upon and
parameters to further describe the method. Methods are idempotent unless
otherwise noted. Methods are also designed to require little or no state
maintenance at the media server.</t>
<section anchor="sec_message-types" title="Message Types">
<t>RTSP messages consist of requests from client to server, or server
to client, and responses in the reverse direction. Request (<xref
target="sec_request"/>) and Response (<xref target="sec_response"/>)
messages use a format based on the generic message format of RFC 5322
<xref target="RFC5322"/> for transferring bodies (the payload of the
message). Both types of messages consist of a start-line, zero or more
header fields (also known as "headers"), an empty line (i.e., a line
with nothing preceding the CRLF) indicating the end of the headers,
and possibly the data of the message body. The below <xref
target="RFC5234">ABNF</xref> is for information and the formal message
specification is present in <xref
target="sec_syntax-prot-message"/>.</t>
<figure>
<artwork><![CDATA[
generic-message = start-line
*(rtsp-header CRLF)
CRLF
[ message-body-data ]
start-line = Request-Line | Status-Line
]]></artwork>
</figure>
<t>In the interest of robustness, agents MUST ignore any empty line(s)
received where a Request-Line or Response-Line is expected. In other
words, if the agent is reading the protocol stream at the beginning of
a message and receives a CRLF first, it MUST ignore the CRLF.</t>
</section>
<section anchor="sec_message-headers" title="Message Headers">
<t>RTSP header fields (see <xref target="sec_headers"/>) include
general-header, request-header, response-header, and message-body
header fields.</t>
<t>The order in which header fields with differing field names are
received is not significant. However, it is "good practice" to send
general-header fields first, followed by request-header or response-
header fields, and ending with the Message-body header fields.</t>
<t>Multiple header fields with the same field-name MAY be present in a
message if and only if the entire field-value for that header field is
defined as a comma-separated list. It MUST be possible to combine the
multiple header fields into one "field-name: field-value" pair,
without changing the semantics of the message, by appending each
subsequent field-value to the first, each separated by a comma. The
order in which header fields with the same field-name are received is
therefore significant to the interpretation of the combined field
value, and thus a proxy MUST NOT change the order of these field
values when a message is forwarded.</t>
<t>Unknown message headers MUST be ignored (skipping over the header
to the next protocol element, and not causing an error) by a RTSP
server or client. An RTSP Proxy MUST forward unknown message headers.
Message headers defined outside of this specification that are
required to be interpreted by the RTSP agent will need to use <xref
target="sec_feature_tags">feature tags</xref> and include them in the
appropriate <xref target="sec_Require">Require</xref> or <xref
target="sec_Proxy-Require">Proxy-Require</xref> header.</t>
</section>
<section anchor="sec_message-body" title="Message Body">
<t>The message body (if any) of an RTSP message is used to carry
further information for a particular resource associated with the
request or response. An example of a message body is a Session
Description Protocol (SDP) message.</t>
<t>The presence of a message body in either a request or a response
MUST be signaled by the inclusion of a Content-Length header (see
<xref target="sec_Content-Length"/>) and Content-Type (see <xref
target="sec_Content-Type"/>). A message body MUST NOT be included in a
request or response if the specification of the particular method (see
<xref target="sec_methods">Method Definitions </xref>) does not allow
sending a message body. In case a message body is received in a
message when not expected the message body data SHOULD be discarded.
This is to allow future extensions to define optional use of a message
body.</t>
</section>
<section title="Message Length">
<t>An RTSP Message that does not contain any message body is
terminated by the first empty line after the header fields (Note: An
empty line is a line with nothing preceding the CRLF.). In RTSP
messages that contain message bodies the empty line is followed by the
message body. The length of that body is determined by the value of
the <xref target="sec_Content-Length">Content-Length header</xref>.
The value in the header represents the length of the message-body in
octets. If this header field is not present, a value of zero is
assumed, i.e., no message body present in the message. Unlike an HTTP
message, an RTSP message MUST contain a Content-Length header whenever
it contains a message body. Note that RTSP does not support the
HTTP/1.1 "chunked" transfer coding (see [H3.6.1]).</t>
<t><list style="empty">
<t>Given the moderate length of presentation descriptions
returned, the server should always be able to determine its
length, even if it is generated dynamically, making the chunked
transfer encoding unnecessary.</t>
</list></t>
</section>
</section>
<section anchor="sec_general-header" title="General Header Fields">
<t>General headers are headers that may be used in both requests and
responses. The general-headers are listed in <xref
target="tab_headers-general"/>:</t>
<texttable anchor="tab_headers-general"
title="The general headers used in RTSP">
<preamble/>
<ttcol align="left">Header Name</ttcol>
<ttcol align="left">Defined in Section</ttcol>
<c>Accept-Ranges</c>
<c><xref target="sec_Accept-Ranges"/></c>
<c>Cache-Control</c>
<c><xref target="sec_Cache-Control"/></c>
<c>Connection</c>
<c><xref target="sec_Connection"/></c>
<c>CSeq</c>
<c><xref target="sec_CSeq"/></c>
<c>Date</c>
<c><xref target="sec_Date"/></c>
<c>Media-Properties</c>
<c><xref target="sec_Media-Properties"/></c>
<c>Media-Range</c>
<c><xref target="sec_Media-Range"/></c>
<c>Pipelined-Requests</c>
<c><xref target="sec_Pipelined-Requests"/></c>
<c>Proxy-Supported</c>
<c><xref target="sec_Proxy-Supported"/></c>
<c>Range</c>
<c><xref target="sec_Range"/></c>
<c>RTP-Info</c>
<c><xref target="sec_RTP-Info"/></c>
<c>Scale</c>
<c><xref target="sec_Scale"/></c>
<c>Seek-Style</c>
<c><xref target="sec_Seek-Style"/></c>
<c>Server</c>
<c><xref target="sec_Server"/></c>
<c>Session</c>
<c><xref target="sec_Session"/></c>
<c>Speed</c>
<c><xref target="sec_Speed"/></c>
<c>Supported</c>
<c><xref target="sec_Supported"/></c>
<c>Timestamp</c>
<c><xref target="sec_Timestamp"/></c>
<c>Transport</c>
<c><xref target="sec_Transport"/></c>
<c>User-Agent</c>
<c><xref target="sec_User-Agent"/></c>
<c>Via</c>
<c><xref target="sec_Via"/></c>
</texttable>
</section>
<section anchor="sec_request" title="Request">
<t>A request message uses the format outlined below regardless of the
direction of a request, client to server or server to client: <list
hangIndent="3" style="symbols">
<t>Request line, containing the method to be applied to the
resource, the identifier of the resource, and the protocol version
in use;</t>
<t>Zero or more Header lines, that can be of the following types:
general-headers (<xref target="sec_general-header"/>),
request-headers (<xref target="sec_request-header"/>), or message
body headers (<xref target="sec_message-header"/>);</t>
<t>One empty line (CRLF) to indicate the end of the header
section;</t>
<t>Optionally a message-body, consisting of one or more lines. The
length of the message body in octets is indicated by the
Content-Length message header.</t>
</list></t>
<section anchor="sec_request-line" title="Request Line">
<t>The request line provides the key information about the request:
what method, on what resources and using which RTSP version. The
methods that are defined by this specification are listed in <xref
target="tab_request-methods"/>.</t>
<texttable anchor="tab_request-methods" title="The RTSP Methods">
<preamble/>
<ttcol align="left">Method</ttcol>
<ttcol align="left">Defined in Section</ttcol>
<c>DESCRIBE</c>
<c><xref target="sec_DESCRIBE"/></c>
<c>GET_PARAMETER</c>
<c><xref target="sec_GET_PARAMETER"/></c>
<c>OPTIONS</c>
<c><xref target="sec_OPTIONS"/></c>
<c>PAUSE</c>
<c><xref target="sec_PAUSE"/></c>
<c>PLAY</c>
<c><xref target="sec_PLAY"/></c>
<c>PLAY_NOTIFY</c>
<c><xref target="sec_PLAY_NOTIFY"/></c>
<c>REDIRECT</c>
<c><xref target="sec_REDIRECT"/></c>
<c>SETUP</c>
<c><xref target="sec_SETUP"/></c>
<c>SET_PARAMETER</c>
<c><xref target="sec_SET_PARAMETER"/></c>
<c>TEARDOWN</c>
<c><xref target="sec_TEARDOWN"/></c>
</texttable>
<t>The syntax of the RTSP request line is the following: <list
style="hanging">
<t><Method> SP <Request-URI> SP <RTSP-Version>
CRLF</t>
</list> Note: This syntax cannot be freely changed in future
versions of RTSP. This line needs to remain parsable by older RTSP
implementations since it indicates the RTSP version of the
message.</t>
<t>In contrast to HTTP/1.1 <xref target="RFC2616"/>, RTSP requests
identify the resource through an absolute RTSP URI (including scheme,
host, and port) (see <xref target="sec_url"/>) rather than just the
absolute path.</t>
<t><list style="hanging">
<t>HTTP/1.1 requires servers to understand the absolute URI, but
clients are supposed to use the Host request-header. This is
purely needed for backward-compatibility with HTTP/1.0 servers, a
consideration that does not apply to RTSP.</t>
</list></t>
<t>An asterisk "*" can be used instead of an absolute URI in the
Request-URI part to indicate that the request does not apply to a
particular resource, but to the server or proxy itself, and is only
allowed when the request method does not necessarily apply to a
resource.</t>
<t>For example: <list style="hanging">
<t>OPTIONS * RTSP/2.0</t>
</list></t>
<t>An OPTIONS in this form will determine the capabilities of the
server or the proxy that first receives the request. If the capability
of the specific server needs to be determined, without regard to the
capability of an intervening proxy, the server should be addressed
explicitly with an absolute URI that contains the server's
address.</t>
<t>For example: <list style="hanging">
<t>OPTIONS rtsp://example.com RTSP/2.0</t>
</list></t>
</section>
<section anchor="sec_request-header" title="Request Header Fields">
<t>The RTSP headers in <xref target="tab_request-header"/> can be
included in a request, as request-headers, to modify the specifics of
the request.</t>
<texttable anchor="tab_request-header"
title="The RTSP request headers">
<preamble/>
<ttcol align="left">Header</ttcol>
<ttcol align="left">Defined in Section</ttcol>
<c>Accept</c>
<c><xref target="sec_Accept"/></c>
<c>Accept-Credentials</c>
<c><xref target="sec_Accept-Credentials"/></c>
<c>Accept-Encoding</c>
<c><xref target="sec_Accept-Encoding"/></c>
<c>Accept-Language</c>
<c><xref target="sec_Accept-Language"/></c>
<c>Authorization</c>
<c><xref target="sec_Authorization"/></c>
<c>Bandwidth</c>
<c><xref target="sec_Bandwidth"/></c>
<c>Blocksize</c>
<c><xref target="sec_Blocksize"/></c>
<c>From</c>
<c><xref target="sec_From"/></c>
<c>If-Match</c>
<c><xref target="sec_If-Match"/></c>
<c>If-Modified-Since</c>
<c><xref target="sec_If-Modified-Since"/></c>
<c>If-None-Match</c>
<c><xref target="sec_If-None-Match"/></c>
<c>Notify-Reason</c>
<c><xref target="sec_Notify-Reason"/></c>
<c>Proxy-Authorization</c>
<c><xref target="sec_Proxy-Authorization"/></c>
<c>Proxy-Require</c>
<c><xref target="sec_Proxy-Require"/></c>
<c>Referrer</c>
<c><xref target="sec_Referrer"/></c>
<c>Request-Status</c>
<c><xref target="sec_Request-Status"/></c>
<c>Require</c>
<c><xref target="sec_Require"/></c>
<c>Terminate-Reason</c>
<c><xref target="sec_Terminate-Reason"/></c>
</texttable>
<t>Detailed header definitions are provided in <xref
target="sec_headers"/>.</t>
<t>New request-headers may be defined. If the receiver of the request
is required to understand the request-header, the request MUST include
a corresponding feature tag in a Require or Proxy-Require header to
ensure the processing of the header.</t>
</section>
</section>
<!-- title="Request" -->
<section anchor="sec_response" title="Response">
<t>After receiving and interpreting a request message, the recipient
responds with an RTSP response message. Normally, there is only one,
final, response. Only responses using the response code class 1xx, are
allowed to send one or more 1xx response messages prior to the final
response message.</t>
<t>The valid response codes and the methods they can be used with are
listed in <xref target="tab_status"/>.</t>
<section anchor="sec_status-line" title="Status-Line">
<t>The first line of a Response message is the Status-Line, consisting
of the protocol version followed by a numeric status code and the
textual phrase associated with the status code, with each element
separated by SP characters. No CR or LF is allowed except in the final
CRLF sequence.</t>
<t><RTSP-Version> SP <Status-Code> SP
<Reason-Phrase> CRLF</t>
<section anchor="sec_status-code"
title="Status Code and Reason Phrase">
<t>The Status-Code element is a 3-digit integer result code of the
attempt to understand and satisfy the request. These codes are fully
defined in <xref target="sec_status"/>. The Reason-Phrase is
intended to give a short textual description of the Status-Code. The
Status-Code is intended for use by automata and the Reason-Phrase is
intended for the human user. The client is not required to examine
or display the Reason-Phrase.</t>
<t>The first digit of the Status-Code defines the class of response.
The last two digits do not have any categorization role. There are 5
values for the first digit: <list hangIndent="6" style="hanging">
<t hangText="1xx:">Informational - Request received, continuing
process</t>
<t hangText="2xx:">Success - The action was successfully
received, understood, and accepted</t>
<t hangText="3rr:">Redirection - Further action needs to be
taken in order to complete the request (3rr rather than 3xx is
used as 304 is excluded, see <xref
target="sec_status-redirect"/>)</t>
<t hangText="4xx:">Client Error - The request contains bad
syntax or cannot be fulfilled</t>
<t hangText="5xx:">Server Error - The server failed to fulfill
an apparently valid request</t>
</list> The individual values of the numeric status codes defined
for RTSP/2.0, and an example set of corresponding Reason-Phrases,
are presented in <xref target="tab_status"/>. The reason phrases
listed here are only recommended; they may be replaced by local
equivalents without affecting the protocol. Note that RTSP adopts
most HTTP/1.1 <xref target="RFC2616"/> status codes and adds
RTSP-specific status codes starting at x50 to avoid conflicts with
future HTTP status codes that are desirable to import into RTSP. All
these codes are RTSP specific and RTSP has its own registry separate
from HTTP for status codes.</t>
<t>RTSP status codes are extensible. RTSP applications are not
required to understand the meaning of all registered status codes,
though such understanding is obviously desirable. However,
applications MUST understand the class of any status code, as
indicated by the first digit, and treat any unrecognized response as
being equivalent to the x00 status code of that class, with an
exception for unknown 3xx codes, which MUST be treated as a 302
(Found). The reason being that no 300 (Multiple Choices in HTTP) is
defined for RTSP. An response with an unrecognized status code MUST
NOT be cached. For example, if an unrecognized status code of 431 is
received by the client, it can safely assume that there was
something wrong with its request and treat the response as if it had
received a 400 status code. In such cases, user agents SHOULD
present to the user the message body returned with the response,
since that message body is likely to include human-readable
information which will explain the unusual status.</t>
<texttable anchor="tab_status"
title="Status codes and their usage with RTSP methods">
<preamble/>
<ttcol align="left">Code</ttcol>
<ttcol align="left">Reason</ttcol>
<ttcol align="left">Method</ttcol>
<c>100</c>
<c>Continue</c>
<c>all</c>
<c/>
<c/>
<c/>
<c>200</c>
<c>OK</c>
<c>all</c>
<c/>
<c/>
<c/>
<c>301</c>
<c>Moved Permanently</c>
<c>all</c>
<c>302</c>
<c>Found</c>
<c>all</c>
<c>303</c>
<c>reserved</c>
<c>n/a</c>
<c>304</c>
<c>Not Modified</c>
<c>all</c>
<c>305</c>
<c>Use Proxy</c>
<c>all</c>
<c/>
<c/>
<c/>
<c>400</c>
<c>Bad Request</c>
<c>all</c>
<c>401</c>
<c>Unauthorized</c>
<c>all</c>
<c>402</c>
<c>Payment Required</c>
<c>all</c>
<c>403</c>
<c>Forbidden</c>
<c>all</c>
<c>404</c>
<c>Not Found</c>
<c>all</c>
<c>405</c>
<c>Method Not Allowed</c>
<c>all</c>
<c>406</c>
<c>Not Acceptable</c>
<c>all</c>
<c>407</c>
<c>Proxy Authentication Required</c>
<c>all</c>
<c>408</c>
<c>Request Timeout</c>
<c>all</c>
<c>410</c>
<c>Gone</c>
<c>all</c>
<c>412</c>
<c>Precondition Failed</c>
<c>DESCRIBE, SETUP</c>
<c>413</c>
<c>Request Message Body Too Large</c>
<c>all</c>
<c>414</c>
<c>Request-URI Too Long</c>
<c>all</c>
<c>415</c>
<c>Unsupported Media Type</c>
<c>all</c>
<c>451</c>
<c>Parameter Not Understood</c>
<c>SET_PARAMETER, GET_PARAMETER</c>
<c>452</c>
<c>reserved</c>
<c>n/a</c>
<c>453</c>
<c>Not Enough Bandwidth</c>
<c>SETUP</c>
<c>454</c>
<c>Session Not Found</c>
<c>all</c>
<c>455</c>
<c>Method Not Valid In This State</c>
<c>all</c>
<c>456</c>
<c>Header Field Not Valid</c>
<c>all</c>
<c>457</c>
<c>Invalid Range</c>
<c>PLAY, PAUSE</c>
<c>458</c>
<c>Parameter Is Read-Only</c>
<c>SET_PARAMETER</c>
<c>459</c>
<c>Aggregate Operation Not Allowed</c>
<c>all</c>
<c>460</c>
<c>Only Aggregate Operation Allowed</c>
<c>all</c>
<c>461</c>
<c>Unsupported Transport</c>
<c>all</c>
<c>462</c>
<c>Destination Unreachable</c>
<c>all</c>
<c>463</c>
<c>Destination Prohibited</c>
<c>SETUP</c>
<c>464</c>
<c>Data Transport Not Ready Yet</c>
<c>PLAY</c>
<c>465</c>
<c>Notification Reason Unknown</c>
<c>PLAY_NOTIFY</c>
<c>466</c>
<c>Key Management Error</c>
<c>all</c>
<c>470</c>
<c>Connection Authorization Required</c>
<c>all</c>
<c>471</c>
<c>Connection Credentials not accepted</c>
<c>all</c>
<c>472</c>
<c>Failure to establish secure connection</c>
<c>all</c>
<c/>
<c/>
<c/>
<c>500</c>
<c>Internal Server Error</c>
<c>all</c>
<c>501</c>
<c>Not Implemented</c>
<c>all</c>
<c>502</c>
<c>Bad Gateway</c>
<c>all</c>
<c>503</c>
<c>Service Unavailable</c>
<c>all</c>
<c>504</c>
<c>Gateway Timeout</c>
<c>all</c>
<c>505</c>
<c>RTSP Version Not Supported</c>
<c>all</c>
<c>551</c>
<c>Option Not Supported</c>
<c>all</c>
<c>553</c>
<c>Proxy Unavailable</c>
<c>all</c>
</texttable>
</section>
</section>
<section anchor="sec_response-header" title="Response Headers">
<t>The response-headers allow the request recipient to pass additional
information about the response which cannot be placed in the
Status-Line. This header gives information about the server and about
further access to the resource identified by the Request-URI. All
headers currently classified as response-headers are listed in <xref
target="tab_response-header"/>.</t>
<texttable anchor="tab_response-header"
title="The RTSP response headers">
<preamble/>
<ttcol align="left">Header</ttcol>
<ttcol align="left">Defined in Section</ttcol>
<c>Authentication-Info</c>
<c><xref target="sec_Authentication-Info"/></c>
<c>Connection-Credentials</c>
<c><xref target="sec_Connection-Credentials"/></c>
<c>Location</c>
<c><xref target="sec_Location"/></c>
<c>MTag</c>
<c><xref target="sec_MTag"/></c>
<c>Proxy-Authenticate</c>
<c><xref target="sec_Proxy-Authenticate"/></c>
<c>Public</c>
<c><xref target="sec_Public"/></c>
<c>Retry-After</c>
<c><xref target="sec_Retry-After"/></c>
<c>Unsupported</c>
<c><xref target="sec_Unsupported"/></c>
<c>WWW-Authenticate</c>
<c><xref target="sec_WWW-Authenticate"/></c>
</texttable>
<t>Response-header names can be extended reliably only in combination
with a change in the protocol version. However, the usage of
feature-tags in the request allows the responding party to learn the
capability of the receiver of the response. A new or experimental
header MAY be given the semantics of response-header if all parties in
the communication recognize them to be a response-header. Unrecognized
headers in responses are treated as message-body headers and hence
MUST be ignored.</t>
</section>
</section>
<!-- title="Response" -->
<section anchor="sec_entity" title="Message Body">
<t>Request and Response messages MAY transfer a message body, if not
otherwise restricted by the request method or response status code. The
message body consists of the content data itself (see also <xref
target="sec_message-body"/>).</t>
<t>The SET_PARAMETER and GET_PARAMETER requests and responses, and the
DESCRIBE response as defined by this specification MAY have a message
body; the purpose of the message body is defined in each case. All 4xx
and 5xx responses MAY also have a message body to carry additional
response information. Generally a message body MAY be attached to any
RTSP 2.0 request or response, but the content of the message body MAY be
ignored by the receiver. Extensions to this specification can specify
the purpose and content of message bodies, including requiring their
inclusion.</t>
<t>In this section, both sender and recipient refer to either the client
or the server, depending on who sends and who receives the message
body.</t>
<section anchor="sec_message-header" title="Message-Body Header Fields">
<t>Message-body header fields define meta-information about the
content data in the message body. The message-body header fields are
listed in <xref target="tab_message-header-tab"/>.</t>
<texttable anchor="tab_message-header-tab"
title="The RTSP message-body headers">
<preamble/>
<ttcol align="left">Header</ttcol>
<ttcol align="left">Defined in Section</ttcol>
<c>Allow</c>
<c><xref target="sec_Allow"/></c>
<c>Content-Base</c>
<c><xref target="sec_Content-Base"/></c>
<c>Content-Encoding</c>
<c><xref target="sec_Content-Encoding"/></c>
<c>Content-Language</c>
<c><xref target="sec_Content-Language"/></c>
<c>Content-Length</c>
<c><xref target="sec_Content-Length"/></c>
<c>Content-Location</c>
<c><xref target="sec_Content-Location"/></c>
<c>Content-Type</c>
<c><xref target="sec_Content-Type"/></c>
<c>Expires</c>
<c><xref target="sec_Expires"/></c>
<c>Last-Modified</c>
<c><xref target="sec_Last-Modified"/></c>
</texttable>
<t>The extension-header mechanism allows additional message-body
header fields to be defined without changing the protocol, but these
fields cannot be assumed to be recognizable by the recipient.
Unrecognized header fields MUST be ignored by the recipient and
forwarded by proxies.</t>
</section>
<section title="Message Body">
<t>An RTSP message with a message body MUST include the Content-Type
and Content-Length headers. When a message body is included with a
message, the data type of that content data is determined via the
header fields Content-Type and Content-Encoding.</t>
<t>Content-Type specifies the media type of the underlying data.
Content-Encoding may be used to indicate any additional content
codings applied to the data, usually for the purpose of data
compression, that are a property of the requested resource. There is
no default encoding.</t>
<t>The Content-Length of a message is the length of the content,
measured in octets.</t>
</section>
<section anchor="sec_Message-Body-Format-Negotiation"
title="Message Body Format Negotiation">
<t>The content format of the message body is provided using the <xref
target="sec_Content-Type">Content-Type header</xref>. To enable the
responder of a request to determine which media type it should use,
the requestor may include the <xref target="sec_Accept">Accept
header</xref> in a request to identify supported media types or media
type ranges suitable to the response. In case the responder is not
supporting any of the specified formats, then the request response
will be a 406 (Not Acceptable) error code.</t>
<t>The media types that may be used on requests with message bodies
need to be determined through the use of feature-tags, specification
requirement or trial and error. Trial and error works because when the
responder does not support the media type of the message body it will
respond with a 415 (Unsupported Media Type).</t>
<t>The formats supported and their negotiation is done individually on
a per method and direction (request or response body) direction.
Requirements on supporting particular media types for use as message
bodies in requests and response SHALL also be specified on per method
and direction basis.</t>
</section>
</section>
<!-- title="Entity -->
<section anchor="sec_connections" title="Connections">
<t>RTSP Messages are transferred between RTSP agents and proxies using a
transport connection. This transport connection uses TCP or TCP/TLS.
This transport connection is referred to as the connection or possibly
RTSP connection within this document.</t>
<t>RTSP requests can be transmitted using the two different connection
scenarios listed below: <list hangIndent="3" style="symbols">
<t>persistent - a transport connection is used for several
request/response transactions;</t>
<t>transient - a transport connection is used for a single
request/response transaction.</t>
</list></t>
<t>RFC 2326 attempted to specify an optional mechanism for transmitting
RTSP messages in connectionless mode over a transport protocol such as
UDP. However, it was not specified in sufficient detail to allow for
interoperable implementations. In an attempt to reduce complexity and
scope, and due to lack of interest, RTSP 2.0 does not attempt to define
a mechanism for supporting RTSP over UDP or other connectionless
transport protocols. A side-effect of this is that RTSP requests MUST
NOT be sent to multicast groups since no connection can be established
with a specific receiver in multicast environments.</t>
<t>Certain RTSP headers, such as the CSeq header (<xref
target="sec_CSeq"/>), which may appear to be relevant only to
connectionless transport scenarios are still retained and MUST be
implemented according to the specification. In the case of CSeq, it is
quite useful for matching responses to requests if the requests are
pipelined (see <xref target="Pipelining"/>). It is also useful in
proxies for keeping track of the different requests when aggregating
several client requests on a single TCP connection.</t>
<section title="Reliability and Acknowledgements">
<t>Since RTSP messages are transmitted using reliable transport
protocols, they MUST NOT be retransmitted at the RTSP protocol level.
Instead, the implementation must rely on the underlying transport to
provide reliability. The RTSP implementation may use any indication of
reception acknowledgment of the message from the underlying transport
protocols to optimize the RTSP behavior.</t>
<t><list style="hanging">
<t>If both the underlying reliable transport such as TCP and the
RTSP application retransmit requests, each packet loss or message
loss may result in two retransmissions. The receiver typically
cannot take advantage of the application-layer retransmission
since the transport stack will not deliver the application-layer
retransmission before the first attempt has reached the receiver.
If the packet loss is caused by congestion, multiple
retransmissions at different layers will exacerbate the
congestion.</t>
</list></t>
<t>Lack of acknowledgment of an RTSP request should be handled within
the constraints of the connection timeout considerations described
below (<xref target="sec_connection-timeout"/>).</t>
</section>
<section anchor="sec_connections-usage" title="Using Connections">
<t>A TCP transport can be used for both persistent connections (for
several message exchanges) and transient connections (for a single
message exchange). Implementations of this specification MUST support
RTSP over TCP. The scheme of the RTSP URI (<xref target="sec_url"/>)
indicates the default port that the server will listen on if the port
is not explicitly given.</t>
<t>In addition to the registered default ports, i.e., 554 (rtsp) and
322 (rtsps), there is an alternative port 8554 registered. This port
may provide some benefits from non-registered ports if a RTSP server
is unable to use the default ports. The benefits may include
pre-configured security policies as well as classifiers in network
monitoring tools.</t>
<t>A RTSP client opening a TCP connection for accessing a particular
resource as identified by a URI uses the IP address and port derived
from the host and port parts of the URI. The IP address is either the
explicit address provided in the URI or any of the addresses provided
when performing A and AAAA record DNS lookups of the host name in the
URI.</t>
<t>A server MUST handle both persistent and transient connections.</t>
<t><list style="hanging">
<t>Transient connections facilitate mechanisms for fault
tolerance. They also allow for application layer mobility. A
server and client pair that support transient connections can
survive the loss of a TCP connection; e.g., due to a NAT timeout.
When the client has discovered that the TCP connection has been
lost, it can set up a new one when there is need to communicate
again.</t>
</list></t>
<t>A persistent connection is RECOMMENDED to be used for all
transactions between the server and client, including messages for
multiple RTSP sessions. However, a persistent connection MAY be closed
after a few message exchanges. For example, a client may use a
persistent connection for the initial SETUP and PLAY message exchanges
in a session and then close the connection. Later, when the client
wishes to send a new request, such as a PAUSE for the session, a new
connection would be opened. This connection may either be transient or
persistent.</t>
<t>An RTSP agent MAY use one connection to handle multiple RTSP
sessions on the same server. The RTSP agent SHALL NOT use more than
one connection per RTSP session at any given point.</t>
<t><list style="hanging">
<t>Using a single connection for multiple RTSP session saves
complexity by enabling the server to maintain less state about its
connection resources on the server. Not using more than one
connection at a time for a particular RTSP session avoids wasting
connection resources and allows the server to track only the most
recently used client to server connection for each RTSP session as
being the currently valid server to client connection.</t>
</list></t>
<t>RTSP allows a server to send requests to a client. However, this
can be supported only if a client establishes a persistent connection
with the server. In cases where a persistent connection does not exist
between a server and its client, due to the lack of a signaling
channel the server may be forced to silently discard RTSP messages,
and may even drop an RTSP session without notifying the client. An
example of such a case is when the server desires to send a REDIRECT
request for an RTSP session to the client but is not able to do so
because it cannot reach the client. A server that attempts to send a
request to a client that has no connection currently to the server
SHALL discard the request directly. <list style="hanging">
<t>Without a persistent connection between the client and the
server, the media server has no reliable way of reaching the
client. Because the likely failure of server to client established
connections the server will not even attempt establishing any
connection.</t>
<t>Queuing of server to client requests has been considered.
However a security issue exists as to how it might be possible to
authorize a client establishing a new connection as being a
legitimate receiver of a request related to a particular RTSP
session without the client first issuing requests related to the
request. Thus, it would be likely to make any such requests even
more delayed and less useful.</t>
</list></t>
<t>The sending of client and server requests can be asynchronous
events. To avoid deadlock situations both client and server MUST be
able to send and receive requests simultaneously. As an RTSP response
may be queued up for transmission, reception or processing behind the
peer RTSP agent's own requests, all RTSP agents are required to have a
certain capability of handling outstanding messages. A potential issue
is that outstanding requests may timeout despite them being processed
by the peer due to the response being caught in the queue behind a
number of requests that the RTSP agent is processing but that take
some time to complete. To avoid this problem an RTSP agent is
recommended to buffer incoming messages locally so that any response
messages can be processed immediately upon reception. If responses are
separated from requests and directly forwarded for processing, not
only can the result be used immediately, the state associated with
that outstanding request can also be released. However, buffering a
number of requests on the receiving RTSP agent consumes resources and
enables a resource exhaustion attack on the agent. Therefore this
buffer should be limited so that an unreasonable number of requests or
total message size is not allowed to consume the receiving agent's
resources. In most APIs, having the receiving agent stop reading from
the TCP socket will result in TCP's window being clamped. Thus forcing
the buffering onto the sending agent when the load is larger than
expected. However, as both RTSP message sizes and frequency may be
changed in the future by protocol extensions, an agent should be
careful against taking harsher measurements against a potential
attack. When under attack an RTSP agent can close TCP connections and
release state associated with that TCP connection.</t>
<t>To provide some guidance on what is reasonable the following
guidelines are given. It is RECOMMENDED that: <list style="symbols">
<t>an RTSP agent should not have more than 10 outstanding requests
per RTSP session;</t>
<t>an RTSP agent should not have more than 10 outstanding requests
that are not related to an RTSP session or that are requesting to
create an RTSP session.</t>
</list></t>
<t>In light of the above, it is RECOMMENDED that clients use
persistent connections whenever possible. A client that supports
persistent connections MAY "pipeline" its requests (see <xref
target="Pipelining"/>).</t>
<t>RTSP Agents can send requests to multiple different destinations,
either servers or client contexts over the same connection to a proxy.
Then the proxy forks the message to the different destinations over
proxy to agent connections. In these cases when multiple requests are
outstanding the requesting agent MUST be ready to receive the
responses out of order compared to the order they where sent on the
connection. The order between multiple messages for each destination
will be maintained, however, the order between response from different
destinations can be different.<list style="empty">
<t>The reason for this is to avoid a head-of-line blocking
sitauation. In a sequence of requests an early outstanding request
may take time to be processed at one destination. Simultaneously,
a response from any other destination that was later in the
sequence of requests, may have arrived at the proxy. Thus allowing
out-of-order responses avoids forcing the proxy to buffer this
response and instead deliver it as soon as possible. Note, this
will not affect the order in which the messages sent to each
separate destination were processed at request destination.</t>
</list></t>
<t>This scenario can occur in two cases involving proxies. The first
is a client issuing requests for sessions on different servers using a
common client to proxy connection. The second is for server to client
requests, like REDIRECT being sent by the server over a common
transport connection the proxy created for its different connecting
clients.</t>
</section>
<section title="Closing Connections">
<t>The client MAY close a connection at any point when no outstanding
request/response transactions exist for any RTSP session being managed
through the connection. The server, however, SHOULD NOT close a
connection until all RTSP sessions being managed through the
connection have been timed out (<xref target="sec_Session"/>). A
server SHOULD NOT close a connection immediately after responding to a
session-level TEARDOWN request for the last RTSP session being
controlled through the connection. Instead, the server should wait for
a reasonable amount of time for the client to receive and act upon the
TEARDOWN response, and initiate the connection closing. The server
SHOULD wait at least 10 seconds after sending the TEARDOWN response
before closing the connection.</t>
<t><list style="hanging">
<t>This is to ensure that the client has time to issue a SETUP for
a new session on the existing connection after having torn the
last one down. 10 seconds should give the client ample opportunity
to get its message to the server.</t>
</list></t>
<t>A server SHOULD NOT close the connection directly as a result of
responding to a request with an error code.</t>
<t><list style="hanging">
<t>Certain error responses such as "460 Only Aggregate Operation
Allowed" (<xref target="sec_error460"/>) are used for negotiating
capabilities of a server with respect to content or other factors.
In such cases, it is inefficient for the server to close a
connection on an error response. Also, such behavior would prevent
implementation of advanced/special types of requests or result in
extra overhead for the client when testing for new features. On
the other hand, keeping connections open after sending an error
response poses a Denial of Service security risk (<xref
target="sec_security"/>).</t>
</list></t>
<t>The server MAY close a connection if it receives an incomplete
message and if the message is not completed within a reasonable amount
of time. It is RECOMMENDED that the server waits at least 10 seconds
for the completion of a message or for the next part of the message to
arrive (which is an indication that the transport and the client are
still alive). Servers believing they are under attack or otherwise
starved for resources during that event MAY consider using a shorter
timeout.</t>
<t>If a server closes a connection while the client is attempting to
send a new request, the client will have to close its current
connection, establish a new connection and send its request over the
new connection.</t>
<t>An RTSP message SHOULD NOT be terminated by closing the connection.
Such a message MAY be considered to be incomplete by the receiver and
discarded. An RTSP message is properly terminated as defined in <xref
target="sec_message"/>.</t>
</section>
<section anchor="sec_connection-timeout"
title="Timing Out Connections and RTSP Messages">
<t>Receivers of a request (responder) SHOULD respond to requests in a
timely manner even when a reliable transport such as TCP is used.
Similarly, the sender of a request (requester) SHOULD wait for a
sufficient time for a response before concluding that the responder
will not be acting upon its request.</t>
<t>A responder SHOULD respond to all requests within 5 seconds. If the
responder recognizes that processing of a request will take longer
than 5 seconds, it SHOULD send a 100 (Continue) response as soon as
possible. It SHOULD continue sending a 100 response every 5 seconds
thereafter until it is ready to send the final response to the
requester. After sending a 100 response, the receiver MUST send a
final response indicating the success or failure of the request.</t>
<t>A requester SHOULD wait at least 10 seconds for a response before
concluding that the responder will not be responding to its request.
After receiving a 100 response, the requester SHOULD continue waiting
for further responses. If more than 10 seconds elapses without
receiving any response, the requester MAY assume that the responder is
unresponsive and abort the connection by closing the TCP
connection.</t>
<t>In cases multiple RTSP sessions share the same transport
connection, abandoning a request and closing the connection may have
significant impact on those other sessions. First of all, other RTSP
requests may have become queued up due to the request taking long
time. Secondly also those sessions loose the possibility to receive
server to client requests. To mitigate that situation the RTSP agent
SHOULD establish a new connection and send any queued up and
non-responded requests on this new connection. Secondly, to ensure
that the RTSP server knows which connection that is valid for a
particular RTSP session, the RTSP agent SHOULD send a keep-alive
request, if no other request will be sent immediately for that RTSP
session, for each RTSP session on the old connection. The keep-alive
request will normally be a GET_PARAMETER with a session header to
inform the server that this agent cares about this RTSP session.</t>
<t>A requester SHOULD wait longer than 10 seconds for a response if it
is experiencing significant transport delays on its connection to the
responder. The requester is capable of determining the round trip time
(RTT) of the request/response cycle using the Timestamp header (<xref
target="sec_Timestamp"/>) in any RTSP request.<list style="hanging">
<t>10 seconds was chosen for the following reasons. It gives TCP
time to perform a couple of retransmissions, even if operating on
default values. It is short enough that users may not abandon the
process themselves. However, it should be noted that 10 seconds
can be aggressive on certain type of networks. The 5 seconds value
for 1xx messages is half the timeout giving a reasonable chance of
successful delivery before timeout happens on the requester
side.</t>
</list></t>
</section>
<section anchor="sec_liveness" title="Showing Liveness">
<t>The mechanisms for showing liveness of the client is, any RTSP
request with a Session header, if RTP & RTCP is used an RTCP
message, or through any other used media protocol capable of
indicating liveness of the RTSP client. It is RECOMMENDED that a
client does not wait to the last second of the timeout before trying
to send a liveness message. The RTSP message may be lost or when using
reliable protocols, such as TCP, the message may take some time to
arrive safely at the receiver. To show liveness between RTSP requests
being issued to accomplish other things, the following mechanisms can
be used, in descending order of preference: <list hangIndent="6"
style="hanging">
<t hangText="RTCP:">If RTP is used for media transport RTCP SHOULD
be used. If RTCP is used to report transport statistics, it will
necessarily also function as a keep-alive. The server can
determine the client by network address and port together with the
fact that the client is reporting on the server's RTP sender
sources (SSRCs). A downside of using RTCP is that it only gives
statistical guarantees of reaching the server. However, the
probability of a false client timeout is so low that it can be
ignored in most cases. For example, assume a session with 60
seconds timeout and enough bitrate assigned to RTCP messages to
send a message from client to server on average every 5 seconds.
That client has, for a network with 5% packet loss, a probability
of failing to confirm liveness within the timeout interval for
that session of 2.4*E-16. Sessions with shorter timeouts, or much
higher packet loss, or small RTCP bandwidths SHOULD also implement
one or more of the mechanisms below.</t>
<t hangText="SET_PARAMETER:">When using SET_PARAMETER for
keep-alive, a body SHOULD NOT be included. This method is the
RECOMMENDED RTSP method to use for a request intended only to
perform keep-alive. Support of SET_PARAMETER is mandatory for RTSP
Servers to ensure clients can use this method.</t>
<t hangText="GET_PARAMETER:">When using GET_PARAMETER for
keep-alive, no body SHOULD be included. Dependent on
implementation support in server. Use OPTIONS method to determine
if there are method support or simply try.</t>
<t hangText="OPTIONS:">This method is also usable, but it causes
the server to perform more unnecessary processing and results in
bigger responses than necessary for the task. The reason is that
the server needs to determine the capabilities associated with the
media resource to correctly populate the Public and Allow
headers.</t>
</list></t>
<t>The timeout parameter of the <xref target="sec_Session">Session
header</xref> MAY be included in a SETUP response, and MUST NOT be
included in requests. The server uses it to indicate to the client how
long the server is prepared to wait between RTSP commands or other
signs of life before closing the session due to lack of activity (see
<xref target="sec_machine"/>). The timeout is measured in seconds,
with a default of 60 seconds. The length of the session timeout MUST
NOT be changed in an established session.</t>
</section>
<section title="Use of IPv6">
<t>Explicit <xref target="RFC2460">IPv6</xref> support was not present
in RTSP 1.0 (RFC 2326). RTSP 2.0 has been updated for explicit IPv6
support. Implementations of RTSP 2.0 MUST understand literal IPv6
addresses in URIs and RTSP headers. Although the general URI format
envisages potential future new versions of the literal IP address,
usage of any such new version would require other modifications to the
RTSP specification (e.g. address fields in the <xref
target="sec_Transport">Transport header</xref>).</t>
</section>
<section anchor="sec-overload-control" title="Overload Control">
<t>Overload in RTSP can occur when servers and proxies have
insufficient resources to complete the processing of a request. An
improper handling of such an overload situation at proxies and servers
can impact the operation of the RTSP deployment, and probably worsen
the situation. RTSP defines the 503 (Service Unavailable) response
(<xref target="sec_error_503"/>) to let servers and proxies notify
requesting proxies and RTSP clients about an overload situation. In
conjunction with the Retry-After header (<xref
target="sec_Retry-After"/>) the server or proxy can indicate the time
after which the requesting entity can send another request to the
proxy or server.</t>
<t>There are two scopes of such 503 answers, one for established RTSP
sessions, where the request resulting in the 503 response as well as
the response carries a Session header identifying the session which is
suffering overload. This response only applies to this particular
session. The other scope is the general RTSP server as identified by
the host in the request URL. Such a 503 answer with any Retry-After
header applies to all non-session specific requests to that server,
including SETUP request intended to create a new RTSP session.</t>
<t>Another scope for overload situation exists, which is the RTSP
proxy. To enable an RTSP proxy to signal that it is overloaded, or
otherwise unavailable and can't handle the request, a 553 response
code has been defined with the meaning "Proxy Unavailable". As with
servers, there is a separation in response scopes between requests
associated with existing RTSP sessions, and requests to create new
sessions or general proxy requests.</t>
<t>Simply implementing and using the 503 (Service Unavailable) and 553
(Proxy Unavailable) is not sufficient for properly handling overload
situations. For instance, a simplistic approach would be to send the
503 response with a Retry-After header set to a fixed value. However,
this can cause the situation where multiple RTSP clients again send
requests to a proxy or server at roughly the same time which may again
cause an overload situation, or if the "old" overload situation is not
yet solved, i.e., the length indicated in the Retry-After header was
too short.</t>
<t>An RTSP server or proxy in an overload situation must select the
value of the Retry-After header carefully and bearing in mind its
current load situation. It is REQUIRED to increase the timeout period
in proportion to the current load on the server, i.e., an increasing
workload should result in an increased length of the indicated
unavailability. It is REQUIRED to not send the same value in the
Retry-After header to all requesting proxies and clients, but to add a
variation to the mean value of the Retry-After header.</t>
<t>A more complex case may arise when a load balancing RTSP proxy is
in use, i.e., where an RTSP proxy is used to select amongst a set of
RTSP servers to handle the requests, or when multiple server addresses
are available for a given server name. The proxy or client may receive
a 503 (Service Unavailable) or 553 (Proxy Unavailable) from one of its
RTSP servers or proxies, or a TCP timeout (if the server is even
unable to handle the request message). The proxy or client simply
retries the other addresses or configured proxies, but may also
receive a 503 (Service Unavailable) or 553 (Proxy Unavailable)
response or TCP timeouts from those addresses. In such a situation,
where none of the RTSP servers/proxies/addresses can handle the
request, the RTSP agent has to wait before it can send any new
requests to the RTSP server. Any additional request to a specific
address MUST be delayed according to the Retry-After headers received.
For addresses where no response was received or TCP timeout occurred,
an initial wait timer SHOULD be set to 5 seconds. That timer MUST be
doubled for each additional failure to connect or receive response
until the value exceeds 30 minutes when the timers mean value may be
set to 30 minutes. It is REQUIRED to not set the same value in the
timer for each scheduling, but instead to add a variation to the mean
value, resulting in picking a random value within the range of 0.5 to
1.5 of the mean value.</t>
</section>
</section>
<!-- title="Connections" -->
<section anchor="sec_capability" title="Capability Handling">
<t>This section describes the available capability handling mechanism
which allows RTSP to be extended. Extensions to this version of the
protocol are basically done in two ways. First, new headers can be
added. Secondly, new methods can be added. The capability handling
mechanism is designed to handle both cases.</t>
<t>When a method is added, the involved parties can use the OPTIONS
method to discover whether it is supported. This is done by issuing an
OPTIONS request to the other party. Depending on the URI it will either
apply in regards to a certain media resource, the whole server in
general, or simply the next hop. The OPTIONS response MUST contain a
Public header which declares all methods supported for the indicated
resource.</t>
<t>It is not necessary to use OPTIONS to discover support of a method,
as the client could simply try the method. If the receiver of the
request does not support the method it will respond with an error code
indicating the method is either not implemented (501) or does not apply
for the resource (405). The choice between the two discovery methods
depends on the requirements of the service.</t>
<t>Feature-tags are defined to handle functionality additions that are
not new methods. Each feature-tag represents a certain block of
functionality. The amount of functionality that a feature-tag represents
can vary significantly. A feature-tag can for example represent the
functionality a single RTSP header provides. Another feature-tag can
represent much more functionality, such as the <xref
target="play.basic">"play.basic" feature-tag</xref> which represents the
minimal media delivery for playback implementation.</t>
<t>Feature-tags are used to determine whether the client, server or
proxy supports the functionality that is necessary to achieve the
desired service. To determine support of a feature-tag, several
different headers can be used, each explained below: <list
hangIndent="6" style="hanging">
<t hangText="Supported:">This header is used to determine the
complete set of functionality that both client and server have in
general and is not dependent on a specific resource. The intended
usage is to determine before one needs to use a functionality that
it is supported. It can be used in any method, but OPTIONS is the
most suitable one as it at the same time determines all methods that
are implemented. When sending a request the requester declares all
its capabilities by including all supported feature-tags. This
results in the receiver learning the requester's feature support.
The receiver then includes its set of features in the response.</t>
<t hangText="Proxy-Supported:">This header is used similarly to the
Supported header, but instead of giving the supported functionality
of the client or server it provides both the requester and the
responder a view of the common functionality supported in general by
all members of the proxy chain between the two supports and not
dependent on the resource. Proxies are required to add this header
whenever the Supported header is present, but proxies may also add
it independently of the requester.</t>
<t hangText="Require:">This header can be included in any request
where the end-point, i.e., the client or server, is required to
understand the feature to correctly perform the request. This can,
for example, be a SETUP request where the server is required to
understand a certain parameter to be able to set up the media
delivery correctly. Ignoring this parameter would not have the
desired effect and is not acceptable. Therefore the end-point
receiving a request containing a Require MUST negatively acknowledge
any feature that it does not understand and not perform the request.
The response in cases where features are not supported are 551
(Option Not Supported). Also the features that are not supported are
given in the Unsupported header in the response.</t>
<t hangText="Proxy-Require:">This header has the same purpose and
workings as Require except that it only applies to proxies and not
the end-point. Features that need to be supported by both proxies
and end-points need to be included in both the Require and
Proxy-Require header.</t>
<t hangText="Unsupported:">This header is used in a 551 error
response, to indicate which features were not supported. Such a
response is only the result of the usage of the Require and/or
Proxy-Require header where one or more features where not supported.
This information allows the requester to make the best of situations
as it knows which features are not supported.</t>
</list></t>
<section anchor="play.basic" title="Feature Tag: play.basic ">
<t>The play.basic feature tag represents an RTSP implementation
offering all the normative RTSP protocol features specified in this
specification. This specification is both a RTSP core specification as
well providing mechanisms for the setup and control of playback of
media. Thus following all normative parts in the main sections (the
ones with numbers), but omitting the appendices (starting with
letters), unless explicitly specified in a main section as being a
required appendix.</t>
<t><list style="empty">
<t>Note: This feature tag does not mandate any media delivery
protocol, such as RTP.</t>
<t>In RTSP 1.0 there was a minimal implementation section.
However, that was not consistent with the rest of the
specification. So rather than making an attempt to explicitly
enumerate the features for play.basic this specification has to be
taken as a whole and the necessary features normatively defined as
being required are included.</t>
</list></t>
</section>
</section>
<section anchor="Pipelining" title="Pipelining Support">
<t>Pipelining is a general method to improve performance of request
response protocols by allowing the requesting agent to have more than
one request outstanding and send them over the same persistent
connection. For RTSP, where the relative order of requests will matter,
it is important to maintain the order of the requests. Because of this,
the responding agent MUST process the incoming requests in their sending
order. The sending order can be determined by the CSeq header and its
sequence number. For TCP the delivery order will be the same between two
agents, as the sending order. The processing of the request MUST also
have been finished before processing the next request from the same
agent. The responses MUST be sent in the order the requests were
processed.</t>
<t>RTSP 2.0 has extended support for pipelining compared to RTSP 1.0.
The major improvement is to allow all requests involved in setting up
and initiating media delivery to be pipelined after each other. This is
accomplished by the utilization of the Pipelined-Requests header (see
<xref target="sec_Pipelined-Requests"/>). This header allows a client to
request that two or more requests are processed in the same RTSP session
context which the first request creates. In other words, a client can
request that two or more media streams are set-up and then played
without needing to wait for a single response. This speeds up the
initial startup time for an RTSP session by at least one RTT.</t>
<t>If a pipelined request builds on the successful completion of one or
more prior requests the requester must verify that all requests were
executed as expected. A common example will be two SETUP requests and a
PLAY request. In case one of the SETUP fails unexpectedly, the PLAY
request can still be successfully executed. However, the resulting
presentation will not be as expected by the requesting client, as only a
single media instead of two will be played. In this case the client can
send a PAUSE request, correct the failing SETUP request and then request
it to be played.</t>
</section>
<section anchor="sec_methods" title="Method Definitions">
<t>The method indicates what is to be performed on the resource
identified by the Request-URI. The method name is case-sensitive. New
methods may be defined in the future. Method names MUST NOT start with a
$ character (decimal 36) and MUST be a token as defined by the ABNF
<xref target="RFC5234"/> in the syntax chapter <xref
target="sec_syntax"/>. The methods are summarized in <xref
target="tab_methods"/>.</t>
<texttable anchor="tab_methods"
title="Overview of RTSP methods, their direction, and what objects (P: presentation, S: stream) they operate on. Further it indicates if a server or a client implementation are required (mandatory), recommended or if it is optional to implement the method.">
<preamble/>
<ttcol align="left">method</ttcol>
<ttcol align="left">direction</ttcol>
<ttcol align="left">object</ttcol>
<ttcol align="left">Server req.</ttcol>
<ttcol align="left">Client req.</ttcol>
<c>DESCRIBE</c>
<c>C -> S</c>
<c>P,S</c>
<c>recommended</c>
<c>recommended</c>
<c>GET_PARAMETER</c>
<c>C -> S</c>
<c>P,S</c>
<c>optional</c>
<c>optional</c>
<c><!--GET_PARAMETER--></c>
<c>S -> C</c>
<c>P,S</c>
<c>optional</c>
<c>optional</c>
<c>OPTIONS</c>
<c>C -> S</c>
<c>P,S</c>
<c>required</c>
<c>required</c>
<c><!--OPTIONS--></c>
<c>S -> C</c>
<c>P,S</c>
<c>optional</c>
<c>optional</c>
<c>PAUSE</c>
<c>C -> S</c>
<c>P,S</c>
<c>required</c>
<c>required</c>
<c>PLAY</c>
<c>C -> S</c>
<c>P,S</c>
<c>required</c>
<c>required</c>
<c>PLAY_NOTIFY</c>
<c>S -> C</c>
<c>P,S</c>
<c>required</c>
<c>required</c>
<c>REDIRECT</c>
<c>S -> C</c>
<c>P,S</c>
<c>optional</c>
<c>required</c>
<c>SETUP</c>
<c>C -> S</c>
<c>S</c>
<c>required</c>
<c>required</c>
<c>SET_PARAMETER</c>
<c>C -> S</c>
<c>P,S</c>
<c>required</c>
<c>optional</c>
<c><!--SET_PARAMETER--></c>
<c>S -> C</c>
<c>P,S</c>
<c>optional</c>
<c>optional</c>
<c>TEARDOWN</c>
<c>C -> S</c>
<c>P,S</c>
<c>required</c>
<c>required</c>
<c><!--TEARDOWN--></c>
<c>S -> C</c>
<c>P</c>
<c>required</c>
<c>required</c>
</texttable>
<t><list style="hanging">
<t>Note on <xref target="tab_methods"/>: GET_PARAMETER is optional.
For example, a fully functional server can be built to deliver media
without any parameters. However, SET_PARAMETER is required, i.e.,
mandatory to implement for the server, this is due to its usage for
keep-alive. PAUSE is required because it is the only way of leaving
the Play state without terminating the whole session.</t>
</list></t>
<t>If an RTSP agent does not support a particular method, it MUST return
501 (Not Implemented) and the requesting RTSP agent, in turn, SHOULD NOT
try this method again for the given agent / resource combination. An
RTSP proxy whose main function is to log or audit and not modify
transport or media handling in any way MAY forward RTSP messages with
unknown methods. Note that the proxy still needs to perform the minimal
required processing, like adding the Via header.</t>
<section anchor="sec_OPTIONS" title="OPTIONS">
<t>The semantics of the RTSP OPTIONS method is similar to that of the
HTTP OPTIONS method described in [H9.2]. In RTSP however, OPTIONS is
bi-directional, in that a client can send the request to a server and
vice versa. A client MUST implement the capability to send an OPTIONS
request and a server or a proxy MUST implement the capability to
respond to an OPTIONS request. In addition to this "MUST implement"
functionality, clients, servers and proxies MAY provide support both
for sending OPTIONS requests and generating responses to the
requests.</t>
<t>An OPTIONS request may be issued at any time. Such a request does
not modify the session state. However, it may prolong the session
lifespan (see below). The URI in an OPTIONS request determines the
scope of the request and the corresponding response. If the
Request-URI refers to a specific media resource on a given host, the
scope is limited to the set of methods supported for that media
resource by the indicated RTSP agent. A Request-URI with only the host
address limits the scope to the specified RTSP agent's general
capabilities without regard to any specific media. If the Request-URI
is an asterisk ("*"), the scope is limited to the general capabilities
of the next hop (i.e., the RTSP agent in direct communication with the
request sender).</t>
<t>Regardless of the scope of the request, the Public header MUST
always be included in the OPTIONS response listing the methods that
are supported by the responding RTSP agent. In addition, if the scope
of the request is limited to a media resource, the Allow header MUST
be included in the response to enumerate the set of methods that are
allowed for that resource unless the set of methods completely matches
the set in the Public header. If the given resource is not available,
the RTSP agent SHOULD return an appropriate response code such as 3rr
or 4xx. The Supported header MAY be included in the request to query
the set of features that are supported by the responding RTSP
agent.</t>
<t>The OPTIONS method can be used to keep an RTSP session alive.
However, this is not the preferred way of session keep-alive
signaling, see <xref target="sec_Session"/>. An OPTIONS request
intended for keeping alive an RTSP session MUST include the Session
header with the associated session identifier. Such a request SHOULD
also use the media or the aggregated control URI as the
Request-URI.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
C->S: OPTIONS rtsp://server.example.com RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
Proxy-Require: gzipped-messages
Supported: play.basic
S->C: RTSP/2.0 200 OK
CSeq: 1
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, OPTIONS
Supported: play.basic, setup.rtp.rtcp.mux, play.scale
Server: PhonyServer/1.1
]]></artwork>
</figure>
<t>Note that some of the feature-tags in Supported and Proxy-Require
are fictitious features.</t>
</section>
<section anchor="sec_DESCRIBE" title="DESCRIBE">
<t>The DESCRIBE method is used to retrieve the description of a
presentation or media object from a server. The Request-URI of the
DESCRIBE request identifies the media resource of interest. The client
MAY include the Accept header in the request to list the description
formats that it understands. The server MUST respond with a
description of the requested resource and return the description in
the message body of the response, if the DESCRIBE method request can
be successfully fulfilled. The DESCRIBE reply-response pair
constitutes the media initialization phase of RTSP.</t>
<t>The DESCRIBE response SHOULD contain all media initialization
information for the resource(s) that it describes. Servers SHOULD NOT
use the DESCRIBE response as a means of media indirection by having
the description point at another server; instead, using the 3rr
responses is RECOMMENDED.</t>
<t><list style="hanging">
<t>By forcing a DESCRIBE response to contain all media
initialization information for the set of streams that it
describes, and discouraging the use of DESCRIBE for media
indirection, any looping problems can be avoided that might have
resulted from other approaches.</t>
</list></t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/2.0
CSeq: 312
User-Agent: PhonyClient/1.2
Accept: application/sdp, application/example
S->C: RTSP/2.0 200 OK
CSeq: 312
Date: Thu, 23 Jan 1997 15:35:06 GMT
Server: PhonyServer/1.1
Content-Base: rtsp://server.example.com/fizzle/foo/
Content-Type: application/sdp
Content-Length: 358
v=0
o=MNobody 2890844526 2890842807 IN IP4 192.0.2.46
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.example.com/lectures/sdp.ps
e=seminar@example.com (Seminar Management)
c=IN IP4 0.0.0.0
a=control:*
t=2873397496 2873404696
m=audio 3456 RTP/AVP 0
a=control:audio
m=video 2232 RTP/AVP 31
a=control:video
]]></artwork>
</figure>
<t>Media initialization is a requirement for any RTSP-based system,
but the RTSP specification does not dictate that this is required to
be done via the DESCRIBE method. There are three ways that an RTSP
client may receive initialization information: <list hangIndent="3"
style="symbols">
<t>via an RTSP DESCRIBE request</t>
<t>via some other protocol (HTTP, email attachment, etc.)</t>
<t>via some form of user interface</t>
</list></t>
<t>If a client obtains a valid description from an alternate source,
the client MAY use this description for initialization purposes
without issuing a DESCRIBE request for the same media. The client
should use any MTag to either validate the presentation description or
make the session establishment conditional on being valid.</t>
<t>It is RECOMMENDED that minimal servers support the DESCRIBE method,
and highly recommended that minimal clients support the ability to act
as "helper applications" that accept a media initialization file from
a user interface, and/or other means that are appropriate to the
operating environment of the clients.</t>
</section>
<section anchor="sec_SETUP" title="SETUP">
<t>The description below uses the following states in a protocol state
machine that is related to a specific session when that session has
been created. The state transitions are driven by protocol
interactions. For additional information about the state machine see
<xref target="sec_machine"/>.</t>
<t><list hangIndent="6" style="hanging">
<t hangText="Init:">Initial state: no session exists.</t>
<t hangText="Ready:">Session is ready to start playing.</t>
<t hangText="Play:">Session is playing, i.e., sending media stream
data in the direction S->C.</t>
</list></t>
<t>The SETUP request for a URI specifies the transport mechanism to be
used for the streamed media. The SETUP method may be used in two
different cases; Create an RTSP session and change the transport
parameters of already set up media stream(s). SETUP can be used in all
three states; Init, and Ready, for both purposes and in PLAY to change
the transport parameters. There is also a third possible usage for the
SETUP method which is not specified in this memo: adding a media to a
session. Using SETUP to add media to an existing session, when the
session is in Play state, is unspecified.</t>
<t>The Transport header, see <xref target="sec_Transport"/>, specifies
the media transport parameters acceptable to the client for data
transmission; the response will contain the transport parameters
selected by the server. This allows the client to enumerate in
descending order of preference the transport mechanisms and parameters
acceptable to it, while the server can select the most appropriate. It
is expected that the session description format used will enable the
client to select a limited number of possible configurations that are
offered to the server to choose from. All transport related parameters
SHALL be included in the Transport header; the use of other headers
for this purpose is NOT RECOMMENDED due to middleboxes, such as
firewalls or NATs.</t>
<t>For the benefit of any intervening firewalls, a client MUST
indicate the known transport parameters, even if it has no influence
over these parameters, for example, where the server advertises a
fixed multicast address as destination.</t>
<t><list style="hanging">
<t>Since SETUP includes all transport initialization information,
firewalls and other intermediate network devices (which need this
information) are spared the more arduous task of parsing the
DESCRIBE response, which has been reserved for media
initialization.</t>
</list></t>
<t>The client MUST include the Accept-Ranges header in the request
indicating all supported unit formats in the Range header. This allows
the server to know which formats it may use in future session related
responses, such as a PLAY response without any range in the request.
If the client does not support a time format necessary for the
presentation, the server MUST respond using 456 (Header Field Not
Valid for Resource) and include the Accept-Ranges header with the
range unit formats supported for the resource.</t>
<t>In a SETUP response the server MUST include the Accept-Ranges
header (see <xref target="sec_Accept-Ranges"/>) to indicate which time
formats are acceptable to use for this media resource.</t>
<t>The SETUP response 200 OK MUST include the Media-Properties header
(see <xref target="sec_Media-Properties"/> ). The combination of the
parameters of the Media-Properties header indicates the nature of the
content present in the session (see also <xref
target="sec_Media-Properties-Intro"/>). For example, a live stream
with time shifting is indicated by<list style="symbols">
<t>Random Access set to Random-Access,</t>
<t>Content Modifications set to Time Progressing,</t>
<t>Retention set to Time-Duration (with specific recording window
time value).</t>
</list></t>
<t>The SETUP response 200 OK MUST include the Media-Range header (see
<xref target="sec_Media-Range"/>) if the media is
Time-Progressing.</t>
<t>A basic example for SETUP:</t>
<figure>
<artwork><![CDATA[
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0
CSeq: 302
Transport: RTP/AVP;unicast;dest_addr=":4588"/":4589",
RTP/AVP/TCP;unicast;interleaved=0-1
Accept-Ranges: npt, clock
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 302
Date: Thu, 23 Jan 1997 15:35:06 GMT
Server: PhonyServer/1.1
Session: 47112344;timeout=60
Transport: RTP/AVP;unicast;dest_addr="192.0.2.53:4588"/
"192.0.2.53:4589"; src_addr="198.51.100.241:6256"/
"198.51.100.241:6257"; ssrc=2A3F93ED
Accept-Ranges: npt
Media-Properties: Random-Access=3.2, Time-Progressing,
Time-Duration=3600.0
Media-Range: npt=0-2893.23]]></artwork>
</figure>
<t>In the above example the client wants to create an RTSP session
containing the media resource "rtsp://example.com/foo/bar/baz.rm". The
transport parameters acceptable to the client are either RTP/AVP/UDP
(UDP per default) to be received on client port 4588 and 4589 at the
address the RTSP setup connection comes from or RTP/AVP interleaved on
the RTSP control channel. The server selects the RTP/AVP/UDP transport
and adds the address and ports it will send and receive RTP and RTCP
from, and the RTP SSRC that will be used by the server.</t>
<t>The server MUST generate a session identifier in response to a
successful SETUP request, unless a SETUP request to a server includes
a session identifier or a Pipelined-Requests header referencing an
existing session context, in which case the server MUST bundle this
SETUP request into the existing session (aggregated session) or return
error 459 (Aggregate Operation Not Allowed) (see <xref
target="sec_error459"/>). An Aggregate control URI MUST be used to
control an aggregated session. This URI MUST be different from the
stream control URIs of the individual media streams included in the
aggregate (see <xref target="sec_aggregated_sessions"/> for aggregated
sessions and for the particular URIs see <xref
target="sec_sdp-control-url"/>). The Aggregate control URI is to be
specified by the session description if the server supports aggregated
control and aggregated control is desired for the session. However,
even if aggregated control is offered the client MAY chose to not set
up the session in aggregated control. If an Aggregate control URI is
not specified in the session description, it is normally an indication
that non-aggregated control should be used. The SETUP of media streams
in an aggregate which has not been given an aggregated control URI is
unspecified.</t>
<t><list style="hanging">
<t>While the session ID sometimes carries enough information for
aggregate control of a session, the Aggregate control URI is still
important for some methods such as SET_PARAMETER where the control
URI enables the resource in question to be easily identified. The
Aggregate control URI is also useful for proxies, enabling them to
route the request to the appropriate server, and for logging,
where it is useful to note the actual resource that a request was
operating on.</t>
</list></t>
<t>A session will exist until it is either removed by a TEARDOWN
request or is timed-out by the server. The server MAY remove a session
that has not demonstrated liveness signs from the client(s) within a
certain timeout period. The default timeout value is 60 seconds; the
server MAY set this to a different value and indicate so in the
timeout field of the Session header in the SETUP response. For further
discussion see <xref target="sec_Session"/>. Signs of liveness for an
RTSP session are: <list hangIndent="3" style="symbols">
<t>Any RTSP request from a client which includes a Session header
with that session's ID.</t>
<t>If RTP is used as a transport for the underlying media streams,
an RTCP sender or receiver report from the client(s) for any of
the media streams in that RTSP session. RTCP Sender Reports may
for example be received in sessions where the server is invited
into a conference session and is valid for keep-alive.</t>
</list></t>
<t>If a SETUP request on a session fails for any reason, the session
state, as well as transport and other parameters for associated
streams MUST remain unchanged from their values as if the SETUP
request had never been received by the server.</t>
<section title="Changing Transport Parameters">
<t>A client MAY issue a SETUP request for a stream that is already
set up or playing in the session to change transport parameters,
which a server MAY allow. If it does not allow changing of
parameters, it MUST respond with error 455 (Method Not Valid In This
State). The reasons to support changing transport parameters include
allowing application layer mobility and flexibility to utilize the
best available transport as it becomes available. If a client
receives a 455 when trying to change transport parameters while the
server is in Play state, it MAY try to put the server in Ready state
using PAUSE, before issuing the SETUP request again. If that also
fails the changing of transport parameters will require that the
client performs a TEARDOWN of the affected media and then to set it
up again. For an aggregated session avoiding tearing down all the
media at the same time will avoid the creation of a new session.</t>
<t>All transport parameters MAY be changed. However, the primary
usage expected is to either change the transport protocol
completely, like switching from Interleaved TCP mode to UDP or vice
versa, or to change the delivery address.</t>
<t>In a SETUP response for a request to change the transport
parameters while in Play state, the server MUST include the Range to
indicate at what point the new transport parameters will be used.
Further, if RTP is used for delivery, the server MUST also include
the RTP-Info header to indicate at what timestamp and RTP sequence
number the change will take place. If both RTP-Info and Range are
included in the response the "rtp_time" parameter and start point in
the Range header MUST be for the corresponding time, i.e., be used
in the same way as for PLAY to ensure the correct synchronization
information is available.</t>
<t>If the transport parameters change while in Play state results in
a change of synchronization related information, for example
changing RTP SSRC, the server MUST provide in the SETUP response the
necessary synchronization information. However, the server is
RECOMMENDED to avoid changing the synchronization information if
possible.</t>
</section>
</section>
<section anchor="sec_PLAY" title="PLAY">
<t>This section describes the usage of the PLAY method in general, for
aggregated sessions, and in different usage scenarios.</t>
<section anchor="sec_PLAY_general" title="General Usage">
<t>The PLAY method tells the server to start sending data via the
mechanism specified in SETUP and which part of the media should be
played out. PLAY requests are valid when the session is in Ready or
Play states. A PLAY request MUST include a Session header to
indicate which session the request applies to.</t>
<t>Upon receipt of the PLAY request, the server MUST position the
normal play time to the beginning of the range specified in the
received Range header, within the limits of the media resource and
in accordance with the <xref target="sec_Seek-Style">Seek-Style
header</xref> and deliver stream data until the end of the range if
given, until a new PLAY request is received, or until the end of the
media is reached. If no Range header is present in the PLAY request
the server SHALL play from current pause point until the end of
media. The pause point defaults at session start to the beginning of
the media. For media that is time-progressing and has no retention,
the pause point will always be set equal to NPT "now", i.e., the
current delivery point. The pause point may also be set to a
particular point in the media by the PAUSE method, see <xref
target="sec_PAUSE"/>. The pause point for media that is currently
playing is equal to the current media position. For time-progressing
media with time-limited retention, if the pause point represents a
position that is older than what is retained by the server, the
pause point will be moved to the oldest retained.</t>
<t>What range values are valid depends on the type of content. For
content that isn't time progressing the range value is valid if the
given range is part of any media within the aggregate. In other
words the valid media range for the aggregate is the union of all of
the media components in the aggregate. If a given range value points
outside of the media, the response MUST be the 457 (Invalid Range)
error code and include the <xref target="sec_Media-Range">
Media-Range header</xref> with the valid range for the media. Except
for time progressing content where the client requests a start point
prior to what is retained, the start point is adjusted to the oldest
retained content. For a start point that is beyond the media front
edge, i.e., beyond the current value for "now", the server SHALL
adjust the start value to the current front edge. The Range header's
stop point value may point beyond the current media edge. In that
case, the server SHALL deliver media from the requested (and
possibly adjusted) start point until the provided stop point, or the
end of the media is reached prior to the specified stop point.
Please note that if one simply wants to play from a particular start
point until the end of media using a Range header with an implicit
stop point is RECOMMENDED.</t>
<t>If a client requests to start playing at the end of media, either
explicitly with a Range header or implicitly with a pause point that
is at the end of media, a 457 (Invalid Range) error MUST be sent and
include the <xref target="sec_Media-Range">Media-Range
header</xref>. It is specified below that the Range header also must
be included in the response and that it will carry the pause point
in the media, in the case of the session being in Ready State. Note
that this also applies if the pause point or requested start point
is at the beginning of the media and a <xref
target="sec_Scale">Scale header</xref> is included with a negative
value (playing backwards).</t>
<t>For media with random access properties a client may express its
preference on which policy for start point selection the server
shall use. This is done by including the <xref
target="sec_Seek-Style">Seek-Style header</xref> in the PLAY
request. The Seek-Style applied will effect the content of the Range
header as it will be adjusted to indicate from what point the media
actually is delivered.</t>
<t>A client desiring to play the media from the beginning MUST send
a PLAY request with a Range header pointing at the beginning, e.g.,
"npt=0-". If a PLAY request is received without a Range header and
media delivery has stopped at the end, the server SHOULD respond
with a 457 "Invalid Range" error response. In that response, the
current pause point MUST be included in a Range header.</t>
<t>All range specifiers in this specification allow for ranges with
an implicit start point (e.g., "npt=-30"). When used in a PLAY
request, the server treats this as a request to start or resume
delivery from the current pause point, ending at the end time
specified in the Range header. If the pause point is located later
than the given end value, a 457 (Invalid Range) response MUST be
given.</t>
<t>The example below will play seconds 10 through 25. It also
requests the server to deliver media from the first Random Access
Point prior to the indicated start point.</t>
<figure>
<artwork><![CDATA[
C->S: PLAY rtsp://audio.example.com/audio RTSP/2.0
CSeq: 835
Session: 12345678
Range: npt=10-25
Seek-Style: RAP
User-Agent: PhonyClient/1.2]]></artwork>
</figure>
<t>Servers MUST include a "Range" header in any PLAY response, even
if no Range header was present in the request. The response MUST use
the same format as the request's range header contained. If no Range
header was in the request, the format used in any previous PLAY
request within the session SHOULD be used. If no format has been
indicated in a previous request the server MAY use any time format
supported by the media and indicated in the Accept-Ranges header in
the SETUP request. It is RECOMMENDED that NPT is used if supported
by the media.</t>
<t>For any error response to a PLAY request, the server's response
depends on the current session state. If the session is in Ready
state, the current pause-point is returned using Range header with
the pause point as the explicit start-point and an implicit
stop-point. For time-progressing content where the pause-point moves
with real-time due to limited retention, the current pause point is
returned. For sessions in Play state, the current playout point and
the remaining parts of the range request is returned. For any media
with retention longer than 0 seconds the currently valid Media-Range
header SHALL also be included in the response.</t>
<t>A PLAY response MAY include a header carrying synchronization
information. As the information necessary is dependent on the media
transport format, further rules specifying the header and its usage
are needed. For RTP the RTP-Info header is specified, see <xref
target="sec_RTP-Info"/>, and used in the following example.</t>
<t>Here is a simple example for a single audio stream where the
client requests the media starting from 3.52 seconds and to the end.
The server sends a 200 OK response with the actual play time which
is 10 ms prior (3.51) and the RTP-Info header that contains the
necessary parameters for the RTP stack.</t>
<figure>
<artwork><![CDATA[
C->S: PLAY rtsp://example.com/audio RTSP/2.0
CSeq: 836
Session: 12345678
Range: npt=3.52-
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 836
Date: Thu, 23 Jan 1997 15:35:06 GMT
Server: PhonyServer/1.0
Range: npt=3.51-324.39
Seek-Style: First-Prior
RTP-Info:url="rtsp://example.com/audio"
ssrc=0D12F123:seq=14783;rtptime=2345962545
S->C: RTP Packet TS=2345962545 => NPT=3.51
Media duration=0.16 seconds]]></artwork>
</figure>
<t>The server replies with the actual start point that will be
delivered. This may differ from the requested range if alignment of
the requested range to valid frame boundaries is required for the
media source. Note that some media streams in an aggregate may need
to be delivered from even earlier points. Also, some media formats
have a very long duration per individual data unit, therefore it
might be necessary for the client to parse the data unit, and select
where to start. The server SHALL also indicate which policy it uses
for selecting the actual start point by including a Seek-Style
header.</t>
<t>In the following example the client receives the first media
packet that stretches all the way up and past the requested
playtime. Thus, it is the client's decision whether to render to the
user the time between 3.52 and 7.05, or to skip it. In most cases it
is probably most suitable not to render that time period.</t>
<figure>
<artwork><![CDATA[
C->S: PLAY rtsp://example.com/audio RTSP/2.0
CSeq: 836
Session: 12345678
Range: npt=7.05-
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 836
Date: Thu, 23 Jan 1997 15:35:06 GMT
Server: PhonyServer/1.0
Range: npt=3.52-
Seek-Style: First-Prior
RTP-Info:url="rtsp://example.com/audio"
ssrc=0D12F123:seq=14783;rtptime=2345962545
S->C: RTP Packet TS=2345962545 => NPT=3.52
Duration=4.15 seconds]]></artwork>
</figure>
<t>After playing the desired range, the presentation does NOT change
to the Ready state, media delivery simply stops. A PAUSE request
MUST be issued to make the stream enter the Ready state. A PLAY
request while the stream is still in the Play state is legal, and
can be issued without an intervening PAUSE request. Such a request
MUST replace the current PLAY action with the new one requested,
i.e., being handled in the same way as if as the request was
received in Ready state. In the case that the range in Range header
has an implicit start time ("-endtime"), the server MUST continue to
play from where it currently was until the specified end point. This
is useful to change the end to at another point than in the previous
request.</t>
<t>The following example plays the whole presentation starting at
SMPTE time code 0:10:20 until the end of the clip. Note: The
RTP-Info headers has been broken into several lines, where following
lines start with whitespace as allowed by the syntax.</t>
<figure>
<artwork><![CDATA[
C->S: PLAY rtsp://audio.example.com/twister.en RTSP/2.0
CSeq: 833
Session: 12345678
Range: smpte=0:10:20-
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 833
Date: Thu, 23 Jan 1997 15:35:06 GMT
Session: 12345678
Server: PhonyServer/1.0
Range: smpte=0:10:22-0:15:45
Seek-Style: Next
RTP-Info:url="rtsp://example.com/twister.en"
ssrc=0D12F123:seq=14783;rtptime=2345962545]]></artwork>
</figure>
<t>For playing back a recording of a live presentation, it may be
desirable to use clock units:</t>
<figure>
<artwork><![CDATA[
C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/2.0
CSeq: 835
Session: 12345678
Range: clock=19961108T142300Z-19961108T143520Z
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 835
Date: Thu, 23 Jan 1997 15:35:06 GMT
Session: 12345678
Server: PhonyServer/1.0
Range: clock=19961108T142300Z-19961108T143520Z
Seek-Style: Next
RTP-Info:url="rtsp://example.com/meeting.en"
ssrc=0D12F123:seq=53745;rtptime=484589019]]></artwork>
</figure>
</section>
<section anchor="sec_aggregated_sessions" title="Aggregated Sessions">
<t>PLAY requests can operate on sessions controlling a single media
and on aggregated sessions controlling multiple media.</t>
<t>In an aggregated session the PLAY request MUST contain an
aggregated control URI. A server MUST respond with error 460 (Only
Aggregate Operation Allowed) if the client PLAY Request-URI is for a
single media. The media in an aggregate MUST be played in sync. If a
client wants individual control of the media, it needs to use
separate RTSP sessions for each media.</t>
<t>For aggregated sessions where the initial SETUP request (creating
a session) is followed by one or more additional SETUP requests, a
PLAY request MAY be pipelined after those additional SETUP requests
without awaiting their responses. This procedure can reduce the
delay from start of session establishment until media play-out has
started with one round trip time. However, a client needs to be
aware that using this procedure will result in the playout of the
server state established at the time of processing the PLAY, i.e.,
after the processing of all the requests prior to the PLAY request
in the pipeline. This state may not be the intended one due to
failure of any of the prior requests. A client can easily determine
this based on the responses from those requests. In case of failure,
the client can halt the media playout using PAUSE and try to
establish the intended state again before issuing another PLAY
request.</t>
</section>
<section title="Updating current PLAY Requests">
<t>Clients can issue PLAY requests while the stream is in Play state
and thus updating their request.</t>
<t>The important difference compared to a PLAY request in Ready
state is the handling of the current play point and how the Range
header in the request is constructed. The session is actively
playing media and the play point will be moving, making the exact
time a request will take effect hard to predict. Depending on how
the PLAY header appears two different cases exist: total replacement
or continuation. A total replacement is signaled by having the first
range specification have an explicit start value, e.g., "npt=45-" or
"npt=45-60", in which case the server stops playout at the current
playout point and then starts delivering media according to the
Range header. This is equivalent to having the client first send a
PAUSE and then a new PLAY request that isn't based on the pause
point. In the case of continuation the first range specifier has an
implicit start point and an explicit stop value (Z), e.g.,
"npt=-60", which indicate that it MUST convert the range specifier
being played prior to this PLAY request (X to Y) into (X to Z) and
continue as this was the request originally played. If the current
delivery point is beyond the stop point, the server SHALL
immediately pause delivery. As the request has been completed
successfully it shall be responded with 200 OK. A PLAY_NOTIFY with
end-of-stream is also sent to indicate the actual stop point. The
pause point is set to the requested stop point.</t>
<t>Following is an example of this behavior: The server has received
requests to play ranges 10 to 15. If the new PLAY request arrives at
the server 4 seconds after the previous one, it will take effect
while the server still plays the first range (10-15). The server
changes the current play to continue to 25 seconds, i.e., the
equivalent single request would be PLAY with "range: npt=10-25".</t>
<figure>
<artwork><![CDATA[
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 834
Session: 12345678
Range: npt=10-15
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 834
Date: Thu, 23 Jan 1997 15:35:06 GMT
Session: 12345678
Server: PhonyServer/1.0
Range: npt=10-15
Seek-Style: Next
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934207921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=789DAF12:seq=57654;rtptime=2792482193
Session: 12345678
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 835
Session: 12345678
Range: npt=-25
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 835
Date: Thu, 23 Jan 1997 15:35:09 GMT
Session: 12345678
Server: PhonyServer/1.0
Range: npt=14-25
Seek-Style: Next
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934239921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=789DAF12:seq=57654;rtptime=2792842193
]]></artwork>
</figure>
<t>A common use of a PLAY request while in Play state is changing
the scale of the media, i.e., entering or leaving fast forward or
fast rewind. The client can issue an updating PLAY request that is
either a continuation or a complete replacement, as discussed above
this section. We give an example of a client that is requesting a
fast forward (scale=2) without giving a stop point and then change
from fast forward to regular playout (scale = 1). In the second PLAY
request the time is set explicitly to be where ever the server
currently plays out (npt=now-) and the server responds with the
actual playback point where the new scale actually takes effect
(npt=2:17:27.144-).</t>
<figure>
<artwork><![CDATA[
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 2034
Session: 12345678
Range: npt=now-
Scale: 2.0
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 2034
Date: Thu, 23 Jan 1997 15:35:06 GMT
Session: 12345678
Server: PhonyServer/1.0
Range: npt=2:17:21.394-
Seek-Style: Next
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934207921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=789DAF12:seq=57654;rtptime=2792482193
[playing in fast forward and now returning to scale = 1]
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 2035
Session: 12345678
Range: npt=now-
Scale: 1.0
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 2035
Date: Thu, 23 Jan 1997 15:35:09 GMT
Session: 12345678
Server: PhonyServer/1.0
Range: npt=2:17:27.144-
Seek-Style: Next
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934239921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=789DAF12:seq=57654;rtptime=2792842193
]]></artwork>
</figure>
</section>
<section title="Playing On-Demand Media">
<t>On-demand media is indicated by the content of the
Media-Properties header in the SETUP response by (see also <xref
target="sec_Media-Properties"/>):<list style="symbols">
<t>Random Access property is set to Random-Access;</t>
<t>Content Modifications set to Immutable;</t>
<t>Retention set to Unlimited or Time-Limited.</t>
</list>Playing on-demand media follows the general usage as
described in <xref target="sec_PLAY_general"/>.</t>
<t/>
</section>
<section title="Playing Dynamic On-Demand Media">
<t>Dynamic on-demand media is indicated by the content of the
Media-Properties header in the SETUP response by (see also <xref
target="sec_Media-Properties"/>):<list style="symbols">
<t>Random Access set to Random-Access;</t>
<t>Content Modifications set to Dynamic;</t>
<t>Retention set to Unlimited or Time-Limited.</t>
</list></t>
<t>Playing on-demand media follows the general usage as described in
<xref target="sec_PLAY_general"/> as long as the media has not been
changed.</t>
<t>There are two ways for the client to be informed about changes of
media resources in Play state. The client will receive a PLAY_NOTIFY
request with Notify-Reason header set to media-properties-update
(see <xref target="sec_Media-Properties-Update-Reason"/>. The client
can use the value of the Media-Range to decide further actions, if
the Media-Range header is present in the PLAY_NOTIFY request. The
second way is that the client issues a GET_PARAMETER request without
a body but including a Media-Range header. The 200 OK response MUST
include the current Media-Range header (see <xref
target="sec_Media-Range"/>).</t>
</section>
<section title="Playing Live Media">
<t>Live media is indicated by the content of the Media-Properties
header in the SETUP response by (see also <xref
target="sec_Media-Properties"/>):<list style="symbols">
<t>Random-Access set to No-Seeking;</t>
<t>Content Modifications set to Time-Progressing;</t>
<t>Retention with Time-Duration set to 0.0.</t>
</list></t>
<t>For live media, the SETUP response 200 OK MUST include the
Media-Range header (see <xref target="sec_Media-Range"/>).</t>
<t>A client MAY send PLAY requests without the Range header. If the
request includes the Range header it MUST use a symbolic value
representing "now". For NPT that range specification is "npt=now-".
The server MUST include the Range header in the response and it MUST
indicate an explicit time value and not a symbolic value. In other
words, "npt=now-" is not valid to be used in the response. Instead
the time since session start is recommended expressed as an open
interval, e.g., "npt=96.23-". An absolute time value (clock) for the
corresponding time MAY be given, i.e., "clock=20030213T143205Z-".
The Absolute Time format can only be used if client has shown
support for it using the Accept-Ranges header.</t>
</section>
<section title="Playing Live with Recording">
<t>Certain media servers may offer recording services of live
sessions to their clients. This recording would normally be from the
beginning of the media session. Clients can randomly access the
media between now and the beginning of the media session. This live
media with recording is indicated by the content of the
Media-Properties header in the SETUP response by (see also <xref
target="sec_Media-Properties"/>):<list style="symbols">
<t>Random Access set to Random-Access;</t>
<t>Content Modifications set to Time-Progressing;</t>
<t>Retention set to Time-Limited or Unlimited</t>
</list></t>
<t>The SETUP response 200 OK MUST include the Media-Range header
(see <xref target="sec_Media-Range"/>) for this type of media. For
live media with recording, the Range header indicates the current
delivery point in the media and the Media-Range header indicates the
currently available media window around the current time. This
window can cover recorded content in the past (seen from current
time in the media) or recorded content in the future (seen from
current time in the media). The server adjusts the delivery point to
the requested border of the window. If the client requests a
delivery point that is located outside the recording window, e.g.,
if the requested point is too far in the past, the server selects
the oldest point in the recording. The considerations in <xref
target="sec_ScaleChange"/> apply if a client requests delivery with
<xref target="sec_Scale">Scale</xref> values other than 1.0 (Normal
playback rate) while delivering live media with recording.</t>
</section>
<section title="Playing Live with Time-Shift">
<t>Certain media servers may offer time-shift services to their
clients. This time shift records a fixed interval in the past, i.e.,
a sliding window recording mechanism, but not past this interval.
Clients can randomly access the media between now and the interval.
This live media with recording is indicated by the content of the
Media-Properties header in the SETUP response by (see also <xref
target="sec_Media-Properties"/>):</t>
<t><list style="symbols">
<t>Random Access set to Random-Access;</t>
<t>Content Modifications set to Time-Progressing;</t>
<t>Retention set to Time-Duration and a value indicating the
recording interval (>0).</t>
</list></t>
<t>The SETUP response 200 OK MUST include the Media-Range header
(see <xref target="sec_Media-Range"/>) for this type of media. For
live media with recording the Range header indicates the current
time in the media and the Media Range indicates a window around the
current time. This window can cover recorded content in the past
(seen from current time in the media) or recorded content in the
future (seen from current time in the media). The server adjusts the
play point to the requested border of the window, if the client
requests a play point that is located outside the recording windows,
e.g., if requested too far in the past, the server selects the
oldest range in the recording. The considerations in <xref
target="sec_ScaleChange"/> apply, if a client requests delivery
using a <xref target="sec_Scale">Scale</xref> value other than 1.0
(Normal playback rate) while delivering live media with
time-shift.</t>
</section>
</section>
<section anchor="sec_PLAY_NOTIFY" title="PLAY_NOTIFY">
<t>The PLAY_NOTIFY method is issued by a server to inform a client
about an asynchronous event for a session in Play state. The Session
header MUST be presented in a PLAY_NOTIFY request and indicates the
scope of the request. Sending of PLAY_NOTIFY requests requires a
persistent connection between server and client, otherwise there is no
way for the server to send this request method to the client.</t>
<t>PLAY_NOTIFY requests have an end-to-end (i.e., server to client)
scope, as they carry the Session header, and apply only to the given
session. The client SHOULD immediately return a response to the
server.</t>
<t>PLAY_NOTIFY requests MAY use both aggregate control URI and
individual media resource URIs depending on the scope of the
notification. This scope may have important distinctions for
aggregated sessions, and each reason for a PLAY_NOTIFY request needs
to specify the interpretation and if aggregated control URIs or
individual URIs may be used in requests.</t>
<t>PLAY_NOTIFY requests MAY be used with a message body, depending on
the value of the Notify-Reason header. It is described in the
particular section for each Notify-Reason if a message body is used.
However, currently there is no Notify-Reason that allows using a
message body. In this case, there is a need to obey some limitations
when adding new Notify-Reasons that intend to use a message body: the
server can send any type of message body, but it is not ensured that
the client can understand the received message body. This is related
to DESCRIBE (see <xref target="sec_DESCRIBE"> </xref> ), but in this
particular case the client can state its acceptable message bodies by
using the Accept header. In the case of PLAY_NOTIFY, the server does
not know which message bodies are understood by the client.</t>
<t>The Notify-Reason header (see <xref target="sec_Notify-Reason"/>)
specifies the reason why the server sends the PLAY_NOTIFY request.
This is extensible and new reasons MAY be added in the future (see
<xref target="sec_iana_Notify-Reason_header"/>). In case the client
does not understand the reason for the notification it MUST respond
with a <xref target="sec_error465">465 (Notification Reason
Unknown)</xref> error code. Servers can send PLAY_NOTIFY with these
types:</t>
<t><list style="symbols">
<t>end-of-stream (see <xref target="sec_end_of_stream"/>);</t>
<t>media-properties-update (see <xref
target="sec_Media-Properties-Update-Reason"/>);</t>
<t>scale-change (see <xref target="sec_ScaleChange"/>).</t>
</list></t>
<section anchor="sec_end_of_stream" title="End-of-Stream">
<t>A PLAY_NOTIFY request with Notify-Reason header set to
end-of-stream indicates the completion or near completion of the
PLAY request and the ending delivery of the media stream(s). The
request MUST NOT be issued unless the server is in the Play state.
The end of the media stream delivery notification may be used to
indicate either a successful completion of the PLAY request
currently being served, or to indicate some error resulting in
failure to complete the request. The <xref
target="sec_Request-Status">Request-Status header</xref> MUST be
included to indicate which request the notification is for and its
completion status. The <xref target="sec_status-code">message
response status codes</xref> are used to indicate how the PLAY
request concluded. The sender of a PLAY_NOTIFY can issue an updated
PLAY_NOTIFY, in the case of a PLAY_NOTIFY sent with wrong
information. For instance, a PLAY_NOTIFY was issued before reaching
the end-of-stream, but some error occurred resulting in that the
previously sent PLAY_NOTIFY contained a wrong time when the stream
will end. In this case a new PLAY_NOTIFY MUST be sent including the
correct status for the completion and all additional
information.</t>
<t>PLAY_NOTIFY requests with Notify-Reason header set to
end-of-stream MUST include a Range header and the Scale header if
the scale value is not 1. The Range header indicates the point in
the stream or streams where delivery is ending with the timescale
that was used by the server in the PLAY response for the request
being fulfilled. The server MUST NOT use the "now" constant in the
Range header; it MUST use the actual numeric end position in the
proper timescale. When end-of-stream notifications are issued prior
to having sent the last media packets, this is evident as the end
time in the Range header is beyond the current time in the media
being received by the client, e.g., "npt=-15", if npt is currently
at 14.2 seconds. The Scale header is to be included so that it is
evident if the media time scale is moving backwards and/or have a
non-default pace. The end-of-stream notification does not prevent
the client from sending a new PLAY request.</t>
<t>If RTP is used as media transport, a RTP-Info header MUST be
included, and the RTP-Info header MUST indicate the last sequence
number in the seq parameter.</t>
<t>For an RTSP Session where media resources are under aggregated
control the media resources will normally end at approximately the
same time, although some small differences may exist, on the scale
of a few hundred of milliseconds. In those cases a RTSP session
under aggregated control SHOULD send only a single PLAY_NOTIFY
request. By using the aggregate control URI in the PLAY_NOTIFY
request the RTSP server indicates that this applies to all media
resources within the session. In cases RTP is used for media
delivery corresponding RTP-Info needs to be included for all media
resources. In cases where one or more media resource has
significantly shorter duration than some other resources in the
aggregated session the server MAY send end-of-stream notifications
using individual media resource URIs to indicate to agents that
there will be no more media for this particular media resource
related to the current active PLAY request. In such cases when the
remaining media resources comes to end-of-stream they MUST send a
PLAY_NOTIFY request using the aggregate control URI to indicate that
no more resources remain.</t>
<t>A PLAY_NOTIFY request with Notify-Reason header set to
end-of-stream MUST NOT carry a message body.</t>
<t>This example request notifies the client about a future
end-of-stream event:</t>
<figure>
<artwork><![CDATA[
S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 854
Notify-Reason: end-of-stream
Request-Status: cseq=853 status=200 reason="OK"
Range: npt=-145
RTP-Info:url="rtsp://example.com/fizzle/foo/audio"
ssrc=0D12F123:seq=14783;rtptime=2345962545,
url="rtsp://example.com/fizzle/video"
ssrc=789DAF12:seq=57654;rtptime=2792482193
Session: uZ3ci0K+Ld-M
Date: Mon, 08 Mar 2010 13:37:16 GMT
C->S: RTSP/2.0 200 OK
CSeq: 854
User-Agent: PhonyClient/1.2
Session: uZ3ci0K+Ld-M]]></artwork>
</figure>
<t/>
</section>
<section anchor="sec_Media-Properties-Update-Reason"
title="Media-Properties-Update">
<t>A PLAY_NOTIFY request with Notify-Reason header set to
media-properties-update indicates an update of the media properties
for the given session (see <xref target="sec_Media-Properties"/>)
and/or the available media range that can be played as indicated by
<xref target="sec_Media-Range">Media-Range</xref>. PLAY_NOTIFY
requests with Notify-Reason header set to media-properties-update
MUST include a Media-Properties and Date header and SHOULD include a
Media-Range header. The Media-Properties header has session scope,
thus for aggregated sessions the PLAY_NOTIFY request MUST be using
the aggregated control URI.</t>
<t>This notification MUST be sent for media that are
Time-Progressing every time an event happens that changes the basis
for making estimates on how the available for play-back media range
will progress with wall clock time. In addition it is RECOMMENDED
that the server sends these notifications approximately every 5
minutes for time-progressing content to ensure the long-term
stability of the client estimation and allowing for clock skew
detection by the client. The time between notifications should be
greater than 1 minute and less than 2 hours. For the reasons just
explained, requests MUST include a Media-Range header to provide
current Media duration and a Range header to indicate the current
playing point and any remaining parts of the requested range.<list
style="hanging">
<t>The recommendation for sending updates every 5 minutes is due
to any clock skew issues. In 5 minutes the clock skew should not
become too significant as this is not used for media playback
and synchronization, only for determining which content is
available to the user.</t>
</list></t>
<t>A PLAY_NOTIFY request with Notify-Reason header set to
media-properties-update MUST NOT carry a message body.</t>
<figure>
<artwork><![CDATA[
S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0
Date: Tue, 14 Apr 2008 15:48:06 GMT
CSeq: 854
Notify-Reason: media-properties-update
Session: uZ3ci0K+Ld-M
Media-Properties: Time-Progressing,
Time-Limited=20080415T153919.36Z, Random-Access=5.0
Media-Range: npt=0-1:37:21.394
Range: npt=1:15:49.873-
C->S: RTSP/2.0 200 OK
CSeq: 854
User-Agent: PhonyClient/1.2
Session: uZ3ci0K+Ld-M]]></artwork>
</figure>
<t/>
</section>
<section anchor="sec_ScaleChange" title="Scale-Change">
<t>The server may be forced to change the rate of media time per
playback time when a client requests delivery using a <xref
target="sec_Scale">Scale</xref> value other than 1.0 (normal
playback rate). For time progressing media with some retention,
i.e., the server stores already sent content, a client requesting to
play with Scale values larger than 1 may catch up with the front end
of the media. The server will then be unable to continue to provide
content at Scale larger than 1 as content is only made available by
the server at Scale=1. Another case is when Scale < 1 and the
media retention is time-duration limited. In this case the delivery
point can reach the oldest media unit available, and further
playback at this scale becomes impossible as there will be no media
available. To avoid having the client lose any media, the scale will
need to be adjusted to the same rate at which the media is removed
from the storage buffer, commonly Scale = 1.0.</t>
<t>Another case is when the content itself consists of spliced
pieces or is dynamically updated. In these cases the server may be
required to change from one supported scale value (different than
Scale=1.0) to another. In this case the server will pick the closest
value and inform the client of what it has picked. In these cases
the media properties will also be sent updating the supported Scale
values. This enables a client to adjust the Scale value used.</t>
<t>To minimize impact on playback in any of the above cases the
server MUST modify the playback properties and set Scale to a
supportable value and continue delivery of the media. When doing
this modification it MUST send a PLAY_NOTIFY message with the
Notify-Reason header set to "scale-change". The request MUST contain
a Range header with the media time when the change took effect, a
Scale header with the new value in use, Session header with the
identifier for the session it applies to and a Date header with the
server wallclock time of the change. For time progressing content
also the Media-Range and the Media-Properties at this point in time
MUST be included. The Media-Properties header MUST be included if
the scale change was due to the content changing what scale values
that is supported.</t>
<t>For media streams being delivered using RTP also a RTP-Info
header MUST be included. It MUST contain the rtptime parameter with
a value corresponding to the point of change in that media and
optionally also the sequence number.</t>
<t>PLAY_NOTIFY requests for aggregated sessions MUST use the
aggregated control URI in the request. The scale change for any
aggregated session applies to all media streams part of the
aggregate.</t>
<t>A PLAY_NOTIFY request with Notify-Reason header set to
"Scale-Change" MUST NOT carry a message body.</t>
<figure>
<artwork><![CDATA[
S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0
Date: Tue, 14 Apr 2008 15:48:06 GMT
CSeq: 854
Notify-Reason: scale-change
Session: uZ3ci0K+Ld-M
Media-Properties: Time-Progressing,
Time-Limited=20080415T153919.36Z, Random-Access=5.0
Media-Range: npt=0-1:37:21.394
Range: npt=1:37:21.394-
Scale: 1
RTP-Info: url="rtsp://example.com/fizzle/foo/audio"
ssrc=0D12F123:rtptime=2345962545,
url="rtsp://example.com/fizzle/videotrack"
ssrc=789DAF12:seq=57654;rtptime=2792482193
C->S: RTSP/2.0 200 OK
CSeq: 854
User-Agent: PhonyClient/1.2
Session: uZ3ci0K+Ld-M]]></artwork>
</figure>
<t/>
</section>
</section>
<section anchor="sec_PAUSE" title="PAUSE">
<t>The PAUSE request causes the stream delivery to immediately be
interrupted (halted). A PAUSE request MUST be done either with the
aggregated control URI for aggregated sessions, resulting in all media
being halted, or the media URI for non-aggregated sessions. Any
attempt to do muting of a single media with a PAUSE request in an
aggregated session MUST be responded to with error 460 (Only Aggregate
Operation Allowed). After resuming playback, synchronization of the
tracks MUST be maintained. Any server resources are kept, though
servers MAY close the session and free resources after being paused
for the duration specified with the timeout parameter of the Session
header in the SETUP message.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 834
Session: 12345678
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 834
Date: Thu, 23 Jan 1997 15:35:06 GMT
Range: npt=45.76-75.00]]></artwork>
</figure>
<t>The PAUSE request causes stream delivery to be interrupted
immediately on receipt of the message and the pause point is set to
the current point in the presentation. That pause point in the media
stream needs to be maintained. A subsequent PLAY request without Range
header resumes from the pause point and plays until media end.</t>
<t>The pause point after any PAUSE request MUST be returned to the
client by adding a Range header with what remains unplayed of the PLAY
request's range. For media with random access properties, if one
desires to resume playing a ranged request, one simply includes the
Range header from the PAUSE response and includes the Seek-Style
header with the Next policy in the PLAY request. For media that is
time-progressing and has retention duration=0 the follow-up PLAY
request to start media delivery again, MUST use "npt=now-" and not the
answer given in the response to PAUSE.</t>
<figure>
<artwork><![CDATA[
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 834
Session: 12345678
Range: npt=10-30
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 834
Date: Thu, 23 Jan 1997 15:35:06 GMT
Server: PhonyServer/1.0
Range: npt=10-30
Seek-Style: First-Prior
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934207921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=4FAD8726:seq=57654;rtptime=2792482193
Session: 12345678
After 11 seconds, i.e., at 21 seconds into the presentation:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 835
Session: 12345678
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 835
Date: 23 Jan 1997 15:35:17 GMT
Server: PhonyServer/1.0
Range: npt=21-30
Session: 12345678]]></artwork>
</figure>
<t>If a client issues a PAUSE request and the server acknowledges and
enters the Ready state, the proper server response, if the player
issues another PAUSE, is still 200 OK. The 200 OK response MUST
include the Range header with the current pause point. See examples
below:</t>
<figure>
<artwork><![CDATA[
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 834
Session: 12345678
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 834
Session: 12345678
Date: Thu, 23 Jan 1997 15:35:06 GMT
Range: npt=45.76-98.36
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 835
Session: 12345678
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 835
Session: 12345678
Date: 23 Jan 1997 15:35:07 GMT
Range: npt=45.76-98.36]]></artwork>
</figure>
</section>
<section anchor="sec_TEARDOWN" title="TEARDOWN">
<t/>
<section title="Client to Server">
<t>The TEARDOWN client to server request stops the stream delivery
for the given URI, freeing the resources associated with it. A
TEARDOWN request MAY be performed on either an aggregated or a media
control URI. However, some restrictions apply depending on the
current state. The TEARDOWN request MUST contain a Session header
indicating what session the request applies to. The TEARDOWN request
MUST NOT include a Terminate-Reason header.</t>
<t>A TEARDOWN using the aggregated control URI or the media URI in a
session under non-aggregated control (single media session) MAY be
done in any state (Ready and Play). A successful request MUST result
in that media delivery being immediately halted and the session
state being destroyed. This MUST be indicated through the lack of a
Session header in the response.</t>
<t>A TEARDOWN using a media URI in an aggregated session MAY only be
done in Ready state. Such a request only removes the indicated media
stream and associated resources from the session. This may result in
a session returning to non-aggregated control, because it only
contains a single media after the request's completion. A session
that will exist after the processing of the TEARDOWN request MUST in
the response to that TEARDOWN request contain a Session header. Thus
the presence of the Session header indicates to the receiver of the
response if the session is still extant or has been removed.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 892
Session: 12345678
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 892
Server: PhonyServer/1.0]]></artwork>
</figure>
<t/>
</section>
<section title="Server to Client">
<t>The server can send TEARDOWN requests in the server to client
direction to indicate that the server has been forced to terminate
the ongoing session. This may happen for several reasons, such as
server maintenance without available backup, or that the session has
been inactive for extended periods of time. The reason is provided
in the <xref target="sec_Terminate-Reason">Terminate-Reason
header</xref>.</t>
<t>When a RTSP client has maintained a RTSP session that otherwise
is inactive for an extended period of time the server may reclaim
the resources. That is done by issuing a TEARDOWN request with the
Terminate-Reason set to "Session-Timeout". This MAY be done when the
client has been inactive in the RTSP session for more than one <xref
target="sec_Session">Session Timeout period</xref>. However, the
server is RECOMMENDED to not perform this operation until an
extended period of inactivity of 10 times the Session Timeout period
has passed. It is up to the operator of the RTSP server to actually
configure how long this extended period of inactivity is. An
operator should take into account when doing this configuration what
the served content is and what this means for the extended period of
inactivity.</t>
<t>In case the server needs to stop providing service to the
established sessions and there is no server to point at in a
REDIRECT request, then TEARDOWN SHALL be used to terminate the
session. This method can also be used when non-recoverable internal
errors have happened and the server has no other option then to
terminate the sessions.</t>
<t>The TEARDOWN request MUST be done only on the session aggregate
control URI (i.e., it is not allowed to terminate individual media
streams, if it is a session aggregate) and MUST include the
following headers; Session and Terminate-Reason headers. The request
only applies to the session identified in the Session header. The
server may include a message to the client's user with the
"user-msg" parameter.</t>
<t>The TEARDOWN request may alternatively be done on the wild card
URI * and without any session header. The scope of such a request is
limited to the next-hop (i.e., the RTSP agent in direct
communication with the server) and applies, as well, to the RTSP
connection between the next-hop RTSP agent and the server. This
request indicates that all sessions and pending requests being
managed via the connection are terminated. Any intervening proxies
SHOULD do all of the following in the order listed: <list
hangIndent="3" style="numbers">
<t>respond to the TEARDOWN request</t>
<t>disconnect the control channel from the requesting server</t>
<t>pass the TEARDOWN request to each applicable client
(typically those clients with an active session or an unanswered
request)</t>
</list></t>
<t><list style="hanging">
<t>Note: The proxy is responsible for accepting TEARDOWN
responses from its clients; these responses MUST NOT be passed
on to either the original server or the target server in the
redirect.</t>
</list></t>
<t/>
</section>
</section>
<section anchor="sec_GET_PARAMETER" title="GET_PARAMETER">
<t>The GET_PARAMETER request retrieves the value of any specified
parameter or parameters for a presentation or stream specified in the
URI. If the Session header is present in a request, the value of a
parameter MUST be retrieved in the specified session context. There
are two ways of specifying the parameters to be retrieved.</t>
<t>The first is by including headers which have been defined such that
you can use them for this purpose. Headers for this purpose should
allow empty, or stripped value parts to avoid having to specify bogus
data when indicating the desire to retrieve a value. The successful
completion of the request should also be evident from any filled out
values in the response. The headers in this specification that MAY be
used for retrieving their current value using GET_PARAMETER are listed
below; additional headers MAY be specified in the future:<list
style="symbols">
<t>Accept-Ranges</t>
<t>Media-Range</t>
<t>Media-Properties</t>
<t>Range</t>
<t>RTP-Info</t>
</list></t>
<t>The other way is to specify a message body that lists the
parameter(s) that are desired to be retrieved. The <xref
target="sec_Content-Type">Content-Type header</xref> is used to
specify which format the message body has. If the receiver of the
request does not support the media type used for the message body, it
SHALL respond using the error code 415 (Unsupported Media Type). The
responder to a GET_PARAMETER request MUST use the media type of the
request for the response. For additional considerations regarding
message body negotiation see <xref
target="sec_Message-Body-Format-Negotiation"/>.</t>
<t>RTSP Agents implementing support for responding to GET_PARAMETER
requests SHALL implement the <xref
target="sec_text-parameters">text/parameters format</xref>. This to
ensure that at least one known format for parameters is implemented
and thus prevent parameter format negotiation failure.</t>
<t>Parameters specified within the body of the message must all be
understood by the request receiving agent. If one or more parameters
are not understood a 451 (Parameter Not Understood) MUST be sent
including a body listing these parameters that weren't understood. If
all parameters are understood their values are filled in and returned
in the response message body.</t>
<t>The method MAY also be used without a message body or any header
that requests parameters for keep-alive purpose. The keep-alive timer
has been updated for any request that is successful, i.e., a 200 OK
response is received. Any non-required header present in such a
request may or may not have been processed. Normally the presence of
filled out values in the header will be indication that the header has
been processed. However, for cases when this is difficult to
determine, it is recommended to use a feature-tag and the Require
header. For this reason it is usually easier if any parameters to be
retrieved are sent in the body, rather than using any header.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 431
User-Agent: PhonyClient/1.2
Session: 12345678
Content-Length: 26
Content-Type: text/parameters
packets_received
jitter
C->S: RTSP/2.0 200 OK
CSeq: 431
Session: 12345678
Server: PhonyServer/1.1
Date: Mon, 08 Mar 2010 13:43:23 GMT
Content-Length: 38
Content-Type: text/parameters
packets_received: 10
jitter: 0.3838]]></artwork>
</figure>
<t/>
</section>
<section anchor="sec_SET_PARAMETER" title="SET_PARAMETER">
<t>This method requests to set the value of a parameter or a set of
parameters for a presentation or stream specified by the URI. The
method MAY also be used without a message body. It is the RECOMMENDED
method to be used in a request sent for the sole purpose of updating
the keep-alive timer. If this request is successful, i.e., a 200 OK
response is received, then the keep-alive timer has been updated. Any
non-required header present in such a request may or may not have been
processed. To allow a client to determine if any such header has been
processed, it is necessary to use a feature tag and the Require
header. Due to this reason it is RECOMMENDED that any parameters are
sent in the body, rather than using any header.</t>
<t>When using a message body to list the parameter(s) that are desired
to be set the <xref target="sec_Content-Type">Content-Type
header</xref> is used to specify which format the message body has. If
the receiver of the request is not supporting the media type used for
the message body, it SHALL respond using the error code 415
(Unsupported Media Type). For additional considerations regarding
message body negotiation see <xref
target="sec_Message-Body-Format-Negotiation"/>.</t>
<t>RTSP Agents implementing support for responding to SET_PARAMETER
requests SHALL implement the <xref
target="sec_text-parameters">text/parameters format</xref>. This to
ensure that at least one known format for parameters is implemented
and thus prevent parameter format negotiation failure.</t>
<t>A request is RECOMMENDED to only contain a single parameter to
allow the client to determine why a particular request failed. If the
request contains several parameters, the server MUST only act on the
request if all of the parameters can be set successfully. A server
MUST allow a parameter to be set repeatedly to the same value, but it
MAY disallow changing parameter values. If the receiver of the request
does not understand or cannot locate a parameter, error 451 (Parameter
Not Understood) MUST be used. When a parameter is not allowed to
change, the error code is 458 (Parameter Is Read-Only). The response
body MUST contain only the parameters that have errors. Otherwise no
body MUST be returned. The response body MUST use the media type of
the request for the response.</t>
<t>Note: transport parameters for the media stream MUST only be set
with the SETUP command.</t>
<t><list style="hanging">
<t>Restricting setting transport parameters to SETUP is for the
benefit of firewalls.</t>
</list></t>
<t><list style="hanging">
<t>The parameters are split in a fine-grained fashion so that
there can be more meaningful error indications. However, it may
make sense to allow the setting of several parameters if an atomic
setting is desirable. Imagine device control where the client does
not want the camera to pan unless it can also tilt to the right
angle at the same time.</t>
</list></t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 421
User-Agent: PhonyClient/1.2
Session: iixT43KLc
Date: Mon, 08 Mar 2010 14:45:04 GMT
Content-length: 20
Content-type: text/parameters
barparam: barstuff
S->C: RTSP/2.0 451 Parameter Not Understood
CSeq: 421
Session: iixT43KLc
Server: PhonyServer/1.0
Date: Mon, 08 Mar 2010 14:44:56 GMT
Content-length: 20
Content-type: text/parameters
barparam: barstuff]]></artwork>
</figure>
<t/>
</section>
<section anchor="sec_REDIRECT" title="REDIRECT">
<t>The REDIRECT method is issued by a server to inform a client that
the service provided will be terminated and where a corresponding
service can be provided instead. This may happen for different
reasons. One is that the server is being administered such that it
must stop providing service. Thus the client is required to connect to
another server location to access the resource indicated by the
Request-URI.</t>
<t>The REDIRECT request SHALL contain a <xref
target="sec_Terminate-Reason">Terminate-Reason header</xref> to inform
the client of the reason for the request. Additional parameters
related to the reason may also be included. The intention here is to
allow a server administrator to do a controlled shutdown of the RTSP
server. That requires sufficient time to inform all entities having
associated state with the server and for them to perform a controlled
migration from this server to a fall back server.</t>
<t>A REDIRECT request with a Session header has end-to-end (i.e.,
server to client) scope and applies only to the given session. Any
intervening proxies SHOULD NOT disconnect the control channel while
there are other remaining end-to-end sessions. The REQUIRED Location
header MUST contain a complete absolute URI pointing to the resource
to which the client SHOULD reconnect. Specifically, the Location MUST
NOT contain just the host and port. A client may receive a REDIRECT
request with a Session header, if and only if, an end-to-end session
has been established.</t>
<t>A client may receive a REDIRECT request without a Session header at
any time when it has communication or a connection established with a
server. The scope of such a request is limited to the next-hop (i.e.,
the RTSP agent in direct communication with the server) and applies to
all sessions controlled, as well as the connection between the
next-hop RTSP agent and the server. A REDIRECT request without a
Session header indicates that all sessions and pending requests being
managed via the connection MUST be redirected. The Location header, if
included in such a request, SHOULD contain an absolute URI with only
the host address and the OPTIONAL port number of the server to which
the RTSP agent SHOULD reconnect. Any intervening proxies SHOULD do all
of the following in the order listed: <list hangIndent="3"
style="numbers">
<t>respond to the REDIRECT request</t>
<t>disconnect the control channel from the requesting server</t>
<t>connect to the server at the given host address</t>
<t>pass the REDIRECT request to each applicable client (typically
those clients with an active session or an unanswered request)</t>
</list></t>
<t><list style="hanging">
<t>Note: The proxy is responsible for accepting REDIRECT responses
from its clients; these responses MUST NOT be passed on to either
the original server or the redirected server.</t>
</list></t>
<t>When the server lacks any alternative server and needs to terminate
a session or all sessions the TEARDOWN request SHALL be used
instead.</t>
<t>When no Terminate-Reason "time" parameter is included in a REDIRECT
request, the client SHALL perform the redirection immediately and
return a response to the server. The server shall consider the session
as terminated and can free any associated state after it receives the
successful (2xx) response. The server MAY close the signaling
connection upon receiving the response and the client SHOULD close the
signaling connection after sending the 2xx response. The exception to
this is when the client has several sessions on the server being
managed by the given signaling connection. In this case, the client
SHOULD close the connection when it has received and responded to
REDIRECT requests for all the sessions managed by the signaling
connection.</t>
<t>The Terminate-Reason header "time" parameter MAY be used to
indicate the wallclock time by when the redirection MUST have taken
place. To allow a client to determine that redirect time without being
time synchronized with the server, the server MUST include a Date
header in the request. The client should have terminated the session
and closed the connection before the redirection time-line terminated.
The server MAY simply cease to provide service when the deadline time
has been reached, or it may issue TEARDOWN requests to the remaining
sessions.</t>
<t>If the REDIRECT request times out following the rules in <xref
target="sec_connection-timeout"/> the server MAY terminate the session
or transport connection that would be redirected by the request. This
is a safeguard against misbehaving clients that refuse to respond to a
REDIRECT request. Thus, removing any incentive to not acknowledge the
reception of a REDIRECT request.</t>
<t>After a REDIRECT request has been processed, a client that wants to
continue to receive media for the resource identified by the
Request-URI will have to establish a new session with the designated
host. If the URI given in the Location header is a valid resource URI,
a client SHOULD issue a DESCRIBE request for the URI.</t>
<t><list style="hanging">
<t>Note: The media resource indicated by the Location header can
be identical, slightly different or totally different. This is the
reason why a new DESCRIBE request SHOULD be issued.</t>
</list></t>
<t>If the Location header contains only a host address, the client MAY
assume that the media on the new server is identical to the media on
the old server, i.e., all media configuration information from the old
session is still valid except for the host address. However, the usage
of conditional SETUP using MTag identifiers is RECOMMENDED as a means
to verify the assumption.</t>
<t>This example request redirects traffic for this session to the new
server at the given absolute time:</t>
<figure>
<artwork><![CDATA[
S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 732
Location: rtsp://s2.example.com:8001
Terminate-Reason: Server-Admin ;time=19960213T143205Z
Session: uZ3ci0K+Ld-M
Date: Thu, 13 Feb 1996 14:30:43 GMT
C->S: RTSP/2.0 200 OK
CSeq: 732
User-Agent: PhonyClient/1.2
Session: uZ3ci0K+Ld-M]]></artwork>
</figure>
<t/>
</section>
</section>
<section anchor="sec_binary" title="Embedded (Interleaved) Binary Data">
<t>In order to fulfill certain requirements on the network side, e.g.,
in conjunction with network address translators that block RTP traffic
over UDP, it may be necessary to interleave RTSP messages and media
stream data. This interleaving should generally be avoided unless
necessary since it complicates client and server operation and imposes
additional overhead. Also, head-of-line blocking may cause problems.
Interleaved binary data SHOULD only be used if RTSP is carried over TCP.
Interleaved data is not allowed inside RTSP messages.</t>
<t>Stream data such as RTP packets is encapsulated by an ASCII dollar
sign (36 decimal), followed by a one-octet channel identifier, followed
by the length of the encapsulated binary data as a binary, two-octet
unsigned integer in network octet order (Appendix B of <xref
target="RFC0791"/>). The stream data follows immediately afterwards,
without a CRLF, but including the upper-layer protocol headers. Each $
block MUST contain exactly one upper-layer protocol data unit, e.g., one
RTP packet. <list style="empty">
<t>Note that this mechanism does not support PDUs larger than 65535
octets, which matches the maximum payload size of regular, non-jumbo
IPv4 and IPv6 packets. If the media delivery protocol intended to be
used has larger PDUs than that, definition of a PDU fragmentation
mechanism will be required to support embedded binary data.</t>
</list></t>
<figure anchor="fig-binary"
title="Embedded Interleaved Binary Data Format">
<artwork><![CDATA[
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| "$" = 36 | Channel ID | Length in octets |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: Binary data (Length according to Length field) :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+]]></artwork>
</figure>
<t>The channel identifier is defined in the Transport header with the
interleaved parameter (<xref target="sec_Transport"/>).</t>
<t>When the transport choice is RTP, RTCP messages are also interleaved
by the server over the TCP connection. The usage of RTCP messages is
indicated by including an interval containing a second channel in the
interleaved parameter of the Transport header, see <xref
target="sec_Transport"/>. If RTCP is used, packets MUST be sent on the
first available channel higher than the RTP channel. The channels are
bi-directional, using the same Channel ID in both directions, and
therefore RTCP traffic is sent on the second channel in both
directions.</t>
<t><list style="hanging">
<t>RTCP is sometimes needed for synchronization when two or more
streams are interleaved in such a fashion. Also, this provides a
convenient way to tunnel RTP/RTCP packets through the RTSP
connection (TCP or TCP/TLS) when required by the network
configuration and transfer them onto UDP when possible.</t>
</list></t>
<t/>
<figure>
<artwork><![CDATA[
C->S: SETUP rtsp://example.com/bar.file RTSP/2.0
CSeq: 2
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
Accept-Ranges: npt, smpte, clock
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 2
Date: Thu, 05 Jun 1997 18:57:18 GMT
Transport: RTP/AVP/TCP;unicast;interleaved=5-6
Session: 12345678
Accept-Ranges: npt
Media-Properties: Random-Access=0.2, Immutable, Unlimited
C->S: PLAY rtsp://example.com/bar.file RTSP/2.0
CSeq: 3
Session: 12345678
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 3
Session: 12345678
Date: Thu, 05 Jun 1997 18:57:19 GMT
RTP-Info: url="rtsp://example.com/bar.file"
ssrc=0D12F123:seq=232433;rtptime=972948234
Range: npt=0-56.8
Seek-Style: RAP
S->C: $005{2 octet length}{"length" octets data, w/RTP header}
S->C: $005{2 octet length}{"length" octets data, w/RTP header}
S->C: $006{2 octet length}{"length" octets RTCP packet}]]></artwork>
</figure>
</section>
<section anchor="sec_proxies" title="Proxies">
<t>RTSP Proxies are RTSP agents that are located in between a client and
a server. A proxy can take on both the role as a client and as server
depending on what it tries to accomplish. RTSP proxies use two transport
layer connections, one from the RTSP client to the RTSP proxy and a
second from the RTSP proxy to the RTSP server. Proxies are introduced
for several different reasons and those listed below are often
combined.</t>
<t><list hangIndent="6" style="hanging">
<t hangText="Caching Proxy:">This type of proxy is used to reduce
the workload on servers and connections. By caching the description
and media streams, i.e., the presentation, the proxy can serve a
client with content, but without requesting it from the server once
it has been cached and has not become stale. See the caching <xref
target="sec_caching"/>. This type of proxy is also expected to
understand RTSP end-point functionality, i.e., functionality
identified in the Require header in addition to what Proxy-Require
demands.</t>
<t hangText="Translator Proxy:">This type of proxy is used to ensure
that an RTSP client gets access to servers and content on an
external network or using content encodings not supported by the
client. The proxy performs the necessary translation of addresses,
protocols or encodings. This type of proxy is expected to also
understand RTSP end-point functionality, i.e., functionality
identified in the Require header in addition to what Proxy-Require
demands.</t>
<t hangText="Access Proxy:">This type of proxy is used to ensure
that an RTSP client gets access to servers on an external network.
Thus this proxy is placed on the border between two domains, e.g., a
private address space and the public Internet. The proxy performs
the necessary translation, usually addresses. This type of proxy is
required to redirect the media to itself or a controlled gateway
that performs the translation before the media can reach the
client.</t>
<t hangText="Security Proxy:">This type of proxy is used to help
facilitate security functions around RTSP. For example when having a
firewalled network, the security proxy requests that the necessary
pinholes in the firewall are opened when a client in the protected
network wants to access media streams on the external side. This
proxy can also limit the client's access to certain types of
content. This proxy can perform its function without redirecting the
media between the server and client. However, in deployments with
private address spaces this proxy is likely to be combined with the
access proxy. Anyway, the functionality of this proxy is usually
closely tied into understanding all aspects of the media
transport.</t>
<t hangText="Auditing Proxy:">RTSP proxies can also provide network
owners with a logging and audit point for RTSP sessions, e.g., for
corporations that track their employees usage of the network. This
type of proxy can perform its function without inserting itself or
any other node in the media transport. This proxy type can also
accept unknown methods as it doesn't interfere with the clients'
requests.</t>
</list></t>
<t>All types of proxies can also be used when using secured
communication with TLS as RTSP 2.0 allows the client to approve
certificate chains used for connection establishment from a proxy, see
<xref target="sec_security-tls-proxy"/>. However, that trust model may
not be suitable for all types of deployment. In those cases, the secured
sessions do by-pass the proxies.</t>
<t>Access proxies SHOULD NOT be used in equipment like NATs and
firewalls that aren't expected to be regularly maintained, like home or
small office equipment. In these cases it is better to use the NAT
traversal procedures defined for RTSP 2.0 <xref
target="I-D.ietf-mmusic-rtsp-nat"/>. The reason for these
recommendations is that any extensions of RTSP resulting in new media
transport protocols or profiles, new parameters, etc. may fail in a
proxy that isn't maintained. This would impede RTSP's future development
and usage.</t>
<section anchor="sec_proxies_ext"
title="Proxies and Protocol Extensions">
<t>The existence of proxies must always be considered when developing
new RTSP extensions. Most types of proxies will need to implement any
new method to operate correctly in the presence of that extension. New
headers can be introduced and will not be blocked by older proxies.
However, it is important to consider if this header and its function
is required to be understood by the proxy or can be forwarded. If the
header needs to be understood, a feature-tag representing the
functionality MUST be included in the Proxy-Require header. Below are
guidelines for analysis if the header needs to be understood. The
transport header and its parameters also shows that headers that are
extensible and require correct interpretation in the proxy also
require handling rules.</t>
<t>Whether a proxy needs to understand a header is not easy to
determine, as they serve a broad variety of functions. When evaluating
if a header needs to be understood, one can divide the functionality
into three main categories:<list style="hanging">
<t hangText="Media modifying:">The caching and translator proxies
are modifying the actual media and therefore need to understand
also the request directed to the server that affects how the media
is rendered. Thus, this type of proxy needs to also understand the
server side functionality.</t>
<t hangText="Transport modifying:">The access and the security
proxy both need to understand how the transport is performed,
either for opening pinholes or to translate the outer headers,
e.g., IP and UDP.</t>
<t hangText="Non-modifying:">The audit proxy is special in that it
does not modify the messages in other ways than to insert the Via
header. That makes it possible for this type to forward RTSP
messages that contain different types of unknown methods, headers
or header parameters.</t>
</list>Based on the above classification, one should evaluate if the
new functionality requires the Transport modifying type of proxies to
understand it or not.</t>
<t/>
</section>
<section anchor="sec_proxies_cseq"
title="Multiplexing and Demultiplexing of Messages">
<t>RTSP proxies may have to multiplex multiple RTSP sessions from
their clients towards RTSP servers. This requires that RTSP requests
from multiple clients are multiplexed onto a common connection for
requests outgoing to an RTSP server and on the way back the responses
are demultiplexed from the server to per client responses. On the
protocol level this requires that request and response messages are
handled in both ways, requiring that there is a mechanism to correlate
what request/response pair exchanged between proxy and server is
mapped to what client (or client request).</t>
<t>This multiplexing of requests and demultiplexing of responses is
done by using the CSeq header field. The proxy has to rewrite the CSeq
in requests to the server and responses from the server and remember
what CSeq is mapped to what client. The proxy also needs to ensure
that the order of the message related to each client is maintained.
<xref target="sec_CSeq"/> is defining the handling of how requests and
responses are rewritten.</t>
</section>
</section>
<!-- title="Proxies" -->
<section anchor="sec_caching" title="Caching">
<t>In HTTP, request-response pairs are cached. RTSP differs
significantly in that respect. Responses are not cacheable, with the
exception of the presentation description returned by DESCRIBE. (Since
the responses for anything but DESCRIBE and GET_PARAMETER do not return
any data, caching is not really an issue for these requests.) However,
it is desirable for the continuous media data, typically delivered
out-of-band with respect to RTSP, to be cached, as well as the session
description.</t>
<t>On receiving a SETUP or PLAY request, a proxy ascertains whether it
has an up-to-date copy of the continuous media content and its
description. It can determine whether the copy is up-to-date by issuing
a SETUP or DESCRIBE request, respectively, and comparing the
Last-Modified header with that of the cached copy. If the copy is not
up-to-date, it modifies the SETUP transport parameters as appropriate
and forwards the request to the origin server. Subsequent control
commands such as PLAY or PAUSE then pass the proxy unmodified. The proxy
delivers the continuous media data to the client, while possibly making
a local copy for later reuse. The exact allowed behavior of the cache is
given by the cache-response directives described in <xref
target="sec_Cache-Control"/>. A cache MUST answer any DESCRIBE requests
if it is currently serving the stream to the requester, as it is
possible that low-level details of the stream description may have
changed on the origin-server.</t>
<t>Note that an RTSP cache, is of the "cut-through" variety. Rather than
retrieving the whole resource from the origin server, the cache simply
copies the streaming data as it passes by on its way to the client.
Thus, it does not introduce additional latency.</t>
<t>To the client, an RTSP proxy cache appears like a regular media
server. To the media origin server an RTSP proxy cache appears like a
client. Just as an HTTP cache has to store the content type, content
language, and so on for the objects it caches, a media cache has to
store the presentation description. Typically, a cache eliminates all
transport-references (e.g., multicast information) from the presentation
description, since these are independent of the data delivery from the
cache to the client. Information on the encodings remains the same. If
the cache is able to translate the cached media data, it would create a
new presentation description with all the encoding possibilities it can
offer.</t>
<section title=" Validation Model">
<t>When a cache has a stale entry that it would like to use as a
response to a client's request, it first has to check with the origin
server (or possibly an intermediate cache with a fresh response) to
see if its cached entry is still usable. We call this "validating" the
cache entry. Since we do not want to have to pay the overhead of
retransmitting the full response if the cached entry is good, and we
do not want to pay the overhead of an extra round trip if the cached
entry is invalid, the RTSP protocol supports the use of conditional
methods.</t>
<t>The key protocol features for supporting conditional methods are
those concerned with "cache validators." When an origin server
generates a full response, it attaches some sort of validator to it,
which is kept with the cache entry. When a client (user agent or proxy
cache) makes a conditional request for a resource for which it has a
cache entry, it includes the associated validator in the request.</t>
<t>The server then checks that validator against the current validator
for the requested resource, and, if they match (see <xref
target="sec.weak_strong_validators"/>), it responds with a special
status code (usually, 304 (Not Modified)) and no message body.
Otherwise, it returns a full response (including message body). Thus,
we avoid transmitting the full response if the validator matches, and
we avoid an extra round trip if it does not match.</t>
<t>In RTSP, a conditional request looks exactly the same as a normal
request for the same resource, except that it carries a special header
(which includes the validator) that implicitly turns the method
(usually DESCRIBE or SETUP) into a conditional.</t>
<t>The protocol includes both positive and negative senses of
cache-validating conditions. That is, it is possible to request either
that a method be performed if and only if a validator matches or if
and only if no validators match.</t>
<t><list style="hanging">
<t>Note: a response that lacks a validator may still be cached,
and served from cache until it expires, unless this is explicitly
prohibited by a cache-control directive (see <xref
target="sec_Cache-Control"/>). However, a cache cannot do a
conditional retrieval if it does not have a validator for the
resource, which means it will not be refreshable after it
expires.</t>
</list>Media streams that are being adapted based on the transport
capacity between the server and the cache makes caching more
difficult. A server needs to consider how it views caching of media
streams that it adapts and potentially instruct any caches to not
cache such streams.</t>
<section title="Last-Modified Dates ">
<t>The Last-Modified header (<xref target="sec_Last-Modified"/>)
value is often used as a cache validator. In simple terms, a cache
entry is considered to be valid if the content has not been modified
since the Last-Modified value.</t>
</section>
<section title="Message Body Tag Cache Validators">
<t>The MTag response-header field value, a message body tag,
provides for an "opaque" cache validator. This might allow more
reliable validation in situations where it is inconvenient to store
modification dates, where the one-second resolution of RTSP-date
values is not sufficient, or where the origin server wishes to avoid
certain paradoxes that might arise from the use of modification
dates.</t>
<t>Message body tags are described in <xref
target="sec_message-tags"/></t>
</section>
<section anchor="sec.weak_strong_validators"
title="Weak and Strong Validators">
<t>Since both origin servers and caches will compare two validators
to decide if they represent the same or different entities, one
normally would expect that if the message body (i.e., the
presentation description) or any associated message body headers
changes in any way, then the associated validator would change as
well. If this is true, then we call this validator a "strong
validator." We call message body (i.e., the presentation
description) or any associated message body headers an entity for a
better understanding.</t>
<t>However, there might be cases when a server prefers to change the
validator only on semantically significant changes, and not when
insignificant aspects of the entity change. A validator that does
not always change when the resource changes is a "weak
validator."</t>
<t>Message body tags are normally "strong validators," but the
protocol provides a mechanism to tag a message body tag as "weak."
One can think of a strong validator as one that changes whenever the
bits of an entity changes, while a weak value changes whenever the
meaning of an entity changes. Alternatively, one can think of a
strong validator as part of an identifier for a specific entity,
while a weak validator is part of an identifier for a set of
semantically equivalent entities.</t>
<t><list style="hanging">
<t>Note: One example of a strong validator is an integer that is
incremented in stable storage every time an entity is
changed.</t>
<t>An entity's modification time, if represented with one-second
resolution, could be a weak validator, since it is possible that
the resource might be modified twice during a single second.</t>
<t>Support for weak validators is optional. However, weak
validators allow for more efficient caching of equivalent
objects.</t>
</list>A "use" of a validator is either when a client generates a
request and includes the validator in a validating header field, or
when a server compares two validators.</t>
<t>Strong validators are usable in any context. Weak validators are
only usable in contexts that do not depend on exact equality of an
entity. For example, either kind is usable for a conditional
DESCRIBE of a full entity. However, only a strong validator is
usable for a sub-range retrieval, since otherwise the client might
end up with an internally inconsistent entity.</t>
<t>Clients MAY issue DESCRIBE requests with either weak validators
or strong validators. Clients MUST NOT use weak validators in other
forms of requests.</t>
<t>The only function that the RTSP protocol defines on validators is
comparison. There are two validator comparison functions, depending
on whether the comparison context allows the use of weak validators
or not: <list style="symbols">
<t>The strong comparison function: in order to be considered
equal, both validators MUST be identical in every way, and both
MUST NOT be weak.</t>
<t>The weak comparison function: in order to be considered
equal, both validators MUST be identical in every way, but
either or both of them MAY be tagged as "weak" without affecting
the result.</t>
</list>A message body tag is strong unless it is explicitly tagged
as weak.</t>
<t>A Last-Modified time, when used as a validator in a request, is
implicitly weak unless it is possible to deduce that it is strong,
using the following rules: <list style="symbols">
<t>The validator is being compared by an origin server to the
actual current validator for the entity and,</t>
<t>That origin server reliably knows that the associated entity
did not change more than once during the second covered by the
presented validator.</t>
</list>OR</t>
<t><list style="symbols">
<t>The validator is about to be used by a client in an
If-Modified-Since, because the client has a cache entry for the
associated entity, and</t>
<t>That cache entry includes a Date value, which gives the time
when the origin server sent the original response, and</t>
<t>The presented Last-Modified time is at least 60 seconds
before the Date value.</t>
</list>OR</t>
<t><list style="symbols">
<t>The validator is being compared by an intermediate cache to
the validator stored in its cache entry for the entity, and</t>
<t>That cache entry includes a Date value, which gives the time
when the origin server sent the original response, and</t>
<t>The presented Last-Modified time is at least 60 seconds
before the Date value.</t>
</list>This method relies on the fact that if two different
responses were sent by the origin server during the same second, but
both had the same Last-Modified time, then at least one of those
responses would have a Date value equal to its Last-Modified time.
The arbitrary 60- second limit guards against the possibility that
the Date and Last- Modified values are generated from different
clocks, or at somewhat different times during the preparation of the
response. An implementation MAY use a value larger than 60 seconds,
if it is believed that 60 seconds is too short.</t>
<t>If a client wishes to perform a sub-range retrieval on a value
for which it has only a Last-Modified time and no opaque validator,
it MAY do this only if the Last-Modified time is strong in the sense
described here.</t>
</section>
<section anchor="sec.rule_entity_lastmod"
title="Rules for When to Use Message Body Tags and Last-Modified Dates">
<t>We adopt a set of rules and recommendations for origin servers,
clients, and caches regarding when various validator types ought to
be used, and for what purposes.</t>
<t>RTSP origin servers: <list style="symbols">
<t>SHOULD send a message body tag validator unless it is not
feasible to generate one.</t>
<t>MAY send a weak message body tag instead of a strong message
body tag, if performance considerations support the use of weak
message body tags, or if it is unfeasible to send a strong
message body tag.</t>
<t>SHOULD send a Last-Modified value if it is feasible to send
one, unless the risk of a breakdown in semantic transparency
that could result from using this date in an If-Modified-Since
header would lead to serious problems.</t>
</list>In other words, the preferred behavior for an RTSP origin
server is to send both a strong message body tag and a Last-Modified
value.</t>
<t>In order to be legal, a strong message body tag MUST change
whenever the associated entity value changes in any way. A weak
message body tag SHOULD change whenever the associated entity
changes in a semantically significant way.</t>
<t><list style="hanging">
<t>Note: in order to provide semantically transparent caching,
an origin server MUST avoid reusing a specific strong message
body tag value for two different entities, or reusing a specific
weak message body tag value for two semantically different
entities. Cache entries might persist for arbitrarily long
periods, regardless of expiration times, so it might be
inappropriate to expect that a cache will never again attempt to
validate an entry using a validator that it obtained at some
point in the past.</t>
</list></t>
<t>RTSP clients: <list style="symbols">
<t>If a message body tag has been provided by the origin server,
MUST use that message body tag in any cache-conditional request
(using If-Match or If-None-Match).</t>
<t>If only a Last-Modified value has been provided by the origin
server, SHOULD use that value in non-subrange cache-conditional
requests (using If-Modified-Since).</t>
<t>If both a message body tag and a Last-Modified value have
been provided by the origin server, SHOULD use both validators
in cache-conditional requests.</t>
</list>An RTSP origin server, upon receiving a conditional request
that includes both a Last-Modified date (e.g., in an
If-Modified-Since header) and one or more message body tags (e.g.,
in an If-Match, If-None-Match, or If-Range header field) as cache
validators, MUST NOT return a response status of 304 (Not Modified)
unless doing so is consistent with all of the conditional header
fields in the request.</t>
<t><list style="hanging">
<t>Note: The general principle behind these rules is that RTSP
servers and clients should transmit as much non-redundant
information as is available in their responses and requests.
RTSP systems receiving this information will make the most
conservative assumptions about the validators they receive.</t>
</list></t>
</section>
<section title="Non-validating Conditionals">
<t>The principle behind message body tags is that only the service
author knows the semantics of a resource well enough to select an
appropriate cache validation mechanism, and the specification of any
validator comparison function more complex than octet-equality would
open up a can of worms. Thus, comparisons of any other headers are
never used for purposes of validating a cache entry.</t>
</section>
</section>
<section anchor="sec.chache_invalidation"
title="Invalidation After Updates or Deletions">
<t>The effect of certain methods performed on a resource at the origin
server might cause one or more existing cache entries to become non-
transparently invalid. That is, although they might continue to be
"fresh," they do not accurately reflect what the origin server would
return for a new request on that resource.</t>
<t>There is no way for the RTSP protocol to guarantee that all such
cache entries are marked invalid. For example, the request that caused
the change at the origin server might not have gone through the proxy
where a cache entry is stored. However, several rules help reduce the
likelihood of erroneous behavior.</t>
<t>In this section, the phrase "invalidate an entity" means that the
cache will either remove all instances of that entity from its
storage, or will mark these as "invalid" and in need of a mandatory
revalidation before they can be returned in response to a subsequent
request.</t>
<t>Some RTSP methods MUST cause a cache to invalidate an entity. This
is either the entity referred to by the Request-URI, or by the
Location or Content-Location headers (if present). These methods are:
<list style="symbols">
<t>DESCRIBE</t>
<t>SETUP</t>
</list>In order to prevent denial of service attacks, an
invalidation based on the URI in a Location or Content-Location header
MUST only be performed if the host part is the same as in the
Request-URI.</t>
<t>A cache that passes through requests for methods it does not
understand SHOULD invalidate any entities referred to by the
Request-URI.</t>
</section>
</section>
<!-- title="Caching" -->
<section anchor="sec_status" title="Status Code Definitions">
<t>Where applicable, HTTP status [H10] codes are reused. See <xref
target="tab_status"/> in <xref target="sec_status-line"/> for a listing
of which status codes may be returned by which requests. All error
messages, 4xx and 5xx MAY return a body containing further information
about the error.</t>
<section title="Success 1xx">
<section title="100 Continue">
<t>The client SHOULD continue with its request. This interim
response is used to inform the client that the initial part of the
request has been received and has not yet been rejected by the
server. The client SHOULD continue by sending the remainder of the
request or, if the request has already been completed, ignore this
response. The server MUST send a final response after the request
has been completed.</t>
</section>
</section>
<section title="Success 2xx">
<t>This class of status code indicates that the client's request was
successfully received, understood, and accepted.</t>
<section title="200 OK">
<t>The request has succeeded. The information returned with the
response is dependent on the method used in the request.</t>
</section>
</section>
<section anchor="sec_status-redirect" title="Redirection 3xx">
<t>The notation "3xx" indicates response codes from 300 to 399
inclusive which are meant for redirection. The response code 304 is
excluded, as it is not used for redirection and instead the "3rr"
notation is used. The 304 response code appears here, rather than a
2xx response code which would have been appropriate, this as 304 has
been used also in <xref target="RFC2326">RTSP 1.0</xref>.</t>
<t>Within RTSP, redirection may be used for load balancing or
redirecting stream requests to a server topologically closer to the
client. Mechanisms to determine topological proximity are beyond the
scope of this specification.</t>
<t>A 3rr code MAY be used to respond to any request. It is RECOMMENDED
that they are used if necessary before a session is established, i.e.,
in response to DESCRIBE or SETUP. However, in cases where a server is
not able to send a REDIRECT request to the client, the server MAY need
to resort to using 3rr responses to inform a client with an
established session about the need for redirecting the session. If a
3rr response is received for a request in relation to an established
session, the client SHOULD send a TEARDOWN request for the session,
and MAY reestablish the session using the resource indicated by the
Location.</t>
<t>If the Location header is used in a response it MUST contain an
absolute URI pointing out the media resource the client is redirected
to, the URI MUST NOT only contain the host name.</t>
<section title="300">
<t>This response code is not used in RTSP 2.0. In the event that an
unknown 3rr status code is received, the agent SHOULD behave as if a
302 response code had been <xref
target="sec_302">received</xref>.</t>
</section>
<section title="301 Moved Permanently">
<t>The requested resource is moved permanently and resides now at
the URI given by the Location header. The user client SHOULD
redirect automatically to the given URI. This response MUST NOT
contain a message-body. The Location header MUST be included in the
response.</t>
</section>
<section anchor="sec_302" title="302 Found">
<t>The requested resource resides temporarily at the URI given by
the Location header. The Location header MUST be included in the
response. This response is intended to be used for many types of
temporary redirects; e.g., load balancing. It is RECOMMENDED that
the server set the reason phrase to something more meaningful than
"Found" in these cases. The user client SHOULD redirect
automatically to the given URI. This response MUST NOT contain a
message-body.</t>
<t>This example shows a client being redirected to a different
server:</t>
<figure>
<artwork><![CDATA[
C->S: SETUP rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 2
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
Accept-Ranges: npt, smpte, clock
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 302 Try Other Server
CSeq: 2
Location: rtsp://s2.example.com:8001/fizzle/foo]]></artwork>
</figure>
</section>
<section title="303 See Other">
<t>This status code MUST NOT be used in RTSP 2.0. However, it was
allowed in <xref target="RFC2326">RTSP 1.0</xref>.</t>
</section>
<section title="304 Not Modified">
<t>If the client has performed a conditional DESCRIBE or SETUP (see
<xref target="sec_If-Modified-Since"/>) and the requested resource
has not been modified, the server SHOULD send a 304 response. This
response MUST NOT contain a message-body.</t>
<t>The response MUST include the following header fields: <list
hangIndent="3" style="symbols">
<t>Date</t>
<t>MTag and/or Content-Location, if the header(s) would have
been sent in a 200 response to the same request.</t>
<t>Expires and Cache-Control if the field-value might differ
from that sent in any previous response for the same
variant.</t>
<!-- , and/or Vary, -->
</list></t>
<t>This response is independent for the DESCRIBE and SETUP requests.
That is, a 304 response to DESCRIBE does NOT imply that the resource
content is unchanged (only the session description) and a 304
response to SETUP does NOT imply that the resource description is
unchanged. The MTag and If-Match headers may be used to link the
DESCRIBE and SETUP in this manner.</t>
</section>
<section title="305 Use Proxy">
<t>The requested resource MUST be accessed through the proxy given
by the Location field. The Location field gives the URI of the
proxy. The recipient is expected to repeat this single request via
the proxy. 305 responses MUST only be generated by origin
servers.</t>
</section>
</section>
<section title="Client Error 4xx">
<section title="400 Bad Request">
<t>The request could not be understood by the server due to
malformed syntax. The client SHOULD NOT repeat the request without
modifications. If the request does not have a CSeq header, the
server MUST NOT include a CSeq in the response.</t>
</section>
<section anchor="sec_error401" title="401 Unauthorized">
<t>The request requires user authentication. The response MUST
include a <xref target="sec_WWW-Authenticate">WWW-Authenticate
header</xref> field containing a challenge applicable to the
requested resource. The client MAY repeat the request with a
suitable Authorization header field. If the request already included
Authorization credentials, then the 401 response indicates that
authorization has been refused for those credentials. If the 401
response contains the same challenge as the prior response, and the
user agent has already attempted authentication at least once, then
the user SHOULD be presented the message body that was given in the
response, since that message body might include relevant diagnostic
information. HTTP access authentication is explained in <xref
target="RFC2617"/>.</t>
</section>
<section title="402 Payment Required">
<t>This code is reserved for future use.</t>
</section>
<section title="403 Forbidden">
<t>The server understood the request, but is refusing to fulfill it.
Authorization will not help and the request SHOULD NOT be repeated.
If the server wishes to make public why the request has not been
fulfilled, it SHOULD describe the reason for the refusal in the
message body. If the server does not wish to make this information
available to the client, the status code 404 (Not Found) can be used
instead.</t>
</section>
<section title="404 Not Found">
<t>The server has not found anything matching the Request-URI. No
indication is given of whether the condition is temporary or
permanent. The 410 (Gone) status code SHOULD be used if the server
knows, through some internally configurable mechanism, that an old
resource is permanently unavailable and has no forwarding address.
This status code is commonly used when the server does not wish to
reveal exactly why the request has been refused, or when no other
response is applicable.</t>
</section>
<section title="405 Method Not Allowed">
<t>The method specified in the request is not allowed for the
resource identified by the Request-URI. The response MUST include an
Allow header containing a list of valid methods for the requested
resource. This status code is also to be used if a request attempts
to use a method not indicated during SETUP.</t>
</section>
<section title="406 Not Acceptable">
<t>The resource identified by the request is only capable of
generating response message bodies which have content
characteristics not acceptable according to the Accept headers sent
in the request.</t>
<t>The response SHOULD include a message body containing a list of
available message body characteristics and location(s) from which
the user or user agent can choose the one most appropriate. The
message body format is specified by the media type given in the
Content-Type header field. Depending upon the format and the
capabilities of the user agent, selection of the most appropriate
choice MAY be performed automatically. However, this specification
does not define any standard for such automatic selection.</t>
<t>If the response could be unacceptable, a user agent SHOULD
temporarily stop receipt of more data and query the user for a
decision on further actions.</t>
</section>
<section title="407 Proxy Authentication Required">
<t>This code is similar to <xref target="sec_error401">401
(Unauthorized)</xref>, but indicates that the client must first
authenticate itself with the proxy. The proxy MUST return a <xref
target="sec_Proxy-Authenticate">Proxy-Authenticate header
field</xref> containing a challenge applicable to the proxy for the
requested resource.</t>
</section>
<section title="408 Request Timeout">
<t>The client did not produce a request within the time that the
server was prepared to wait. The client MAY repeat the request
without modifications at any later time.</t>
</section>
<section title="410 Gone">
<t>The requested resource is no longer available at the server and
the forwarding address is not known. This condition is expected to
be considered permanent. If the server does not know, or has no
facility to determine, whether or not the condition is permanent,
the status code 404 (Not Found) SHOULD be used instead. This
response is cacheable unless indicated otherwise.</t>
<t>The 410 response is primarily intended to assist the task of
repository maintenance by notifying the recipient that the resource
is intentionally unavailable and that the server owners desire that
remote links to that resource be removed. Such an event is common
for limited-time, promotional services and for resources belonging
to individuals no longer working at the server's site. It is not
necessary to mark all permanently unavailable resources as "gone" or
to keep the mark for any length of time -- that is left to the
discretion of the owner of the server.</t>
</section>
<section title="411 Length Required">
<t>This error code is not defined for RTSP. This as the use of <xref
target="sec_Content-Length">Content-Length</xref> is always required
when message bodies are included in RTSP messages.</t>
</section>
<section title="412 Precondition Failed">
<t>The precondition given in one or more of the 'if-' request-header
fields evaluated to false when it was tested on the server. See
these sections for the 'if-' headers: If-Match <xref
target="sec_If-Match"/>, If-Modified-Since <xref
target="sec_If-Modified-Since"/>, and If-None-Match <xref
target="sec_If-None-Match"/>. This response code allows the client
to place preconditions on the current resource meta information
(header field data) and thus prevent the requested method from being
applied to a resource other than the one intended.</t>
</section>
<section title="413 Request Message Body Too Large">
<t>The server is refusing to process a request because the request
message body is larger than the server is willing or able to
process. The server MAY close the connection to prevent the client
from continuing the request.</t>
<t>If the condition is temporary, the server SHOULD include a Retry-
After header field to indicate that it is temporary and after what
time the client MAY try again.</t>
</section>
<section title="414 Request-URI Too Long">
<t>The server is refusing to service the request because the
Request-URI is longer than the server is willing to interpret. This
rare condition is only likely to occur when a client has used a
request with long query information, when the client has descended
into a URI "black hole" of redirection (e.g., a redirected URI
prefix that points to a suffix of itself), or when the server is
under attack by a client attempting to exploit security holes
present in some servers using fixed-length buffers for reading or
manipulating the Request-URI.</t>
</section>
<section title="415 Unsupported Media Type">
<t>The server is refusing to service the request because the message
body of the request is in a format not supported by the requested
resource for the requested method.</t>
</section>
<section title="451 Parameter Not Understood">
<t>The recipient of the request does not support one or more
parameters contained in the request. When returning this error
message the sender SHOULD return a message body containing the
offending parameter(s).</t>
</section>
<section title="452 reserved">
<t>This status code MUST NOT be used in RTSP 2.0. However, it was
allowed in <xref target="RFC2326">RTSP 1.0</xref>.</t>
</section>
<section title="453 Not Enough Bandwidth">
<t>The request was refused because there was insufficient bandwidth.
This may, for example, be the result of a resource reservation
failure.</t>
</section>
<section title="454 Session Not Found">
<t>The RTSP session identifier in the Session header is missing,
invalid, or has timed out.</t>
</section>
<section title="455 Method Not Valid in This State">
<t>The client or server cannot process this request in its current
state. The response MUST contain an Allow header to make error
recovery possible.</t>
</section>
<section title="456 Header Field Not Valid for Resource">
<t>The server could not act on a required request-header. For
example, if PLAY contains the Range header field but the stream does
not allow seeking. This error message may also be used for
specifying when the time format in Range is impossible for the
resource. In that case the Accept-Ranges header MUST be returned to
inform the client of which format(s) that are allowed.</t>
</section>
<section title="457 Invalid Range">
<t>The Range value given is out of bounds, e.g., beyond the end of
the presentation.</t>
</section>
<section title="458 Parameter Is Read-Only">
<t>The parameter to be set by SET_PARAMETER can be read but not
modified. When returning this error message the sender SHOULD return
a message body containing the offending parameter(s).</t>
</section>
<section anchor="sec_error459"
title="459 Aggregate Operation Not Allowed">
<t>The requested method may not be applied on the URI in question
since it is an aggregate (presentation) URI. The method may be
applied on a media URI.</t>
</section>
<section anchor="sec_error460"
title="460 Only Aggregate Operation Allowed">
<t>The requested method may not be applied on the URI in question
since it is not an aggregate control (presentation) URI. The method
may be applied on the aggregate control URI.</t>
</section>
<section anchor="sec_error461" title="461 Unsupported Transport">
<t>The Transport field did not contain a supported transport
specification.</t>
</section>
<section title="462 Destination Unreachable">
<t>The data transmission channel could not be established because
the client address could not be reached. This error will most likely
be the result of a client attempt to place an invalid dest_addr
parameter in the Transport field.</t>
</section>
<section title="463 Destination Prohibited">
<t>The data transmission channel was not established because the
server prohibited access to the client address. This error is most
likely the result of a client attempt to redirect media traffic to
another destination with a dest_addr parameter in the Transport
header.</t>
</section>
<section anchor="sec_error464"
title="464 Data Transport Not Ready Yet">
<t>The data transmission channel to the media destination is not yet
ready for carrying data. However, the responding agent still expects
that the data transmission channel will be established at some point
in time. Note, however, that this may result in a permanent failure
like 462 "Destination Unreachable".</t>
<t>An example when this error may occur is in the case a client
sends a PLAY request to a server prior to ensuring that the TCP
connections negotiated for carrying media data was successfully
established (In violation of this specification). The server would
use this error code to indicate that the requested action could not
be performed due to the failure of completing the connection
establishment.</t>
</section>
<section anchor="sec_error465" title="465 Notification Reason Unknown">
<t>This indicates that the client has received a <xref
target="sec_PLAY_NOTIFY">PLAY_NOTIFY</xref> with a <xref
target="sec_Notify-Reason">Notify-Reason header</xref> unknown to
the client.</t>
</section>
<section title="466 Key Management Error">
<t>This indicates that there has been an error in a Key Management
function used in conjunction with a request. For example usage of
<xref target="RFC3830">MIKEY</xref> according to <xref
target="sec-mikey"/> may result in this error.</t>
</section>
<section title="470 Connection Authorization Required">
<t>The secured connection attempt needs user or client authorization
before proceeding. The next hop's certificate is included in this
response in the Accept-Credentials header.</t>
</section>
<section title="471 Connection Credentials not accepted">
<t>When performing a secure connection over multiple connections, an
intermediary has refused to connect to the next hop and carry out
the request due to unacceptable credentials for the used policy.</t>
</section>
<section title="472 Failure to establish secure connection">
<t>A proxy fails to establish a secure connection to the next hop
RTSP agent. This is primarily caused by a fatal failure at the TLS
handshake, for example due to server not accepting any cipher
suites.</t>
</section>
</section>
<section title="Server Error 5xx">
<t>Response status codes beginning with the digit "5" indicate cases
in which the server is aware that it has erred or is incapable of
performing the request The server SHOULD include a message body
containing an explanation of the error situation, and whether it is a
temporary or permanent condition. User agents SHOULD display any
included message body to the user. These response codes are applicable
to any request method.</t>
<section title="500 Internal Server Error">
<t>The server encountered an unexpected condition which prevented it
from fulfilling the request.</t>
</section>
<section title="501 Not Implemented">
<t>The server does not support the functionality required to fulfill
the request. This is the appropriate response when the server does
not recognize the request method and is not capable of supporting it
for any resource.</t>
</section>
<section title="502 Bad Gateway">
<t>The server, while acting as a gateway or proxy, received an
invalid response from the upstream server it accessed in attempting
to fulfill the request.</t>
</section>
<section anchor="sec_error_503" title="503 Service Unavailable">
<t>The server is currently unable to handle the request due to a
temporary overloading or maintenance of the server. The implication
is that this is a temporary condition which will be alleviated after
some delay. If known, the length of the delay MAY be indicated in a
Retry-After header. If no Retry-After is given, the client SHOULD
handle the response as it would for a 500 response. The client MUST
honor the length, if given in the Retry-After header.</t>
<t><list hangIndent="6" style="hanging">
<t>Note: The existence of the 503 status code does not imply
that a server must use it when becoming overloaded. Some servers
may wish to simply refuse the connection.</t>
</list>The response scope is dependent on the Request. If the
request was in relation to an existing RTSP session, the scope of
the overload response is to this individual RTSP session. If the
request was non-session specific or intended to form a RTSP session
it applies to the RTSP server identified by the host name in the
request URI.</t>
</section>
<section title="504 Gateway Timeout">
<t>The server, while acting as a proxy, did not receive a timely
response from the upstream server specified by the URI or some other
auxiliary server (e.g., DNS) it needed to access in attempting to
complete the request.</t>
</section>
<section title="505 RTSP Version Not Supported">
<t>The server does not support, or refuses to support, the RTSP
protocol version that was used in the request message. The server is
indicating that it is unable or unwilling to complete the request
using the same major version as the client other than with this
error message. The response SHOULD contain a message body describing
why that version is not supported and what other protocols are
supported by that server.</t>
</section>
<section title="551 Option not supported">
<t>A feature-tag given in the Require or the Proxy-Require fields
was not supported. The Unsupported header MUST be returned stating
the feature for which there is no support.</t>
</section>
<section title="553 Proxy Unavailable">
<t>The proxy is currently unable to handle the request due to a
temporary overloading or maintenance of the proxy. The implication
is that this is a temporary condition which will be alleviated after
some delay. If known, the length of the delay MAY be indicated in a
Retry-After header. If no Retry-After is given, the client SHOULD
handle the response as it would for a 500 response. The client MUST
honor the length, if given in the Retry-After header.</t>
<t><list hangIndent="6" style="hanging">
<t>Note: The existence of the 553 status code does not imply
that a proxy must use it when becoming overloaded. Some proxies
may wish to simply refuse the connection.</t>
</list>The response scope is dependent on the Request. If the
request was in relation to an existing RTSP session, the scope of
the overload response is to this individual RTSP session. If the
request was non-session specific or intended to form a RTSP session
it applies to all such requests to this proxy.</t>
</section>
</section>
</section>
<section anchor="sec_headers" title="Header Field Definitions">
<texttable anchor="tab_methods2"
title="Overview of RTSP methods, their direction, and what objects (P: presentation, S: stream) they operate on. Body notes if a method is allowed to carry body and in which direction, R = Request, r=response. Note: All error messages for statuses 4xx and 5xx are allowed to carry a body">
<preamble/>
<ttcol align="left">method</ttcol>
<ttcol align="left">direction</ttcol>
<ttcol align="left">object</ttcol>
<ttcol align="left">acronym</ttcol>
<ttcol align="left">Body</ttcol>
<c>DESCRIBE</c>
<c>C -> S</c>
<c>P,S</c>
<c>DES</c>
<c>r</c>
<c>GET_PARAMETER</c>
<c>C -> S, S -> C</c>
<c>P,S</c>
<c>GPR</c>
<c>R,r</c>
<c>OPTIONS</c>
<c>C -> S, S -> C</c>
<c>P,S</c>
<c>OPT</c>
<c/>
<c>PAUSE</c>
<c>C -> S</c>
<c>P,S</c>
<c>PSE</c>
<c/>
<c>PLAY</c>
<c>C -> S</c>
<c>P,S</c>
<c>PLY</c>
<c/>
<c>PLAY_NOTIFY</c>
<c>S -> C</c>
<c>P,S</c>
<c>PNY</c>
<c>R</c>
<c>REDIRECT</c>
<c>S -> C</c>
<c>P,S</c>
<c>RDR</c>
<c/>
<c>SETUP</c>
<c>C -> S</c>
<c>S</c>
<c>STP</c>
<c/>
<c>SET_PARAMETER</c>
<c>C -> S, S -> C</c>
<c>P,S</c>
<c>SPR</c>
<c>R,r</c>
<c>TEARDOWN</c>
<c>C -> S</c>
<c>P,S</c>
<c>TRD</c>
<c/>
<c><!-- TEARDOWN --></c>
<c>S -> C</c>
<c>P</c>
<c>TRD</c>
<c/>
</texttable>
<t>The general syntax for header fields is covered in <xref
target="sec_message-headers"/>. This section lists the full set of
header fields along with notes on meaning, and usage. The syntax
definition for header fields are present in <xref
target="sec_syntax-prot-header"/>. Throughout this section, we use
[HX.Y] to reference Section X.Y of the current HTTP/1.1 specification
RFC 2616 <xref target="RFC2616"/>. Examples of each header field are
given.</t>
<t>Information about header fields in relation to methods and proxy
processing is summarized in <xref target="tab_headers1a"/>, <xref
target="tab_headers1b"/>, <xref target="tab_headers2a"/>, and <xref
target="tab_headers2b"/>.</t>
<t>The "where" column describes the request and response types in which
the header field can be used. Values in this column are: <list
hangIndent="6" style="hanging">
<t hangText="R:">header field may only appear in requests;</t>
<t hangText="r:">header field may only appear in responses;</t>
<t hangText="2xx, 4xx, etc.:">A numerical value or range indicates
response codes with which the header field can be used;</t>
<t hangText="c:">header field is copied from the request to the
response.</t>
<t hangText="G:">header field is a general-header and may be present
in both requests and responses.</t>
</list></t>
<t>Note: General headers does not always use the "G" value in the where
column. This is due to differencies when the header may be applied in
requests compared to responses. When such differencies exist they are
expressed using two differet rows, one with where being "R" and one with
it being "r".</t>
<t>The "proxy" column describes the operations a proxy may perform on a
header field. An empty proxy column indicates that the proxy MUST NOT do
any changes to that header, all allowed operations are explicitly
stated: <list hangIndent="6" style="hanging">
<t hangText="a:">A proxy can add or concatenate the header field if
not present.</t>
<t hangText="m:">A proxy can modify an existing header field
value.</t>
<t hangText="d:">A proxy can delete a header field value.</t>
<t hangText="r:">A proxy needs to be able to read the header field,
and thus this header field cannot be encrypted.</t>
</list></t>
<t>The rest of the columns relate to the presence of a header field in a
method. The method names when abbreviated, are according to <xref
target="tab_methods2"/>: <list hangIndent="6" style="hanging">
<t hangText="c:">Conditional; requirements on the header field
depend on the context of the message.</t>
<t hangText="m:">The header field is mandatory.</t>
<t hangText="m*:">The header field SHOULD be sent, but
clients/servers need to be prepared to receive messages without that
header field.</t>
<t hangText="o:">The header field is optional.</t>
<t hangText="*:">The header field MUST be present if the message
body is not empty. See <xref target="sec_Content-Length"/>, <xref
target="sec_Content-Type"/> and <xref target="sec_message-body"/>
for details.</t>
<t hangText="-:">The header field is not applicable.</t>
</list></t>
<t>"Optional" means that a Client/Server MAY include the header field in
a request or response. The Client/Server behavior when receiving such
headers varies, for some it may ignore the header field, in other cases
it is a request to process the header. This is regulated by the method
and header descriptions. Example of headers that require processing are
the Require and Proxy-Require header fields discussed in <xref
target="sec_Require"/> and <xref target="sec_Proxy-Require"/>. A
"mandatory" header field MUST be present in a request, and MUST be
understood by the Client/Server receiving the request. A mandatory
response-header field MUST be present in the response, and the header
field MUST be understood by the Client/Server processing the response.
"Not applicable" means that the header field MUST NOT be present in a
request. If one is placed in a request by mistake, it MUST be ignored by
the Client/Server receiving the request. Similarly, a header field
labeled "not applicable" for a response means that the Client/Server
MUST NOT place the header field in the response, and the Client/Server
MUST ignore the header field in the response.</t>
<t>An RTSP agent MUST ignore extension headers that are not
understood.</t>
<t>The From and Location header fields contain a URI. If the URI
contains a comma, or semicolon, the URI MUST be enclosed in double
quotes ("). Any URI parameters are contained within these quotes. If the
URI is not enclosed in double quote, any semicolon-delimited parameters
are header-parameters, not URI parameters.</t>
<texttable anchor="tab_headers1a"
title="Overview of RTSP header fields (A-L) related to methods DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.">
<preamble/>
<ttcol align="left">Header</ttcol>
<ttcol align="left">Where</ttcol>
<ttcol align="left">Proxy</ttcol>
<ttcol align="left">DES</ttcol>
<ttcol align="left">OPT</ttcol>
<ttcol align="left">STP</ttcol>
<ttcol align="left">PLY</ttcol>
<ttcol align="left">PSE</ttcol>
<ttcol align="left">TRD</ttcol>
<c>Accept</c>
<c>R</c>
<c/>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Accept-Credentials</c>
<c>R</c>
<c>rm</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Accept-Encoding</c>
<c>R</c>
<c>r</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Accept-Language</c>
<c>R</c>
<c>r</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Accept-Ranges</c>
<c>G</c>
<c>r</c>
<c>-</c>
<c>-</c>
<c>m</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Accept-Ranges</c>
<c>456</c>
<c>r</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>m</c>
<c>-</c>
<c>-</c>
<c>Allow</c>
<c>r</c>
<c>am</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Allow</c>
<c>405</c>
<c>am</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>Authentication-Info</c>
<c>r</c>
<c/>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o/-</c>
<c>Authorization</c>
<c>R</c>
<c/>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Bandwidth</c>
<c>R</c>
<c/>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Blocksize</c>
<c>R</c>
<c/>
<c>o</c>
<c>-</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Cache-Control</c>
<c>G</c>
<c>r</c>
<c>o</c>
<c>-</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Connection</c>
<c>G</c>
<c>ad</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Connection-Credentials</c>
<c>470,407</c>
<c>ar</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Content-Base</c>
<c>r</c>
<c/>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Content-Base</c>
<c>4xx,5xx</c>
<c/>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Content-Encoding</c>
<c>R</c>
<c>r</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Content-Encoding</c>
<c>r</c>
<c>r</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Content-Encoding</c>
<c>4xx,5xx</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Content-Language</c>
<c>R</c>
<c>r</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Content-Language</c>
<c>r</c>
<c>r</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Content-Language</c>
<c>4xx,5xx</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Content-Length</c>
<c>r</c>
<c>r</c>
<c>*</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Content-Length</c>
<c>4xx,5xx</c>
<c>r</c>
<c>*</c>
<c>*</c>
<c>*</c>
<c>*</c>
<c>*</c>
<c>*</c>
<c>Content-Location</c>
<c>r</c>
<c>r</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Content-Location</c>
<c>4xx,5xx</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Content-Type</c>
<c>r</c>
<c>r</c>
<c>*</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Content-Type</c>
<c>4xx,5xx</c>
<c>ar</c>
<c>*</c>
<c>*</c>
<c>*</c>
<c>*</c>
<c>*</c>
<c>*</c>
<c>CSeq</c>
<c>Gc</c>
<c>rm</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>Date</c>
<c>G</c>
<c>am</c>
<c>o/*</c>
<c>o/*</c>
<c>o/*</c>
<c>o/*</c>
<c>o/*</c>
<c>o/*</c>
<c>Expires</c>
<c>r</c>
<c>r</c>
<c>o</c>
<c>-</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>From</c>
<c>R</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>If-Match</c>
<c>R</c>
<c>r</c>
<c>-</c>
<c>-</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>If-Modified-Since</c>
<c>R</c>
<c>r</c>
<c>o</c>
<c>-</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>If-None-Match</c>
<c>R</c>
<c>r</c>
<c>o</c>
<c>-</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Last-Modified</c>
<c>r</c>
<c>r</c>
<c>o</c>
<c>-</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Location</c>
<c>3rr</c>
<c/>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
</texttable>
<texttable anchor="tab_headers1b"
title="Overview of RTSP header fields (M-W) related to methods DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.">
<preamble/>
<ttcol align="left">Header</ttcol>
<ttcol align="left">Where</ttcol>
<ttcol align="left">Proxy</ttcol>
<ttcol align="left">DES</ttcol>
<ttcol align="left">OPT</ttcol>
<ttcol align="left">STP</ttcol>
<ttcol align="left">PLY</ttcol>
<ttcol align="left">PSE</ttcol>
<ttcol align="left">TRD</ttcol>
<c>Media- Properties</c>
<c>G</c>
<c/>
<c>-</c>
<c>-</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>-</c>
<c>Media-Range</c>
<c>G</c>
<c/>
<c>-</c>
<c>-</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>-</c>
<c>MTag</c>
<c>r</c>
<c>r</c>
<c>o</c>
<c>-</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Pipelined-Requests</c>
<c>G</c>
<c>amdr</c>
<c>-</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Proxy- Authenticate</c>
<c>407</c>
<c>amr</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>Proxy-Authentication-Info</c>
<c>r</c>
<c>amdr</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o/-</c>
<c>Proxy- Authorization</c>
<c>R</c>
<c>rd</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Proxy- Require</c>
<c>R</c>
<c>ar</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Proxy- Require</c>
<c>r</c>
<c>r</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>Proxy- Supported</c>
<c>R</c>
<c>amr</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>Proxy- Supported</c>
<c>r</c>
<c/>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>Public</c>
<c>r</c>
<c>amr</c>
<c>-</c>
<c>m</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Public</c>
<c>501</c>
<c>amr</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>Range</c>
<c>R</c>
<c/>
<c>-</c>
<c>-</c>
<c>-</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Range</c>
<c>r</c>
<c/>
<c>-</c>
<c>-</c>
<c>c</c>
<c>m</c>
<c>m</c>
<c>-</c>
<c>Referrer</c>
<c>R</c>
<c/>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Request- Status</c>
<c>R</c>
<c/>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Require</c>
<c>R</c>
<c/>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Retry-After</c>
<c>3rr,503,553</c>
<c/>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>Retry-After</c>
<c>413</c>
<c/>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>RTP-Info</c>
<c>r</c>
<c/>
<c>-</c>
<c>-</c>
<c>c</c>
<c>c</c>
<c>-</c>
<c>-</c>
<c>Scale</c>
<c>R</c>
<c>r</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Scale</c>
<c>r</c>
<c>amr</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>c</c>
<c>-</c>
<c>-</c>
<c>Seek-Style</c>
<c>R</c>
<c/>
<c>-</c>
<c>-</c>
<c>-</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Seek-Style</c>
<c>r</c>
<c/>
<c>-</c>
<c>-</c>
<c>-</c>
<c>m</c>
<c>-</c>
<c>-</c>
<c>Server</c>
<c>R</c>
<c>r</c>
<c>-</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>o</c>
<c>Server</c>
<c>r</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Session</c>
<c>R</c>
<c>r</c>
<c>-</c>
<c>o</c>
<c>o</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>Session</c>
<c>r</c>
<c>r</c>
<c>-</c>
<c>c</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>o</c>
<c>Speed</c>
<c>R</c>
<c>admr</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Speed</c>
<c>r</c>
<c>admr</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>c</c>
<c>-</c>
<c>-</c>
<c>Supported</c>
<c>R</c>
<c>amr</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Supported</c>
<c>r</c>
<c>amr</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>Terminate-Reason</c>
<c>R</c>
<c>r</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Timestamp</c>
<c>R</c>
<c>admr</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Timestamp</c>
<c>c</c>
<c>admr</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>Transport</c>
<c>G</c>
<c>mr</c>
<c>-</c>
<c>-</c>
<c>m</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Unsupported</c>
<c>r</c>
<c/>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>User-Agent</c>
<c>R</c>
<c/>
<c>m*</c>
<c>m*</c>
<c>m*</c>
<c>m*</c>
<c>m*</c>
<c>m*</c>
<!-- <c>Vary</c>
<c>r</c>
<c/>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>c</c>
-->
<c>Via</c>
<c>R</c>
<c>amr</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Via</c>
<c>c</c>
<c>dr</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>WWW- Authenticate</c>
<c>401</c>
<c/>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
</texttable>
<texttable anchor="tab_headers2a"
title="Overview of RTSP header fields (A-P) related to methods GET_PARAMETER, SET_PARAMETER, REDIRECT, and PLAY_NOTIFY.">
<preamble/>
<ttcol align="left">Header</ttcol>
<ttcol align="left">Where</ttcol>
<ttcol align="left">Proxy</ttcol>
<ttcol align="left">GPR</ttcol>
<ttcol align="left">SPR</ttcol>
<ttcol align="left">RDR</ttcol>
<ttcol align="left">PNY</ttcol>
<c>Accept</c>
<c>R</c>
<c>arm</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Accept-Credentials</c>
<c>R</c>
<c>rm</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>Accept-Encoding</c>
<c>R</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>Accept-Language</c>
<c>R</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>Accept-Ranges</c>
<c>G</c>
<c>rm</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Allow</c>
<c>405</c>
<c>amr</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>-</c>
<c>Authentication-Info</c>
<c>r</c>
<c/>
<c>o/-</c>
<c>o/-</c>
<c>-</c>
<c>-</c>
<c>Authorization</c>
<c>R</c>
<c/>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>Bandwidth</c>
<c>R</c>
<c/>
<c>-</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Blocksize</c>
<c>R</c>
<c/>
<c>-</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Cache-Control</c>
<c>G</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Connection</c>
<c>G</c>
<c/>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Connection-Credentials</c>
<c>470,407</c>
<c>ar</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>Content-Base</c>
<c>R</c>
<c/>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Content-Base</c>
<c>r</c>
<c/>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Content-Base</c>
<c>4xx,5xx</c>
<c/>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Content-Encoding</c>
<c>R</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Content-Encoding</c>
<c>r</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Content-Encoding</c>
<c>4xx,5xx</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Content-Language</c>
<c>R</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Content-Language</c>
<c>r</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Content-Language</c>
<c>4xx,5xx</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Content-Length</c>
<c>R</c>
<c>r</c>
<c>*</c>
<c>*</c>
<c>-</c>
<c>-</c>
<c>Content-Length</c>
<c>r</c>
<c>r</c>
<c>*</c>
<c>*</c>
<c>-</c>
<c>-</c>
<c>Content-Length</c>
<c>4xx,5xx</c>
<c>r</c>
<c>*</c>
<c>*</c>
<c>*</c>
<c>*</c>
<c>Content-Location</c>
<c>R</c>
<c/>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Content-Location</c>
<c>r</c>
<c/>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Content-Location</c>
<c>4xx,5xx</c>
<c/>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Content-Type</c>
<c>R</c>
<c/>
<c>*</c>
<c>*</c>
<c>-</c>
<c>-</c>
<c>Content-Type</c>
<c>r</c>
<c/>
<c>*</c>
<c>*</c>
<c>-</c>
<c>-</c>
<c>Content-Type</c>
<c>4xx,5xx</c>
<c/>
<c>*</c>
<c>*</c>
<c>*</c>
<c>*</c>
<c>CSeq</c>
<c>R,c</c>
<c>mr</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>Date</c>
<c>R</c>
<c>a</c>
<c>o</c>
<c>o</c>
<c>m</c>
<c>o</c>
<c>Date</c>
<c>r</c>
<c>am</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Expires</c>
<c>r</c>
<c>r</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>From</c>
<c>R</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>If-Match</c>
<c>R</c>
<c>r</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>If-Modified-Since</c>
<c>R</c>
<c>am</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>If-None-Match</c>
<c>R</c>
<c>am</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Last-Modified</c>
<c>R</c>
<c>r</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Last-Modified</c>
<c>r</c>
<c>r</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Location</c>
<c>3rr</c>
<c/>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>Location</c>
<c>R</c>
<c/>
<c>-</c>
<c>-</c>
<c>m</c>
<c>-</c>
<c>Media-Properties</c>
<c>R</c>
<c>amr</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>c</c>
<c>Media-Properties</c>
<c>r</c>
<c>mr</c>
<c>c</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Media-Range</c>
<c>R</c>
<c/>
<c>o</c>
<c>-</c>
<c>-</c>
<c>c</c>
<c>Media-Range</c>
<c>r</c>
<c/>
<c>c</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>MTag</c>
<c>r</c>
<c>r</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Notify-Reason</c>
<c>R</c>
<c/>
<c>-</c>
<c>-</c>
<c>-</c>
<c>m</c>
<c>Pipelined-Requests</c>
<c>R</c>
<c>amdr</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Proxy-Authenticate</c>
<c>407</c>
<c>amdr</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>-</c>
<c>Proxy-Authentication-Info</c>
<c>r</c>
<c>amdr</c>
<c>o/-</c>
<c>o/-</c>
<c>-</c>
<c>-</c>
<c>Proxy-Authorization</c>
<c>R</c>
<c>amdr</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>Proxy-Require</c>
<c>R</c>
<c>ar</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>Proxy-Supported</c>
<c>R</c>
<c>amr</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>-</c>
<c>Proxy-Supported</c>
<c>r</c>
<c/>
<c>c</c>
<c>c</c>
<c>c</c>
<c>-</c>
<c>Public</c>
<c>501</c>
<c>admr</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>-</c>
</texttable>
<texttable anchor="tab_headers2b"
title="Overview of RTSP header fields (R-W) related to methods GET_PARAMETER, SET_PARAMETER, REDIRECT, and PLAY_NOTIFY.">
<preamble/>
<ttcol align="left">Header</ttcol>
<ttcol align="left">Where</ttcol>
<ttcol align="left">Proxy</ttcol>
<ttcol align="left">GPR</ttcol>
<ttcol align="left">SPR</ttcol>
<ttcol align="left">RDR</ttcol>
<ttcol align="left">PNY</ttcol>
<c>Range</c>
<c>R</c>
<c/>
<c>o</c>
<c>-</c>
<c>o</c>
<c>m</c>
<c>Referrer</c>
<c>R</c>
<c/>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>Request-Status</c>
<c>R</c>
<c/>
<c>-</c>
<c>-</c>
<c>-</c>
<c>c</c>
<c>Require</c>
<c>R</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>Retry-After</c>
<c>3rr,503</c>
<c/>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Retry-After</c>
<c>413</c>
<c/>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>RTP-Info</c>
<c>R</c>
<c>r</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>C</c>
<c>RTP-Info</c>
<c>r</c>
<c>r</c>
<c>c</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Scale</c>
<c>G</c>
<c/>
<c>-</c>
<c>-</c>
<c>-</c>
<c>c</c>
<c>Seek-Style</c>
<c>G</c>
<c/>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Server</c>
<c>R</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>Server</c>
<c>r</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>-</c>
<c>Session</c>
<c>R</c>
<c>r</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>m</c>
<c>Session</c>
<c>r</c>
<c>r</c>
<c>c</c>
<c>c</c>
<c>o</c>
<c>m</c>
<c>Speed</c>
<c>G</c>
<c/>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Supported</c>
<c>R</c>
<c>adrm</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>Supported</c>
<c>r</c>
<c>adrm</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>-</c>
<c>Terminate-Reason</c>
<c>R</c>
<c>r</c>
<c>-</c>
<c>-</c>
<c>m</c>
<c>-</c>
<c>Timestamp</c>
<c>R</c>
<c>adrm</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>Timestamp</c>
<c>c</c>
<c>adrm</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>-</c>
<c>Transport</c>
<c>G</c>
<c>mr</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>-</c>
<c>Unsupported</c>
<c>r</c>
<c>arm</c>
<c>c</c>
<c>c</c>
<c>c</c>
<c>-</c>
<c>User-Agent</c>
<c>R</c>
<c>r</c>
<c>m*</c>
<c>m*</c>
<c>-</c>
<c>-</c>
<c>User-Agent</c>
<c>r</c>
<c>r</c>
<c>m*</c>
<c>m*</c>
<c>m*</c>
<c>m*</c>
<!-- <c>Vary</c>
<c>r</c>
<c/>
<c>c</c>
<c>c</c>
<c>-</c>
<c>-</c>
-->
<c>Via</c>
<c>R</c>
<c>amr</c>
<c>o</c>
<c>o</c>
<c>o</c>
<c>-</c>
<c>Via</c>
<c>c</c>
<c>dr</c>
<c>m</c>
<c>m</c>
<c>m</c>
<c>-</c>
<c>WWW-Authenticate</c>
<c>401</c>
<c/>
<c>m</c>
<c>m</c>
<c>m</c>
<c>-</c>
</texttable>
<section anchor="sec_Accept" title="Accept">
<t>The Accept request-header field can be used to specify certain
presentation description and parameter <xref target="RFC6838">media
types</xref> which are acceptable for the response to DESCRIBE and
GET_PARAMETER requests.</t>
<t>See <xref target="sec_syntax-prot-header"/> for the syntax.</t>
<t>The asterisk "*" character is used to group media types into
ranges, with "*/*" indicating all media types and "type/*" indicating
all subtypes of that type. The media-range MAY include media type
parameters that are applicable to that range.</t>
<t>Each media-range MAY be followed by one or more accept-params,
beginning with the "q" parameter for indicating a relative quality
factor. The first "q" parameter (if any) separates the media-range
parameter(s) from the accept-params. Quality factors allow the user or
user agent to indicate the relative degree of preference for that
media-range, using the qvalue scale from 0 to 1 (section 3.9). The
default value is q=1.</t>
<t>Example of use:</t>
<figure>
<artwork><![CDATA[
Accept: application/example ;q=0.7, application/sdp
]]></artwork>
</figure>
<t>Indicates that the requesting agent prefers the media type
application/sdp through the default 1.0 rating but also accepts the
application/example media type with a 0.7 quality rating.</t>
<t>If no Accept header field is present, then it is assumed that the
client accepts all media types. If an Accept header field is present,
and if the server cannot send a response which is acceptable according
to the combined Accept field value, then the server SHOULD send a 406
(not acceptable) response.</t>
</section>
<section anchor="sec_Accept-Credentials" title="Accept-Credentials">
<t>The Accept-Credentials header is a request-header used to indicate
to any trusted intermediary how to handle further secured connections
to proxies or servers. See <xref target="sec_security-framework"/> for
the usage of this header. It MUST NOT be included in server to client
requests.</t>
<t>In a request the header MUST contain the method (User, Proxy, or
Any) for approving credentials selected by the requester. The method
MUST NOT be changed by any proxy, unless it is "Proxy" when a proxy
MAY change it to "user" to take the role of user approving each
further hop. If the method is "User" the header contains zero or more
of credentials that the client accepts. The header may contain zero
credentials in the first RTSP request to a RTSP server when using the
"User" method. This is because the client has not yet received any
credentials to accept. Each credential MUST consist of one URI
identifying the proxy or server, the hash algorithm identifier, and
the hash over that agent's ASN.1 distinguished encoding rules (DER)
encoded certificate <xref target="RFC5280"/> in <xref
target="RFC4648">BASE64, according to Section 4 of</xref> and where
the padding bits are set to zero. All RTSP clients and proxies MUST
implement the SHA-256<xref target="FIPS-pub-180-2"/> algorithm for
computation of the hash of the DER encoded certificate. The SHA-256
algorithm is identified by the token "sha-256".</t>
<t>The intention with allowing for other hash algorithms is to enable
the future retirement of algorithms that are not implemented somewhere
else than here. Thus the definition of future algorithms for this
purpose is intended to be extremely limited. A feature tag can be used
to ensure that support for the replacement algorithm exists.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
Accept-Credentials:User
"rtsps://proxy2.example.com/";sha-256;exaIl9VMbQMOFGClx5rXnPJKVNI=,
"rtsps://server.example.com/";sha-256;lurbjj5khhB0NhIuOXtt4bBRH1M=
]]></artwork>
</figure>
</section>
<section anchor="sec_Accept-Encoding" title="Accept-Encoding">
<t>The Accept-Encoding request-header field is similar to Accept, but
restricts the content-codings (see <xref
target="sec_Content-Encoding"/>),i.e., transformation codings of the
message body, such as gzip compression, that are acceptable in the
response.</t>
<t>A server tests whether a content-coding is acceptable, according to
an Accept-Encoding field, using these rules:</t>
<t><list style="numbers">
<t>If the content-coding is one of the content-codings listed in
the Accept-Encoding field, then it is acceptable, unless it is
accompanied by a qvalue of 0. (As defined in [H3.9], a qvalue of 0
means "not acceptable.")</t>
<t>The special "*" symbol in an Accept-Encoding field matches any
available content-coding not explicitly listed in the header
field.</t>
<t>If multiple content-codings are acceptable, then the acceptable
content-coding with the highest non-zero qvalue is preferred.</t>
<t>The "identity" content-coding is always acceptable, i.e., no
transformation at all, unless specifically refused because the
Accept-Encoding field includes "identity;q=0", or because the
field includes "*;q=0" and does not explicitly include the
"identity" content-coding. If the Accept-Encoding field-value is
empty, then only the "identity" encoding is acceptable.</t>
</list>If an Accept-Encoding field is present in a request, and if
the server cannot send a response which is acceptable according to the
Accept-Encoding header, then the server SHOULD send an error response
with the 406 (Not Acceptable) status code.</t>
<t>If no Accept-Encoding field is present in a request, the server MAY
assume that the client will accept any content coding. In this case,
if "identity" is one of the available content-codings, then the server
SHOULD use the "identity" content-coding, unless it has additional
information that a different content-coding is meaningful to the
client.</t>
</section>
<section anchor="sec_Accept-Language" title="Accept-Language">
<t>The Accept-Language request-header field is similar to Accept, but
restricts the set of natural languages that are preferred as a
response to the request. Note that the language specified applies to
the presentation description and any reason phrases, but not the media
content.</t>
<t>A language tag identifies a natural language spoken, written, or
otherwise conveyed by human beings for communication of information to
other human beings. Computer languages are explicitly excluded. The
syntax and registry of RTSP 2.0 language tags is the same as that
defined by <xref target="RFC5646"/>.</t>
<t>Each language-range MAY be given an associated quality value which
represents an estimate of the user's preference for the languages
specified by that range. The quality value defaults to "q=1". For
example:</t>
<t><list style="hanging">
<t>Accept-Language: da, en-gb;q=0.8, en;q=0.7</t>
</list></t>
<t>would mean: "I prefer Danish, but will accept British English and
other types of English." A language-range matches a language-tag if it
exactly equals the full tag, or if it exactly equals a prefix of the
tag, i.e., the primary-tag in the ABNF, such that the character
following primary-tag is "-". The special range "*", if present in the
Accept-Language field, matches every tag not matched by any other
range present in the Accept-Language field.</t>
<t><list style="hanging">
<t>Note: This use of a prefix matching rule does not imply that
language tags are assigned to languages in such a way that it is
always true that if a user understands a language with a certain
tag, then this user will also understand all languages with tags
for which this tag is a prefix. The prefix rule simply allows the
use of prefix tags if this is the case.</t>
</list></t>
<t>In the process of selecting a language, each language-tag is
assigned a qualification factor, i.e., if a language being supported
by the client is actually supported by the server and what
"preference" level the language achieves. The quality value (q-value)
of the longest language-range in the field that matches the
language-tag is assigned as the qualification factor for a particular
language-tag. If no language-range in the field matches the tag, the
language qualification factor assigned is 0. If no Accept-Language
header is present in the request, the server SHOULD assume that all
languages are equally acceptable. If an Accept-Language header is
present, then all languages which are assigned a qualification factor
greater than 0 are acceptable.</t>
</section>
<section anchor="sec_Accept-Ranges" title="Accept-Ranges">
<t>The Accept-Ranges general-header field allows indication of the
format supported in the Range header. The client MUST include the
header in SETUP requests to indicate which formats are acceptable when
received in PLAY and PAUSE responses, and REDIRECT requests. The
server MUST include the header in SETUP and 456 error responses to
indicate the formats supported for the resource indicated by the
request URI. The header MAY be included in GET_PARAMETER request and
response pairs. The GET_PARAMETER request MUST contain a Session
header to identify the session context the request is related to. The
requester and responder will indicate their capabilities regarding
Range formats respectively.</t>
<figure>
<artwork><![CDATA[
Accept-Ranges: npt, smpte, clock]]></artwork>
</figure>
<t>The syntax is defined in <xref
target="sec_syntax-prot-header"/>.</t>
</section>
<section anchor="sec_Allow" title="Allow">
<t>The Allow message-body header field lists the methods supported by
the resource identified by the Request-URI. The purpose of this field
is to inform the recipient of the complete set of valid methods
associated with the resource. An Allow header field MUST be present in
a 405 (Method Not Allowed) response. The Allow header MUST also be
present in all OPTIONS responses where the content of the header will
not include exactly the same methods as listed in the Public
header.</t>
<t>The Allow message-body header MUST also be included in SETUP and
DESCRIBE responses, if the methods allowed for the resource are
different from the complete set of methods defined in this memo.</t>
<t>Example of use:</t>
<figure>
<artwork><![CDATA[
Allow: SETUP, PLAY, SET_PARAMETER, DESCRIBE
]]></artwork>
</figure>
</section>
<section anchor="sec_Authentication-Info" title="Authentication-Info">
<t>The Authentication-Info response-header is used by the server to
communicate some information regarding the successful authentication
in the response message. This usage of this header is specified in
<xref target="RFC2617"/> with some RTSP clarification in <xref
target="sec_RTSP-HTTP-Authentication"/>. This header MUST only be used
in response messages related to client to server requests.</t>
</section>
<section anchor="sec_Authorization" title="Authorization">
<t>An RTSP client that wishes to authenticate itself with a server
using <xref target="RFC2617"> authentication mechanism from
HTTP</xref> , usually, but not necessarily, after receiving a 401
response, does so by including an Authorization request-header field
with the request. The Authorization field value consists of
credentials containing the authentication information of the user
agent for the realm of the resource being requested. This header MUST
only be used in client to server requests.</t>
<t>If a request is authenticated and a realm specified, the same
credentials SHOULD be valid for all other requests within this realm
(assuming that the authentication scheme itself does not require
otherwise, such as credentials that vary according to a challenge
value or using synchronized clocks).</t>
<t>When a shared cache (see <xref target="sec_caching"/>) receives a
request containing an Authorization field, it MUST NOT return the
corresponding response as a reply to any other request, unless one of
the following specific exceptions holds:</t>
<t><list style="numbers">
<t>If the response includes the "max-age" cache-control directive,
the cache MAY use that response in replying to a subsequent
request. But (if the specified maximum age has passed) a proxy
cache MUST first revalidate it with the origin server, using the
request-headers from the new request to allow the origin server to
authenticate the new request. (This is the defined behavior for
max-age.) If the response includes "max-age=0", the proxy MUST
always revalidate it before re-using it.</t>
<t>If the response includes the "must-revalidate" cache-control
directive, the cache MAY use that response in replying to a
subsequent request. But if the response is stale, all caches MUST
first revalidate it with the origin server, using the
request-headers from the new request to allow the origin server to
authenticate the new request.</t>
<t>If the response includes the "public" cache-control directive,
it MAY be returned in reply to any subsequent request.</t>
</list></t>
</section>
<section anchor="sec_Bandwidth" title="Bandwidth">
<t>The Bandwidth request-header field describes the estimated
bandwidth available to the client, expressed as a positive integer and
measured in kilobits per second. The bandwidth available to the client
may change during an RTSP session, e.g., due to mobility, congestion,
etc.</t>
<t>Clients may not be able to accurately determine the available
bandwidth, for example because the first hop is not a bottleneck. For
example most local area networks (LAN) will not be a bottleneck if the
server is not in the same LAN. Thus link speeds of WLAN or Ethernet
networks are normally not a basis for estimating the available
bandwidth. Cellular devices or other devices directly connected to a
modem or connection enabling device may more accurately estimate the
bottleneck bandwidth and what is a reasonable share of it for RTSP
controlled media. The client will also need to take into account other
traffic sharing the bottleneck. For example by only assigning a
certain fraction to RTSP and its media streams. It is RECOMMENDED that
only clients that have accurate and explicit information about
bandwidth bottlenecks uses this header.</t>
<t>This header is not a substitute for proper congestion control. It
is only a method providing an initial estimate and coarsely determines
if the selected content can be delivered at all.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
Bandwidth: 62360]]></artwork>
</figure>
</section>
<section anchor="sec_Blocksize" title="Blocksize">
<t>The Blocksize request-header field is sent from the client to the
media server asking the server for a particular media packet size.
This packet size does not include lower-layer headers such as IP, UDP,
or RTP. The server is free to use a blocksize which is lower than the
one requested. The server MAY truncate this packet size to the closest
multiple of the minimum, media-specific block size, or override it
with the media-specific size if necessary. The block size MUST be a
positive decimal number, measured in octets. The server only returns
an error (4xx) if the value is syntactically invalid.</t>
</section>
<section anchor="sec_Cache-Control" title="Cache-Control">
<t>The Cache-Control general-header field is used to specify
directives that MUST be obeyed by all caching mechanisms along the
request/response chain.</t>
<t>Cache directives MUST be passed through by a proxy or gateway
application, regardless of their significance to that application,
since the directives may be applicable to all recipients along the
request/response chain. It is not possible to specify a
cache-directive for a specific cache.</t>
<t>Cache-Control should only be specified in a DESCRIBE,
GET_PARAMETER, SET_PARAMETER and SETUP request and its response. Note:
Cache-Control does not govern just the caching of responses as for
HTTP, instead it also applies to the media stream identified by the
SETUP request. The RTSP requests are generally not cacheable, for
further information see <xref target="sec_caching"/>. Below are the
descriptions of the cache directives that can be included in the
Cache-Control header.</t>
<t><list hangIndent="6" style="hanging">
<t hangText="no-cache:">Indicates that the media stream or RTSP
response MUST NOT be cached anywhere. This allows an origin server
to prevent caching even by caches that have been configured to
return stale responses to client requests. Note, there is no
security function preventing the caching of content.</t>
<t hangText="public:">Indicates that the media stream or RTSP
response is cacheable by any cache.</t>
<t hangText="private:">Indicates that the media stream or RTSP
response is intended for a single user and MUST NOT be cached by a
shared cache. A private (non-shared) cache may cache the media
streams.</t>
<t hangText="no-transform:">An intermediate cache (proxy) may find
it useful to convert the media type of a certain stream. A proxy
might, for example, convert between video formats to save cache
space or to reduce the amount of traffic on a slow link. Serious
operational problems may occur, however, when these
transformations have been applied to streams intended for certain
kinds of applications. For example, applications for medical
imaging, scientific data analysis and those using end-to-end
authentication all depend on receiving a stream that is
bit-for-bit identical to the original media stream or RTSP
response. Therefore, if a response includes the no-transform
directive, an intermediate cache or proxy MUST NOT change the
encoding of the stream or response. Unlike HTTP, RTSP does not
provide for partial transformation at this point, e.g., allowing
translation into a different language.</t>
<t hangText="only-if-cached:">In some cases, such as times of
extremely poor network connectivity, a client may want a cache to
return only those media streams or RTSP responses that it
currently has stored, and not to receive these from the origin
server. To do this, the client may include the only-if-cached
directive in a request. If it receives this directive, a cache
SHOULD either respond using a cached media stream or response that
is consistent with the other constraints of the request, or
respond with a 504 (Gateway Timeout) status. However, if a group
of caches is being operated as a unified system with good internal
connectivity, such a request MAY be forwarded within that group of
caches.</t>
<t hangText="max-stale:">Indicates that the client is willing to
accept a media stream or RTSP response that has exceeded its
expiration time. If max-stale is assigned a value, then the client
is willing to accept a response that has exceeded its expiration
time by no more than the specified number of seconds. If no value
is assigned to max-stale, then the client is willing to accept a
stale response of any age.</t>
<t hangText="min-fresh:">Indicates that the client is willing to
accept a media stream or RTSP response whose freshness lifetime is
no less than its current age plus the specified time in seconds.
That is, the client wants a response that will still be fresh for
at least the specified number of seconds.</t>
<t hangText="must-revalidate:">When the must-revalidate directive
is present in a SETUP response received by a cache, that cache
MUST NOT use the cache entry after it becomes stale to respond to
a subsequent request without first revalidating it with the origin
server. That is, the cache is required to do an end-to-end
revalidation every time, if, based solely on the origin server's
Expires, the cached response is stale.</t>
<t hangText="proxy-revalidate:">The proxy-revalidate directive has
the same meaning as the must-revalidate directive, except that it
does not apply to non-shared user agent caches. It can be used on
a response to an authenticated request to permit the user's cache
to store and later return the response without needing to
revalidate it (since it has already been authenticated once by
that user), while still requiring proxies that service many users
to revalidate each time (in order to make sure that each user has
been authenticated). Note that such authenticated responses also
need the public cache control directive in order to allow them to
be cached at all.</t>
<t hangText="max-age:">When an intermediate cache is forced, by
means of a max-age=0 directive, to revalidate its own cache entry,
and the client has supplied its own validator in the request, the
supplied validator might differ from the validator currently
stored with the cache entry. In this case, the cache MAY use
either validator in making its own request without affecting
semantic transparency.</t>
<t hangText="">However, the choice of validator might affect
performance. The best approach is for the intermediate cache to
use its own validator when making its request. If the server
replies with 304 (Not Modified), then the cache can return its now
validated copy to the client with a 200 (OK) response. If the
server replies with a new message body and cache validator,
however, the intermediate cache can compare the returned validator
with the one provided in the client's request, using the strong
comparison function. If the client's validator is equal to the
origin server's, then the intermediate cache simply returns 304
(Not Modified). Otherwise, it returns the new message body with a
200 (OK) response.</t>
</list></t>
</section>
<section anchor="sec_Connection" title="Connection">
<t>The Connection general-header field allows the sender to specify
options that are desired for that particular connection. It MUST NOT
be communicated by proxies over further connections.</t>
<t>RTSP 2.0 proxies MUST parse the Connection header field before a
message is forwarded and, for each connection-token in this field,
remove any header field(s) from the message with the same name as the
connection-token. Connection options are signaled by the presence of a
connection-token in the Connection header field, not by any
corresponding additional header field(s), since the additional header
field may not be sent if there are no parameters associated with that
connection option.</t>
<t>Message headers listed in the Connection header MUST NOT include
end-to-end headers, such as Cache-Control.</t>
<t>RTSP 2.0 defines the "close" connection option for the sender to
signal that the connection will be closed after completion of the
response. For example, Connection: close in either the request or the
response-header fields indicates that the connection SHOULD NOT be
considered <xref target="sec_connections-usage">`persistent'</xref>
after the current request/response is complete.</t>
<t>The use of the connection option "close" in RTSP messages SHOULD be
limited to error messages when the server is unable to recover and
therefore sees it necessary to close the connection. The reason is
that the client has the choice of continuing using a connection
indefinitely, as long as it sends valid messages.</t>
</section>
<section anchor="sec_Connection-Credentials"
title="Connection-Credentials">
<t>The Connection-Credentials response-header is used to carry the
chain of credentials for any next hop that needs to be approved by the
requester. It MUST only be used in server to client responses.</t>
<t>The Connection-Credentials header in an RTSP response MUST, if
included, contain the credential information (in form of a list of
certificates providing the chain of certification) of the next hop
that an intermediary needs to securely connect to. The header MUST
include the URI of the next hop (proxy or server) and a BASE64 (<xref
target="RFC4648">according to Section 4 of</xref> and where the
padding bits are set to zero) encoded binary structure containing a
sequence of DER encoded X.509v3 certificates <xref
target="RFC5280"/>.</t>
<t>The binary structure starts with the number of certificates
(NR_CERTS) included as a 16 bit unsigned integer. This is followed by
NR_CERTS number of 16 bit unsigned integers providing the size in
octets of each DER encoded certificate. This is followed by NR_CERTS
number of DER encoded X.509v3 certificates in a sequence (chain). This
format is exemplified in <xref target="fig-connection-credentials"/>.
The proxy or server's certificate must come first in the structure.
Each following certificate must directly certify the one preceding it.
Because certificate validation requires that root keys be distributed
independently, the self-signed certificate which specifies the root
certificate authority may optionally be omitted from the chain, under
the assumption that the remote end must already possess it in order to
validate it in any case.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
Connection-Credentials:"rtsps://proxy2.example.com/";MIIDNTCC...
]]></artwork>
</figure>
<t>Where MIIDNTCC... is a Base64 encoding of the following
structure:</t>
<figure anchor="fig-connection-credentials"
title="Connection-Credentials header's Certificate Format Example">
<artwork><![CDATA[
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Number of certificates | Size of certificate #1 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Size of certificate #2 | Size of certificate #3 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: DER Encoding of Certificate #1 :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: DER Encoding of Certificate #2 :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: DER Encoding of Certificate #3 :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+]]></artwork>
</figure>
</section>
<section anchor="sec_Content-Base" title="Content-Base">
<t>The Content-Base message-body header field may be used to specify
the base URI for resolving relative URIs within the message body.</t>
<figure>
<artwork><![CDATA[
Content-Base: rtsp://media.example.com/movie/twister/
]]></artwork>
</figure>
<t>If no Content-Base field is present, the base URI of an message
body is defined either by its Content-Location (if that
Content-Location URI is an absolute URI) or the URI used to initiate
the request, in that order of precedence. Note, however, that the base
URI of the contents within the message-body may be redefined within
that message-body.</t>
</section>
<section anchor="sec_Content-Encoding" title="Content-Encoding">
<t>The Content-Encoding message-body header field is used as a
modifier to the media-type. When present, its value indicates what
additional content codings have been applied to the message body, and
thus what decoding mechanisms must be applied in order to obtain the
media-type referenced by the Content-Type header field.
Content-Encoding is primarily used to allow a document to be
compressed without losing the identity of its underlying media
type.</t>
<t>The content-coding is a characteristic of the message body
identified by the Request-URI. Typically, the message body is stored
with this encoding and is only decoded before rendering or analogous
usage. However, an RTSP proxy MAY modify the content-coding if the new
coding is known to be acceptable to the recipient, unless the
"no-transform" cache-control directive is present in the message.</t>
<t>If the content-coding of a message body is not "identity", then the
response MUST include a Content-Encoding Message-body header that
lists the non-identity content-coding(s) used.</t>
<t>If the content-coding of a message body in a request message is not
acceptable to the origin server, the server SHOULD respond with a
status code of 415 (Unsupported Media Type).</t>
<t>If multiple encodings have been applied to a message body, the
content codings MUST be listed in the order in which they were
applied, first to last from left to right. Additional information
about the encoding parameters MAY be provided by other header fields
not defined by this specification.</t>
</section>
<section anchor="sec_Content-Language" title="Content-Language">
<t>The Content-Language message-body header field describes the
natural language(s) of the intended audience for the enclosed message
body. Note that this might not be equivalent to all the languages used
within the message body.</t>
<t>Language tags are mentioned in <xref
target="sec_Accept-Language"/>. The primary purpose of
Content-Language is to allow a user to identify and differentiate
entities according to the user's own preferred language. Thus, if the
body content is intended only for a Danish-literate audience, the
appropriate field is</t>
<t><list style="hanging">
<t>Content-Language: da</t>
</list>If no Content-Language is specified, the default is that the
content is intended for all language audiences. This might mean that
the sender does not consider it to be specific to any natural
language, or that the sender does not know for which language it is
intended.</t>
<t>Multiple languages MAY be listed for content that is intended for
multiple audiences. For example, a rendition of the "Treaty of
Waitangi," presented simultaneously in the original Maori and English
versions, would call for</t>
<t><list style="hanging">
<t>Content-Language: mi, en</t>
</list>However, just because multiple languages are present within a
message body does not mean that it is intended for multiple linguistic
audiences. An example would be a beginner's language primer, such as
"A First Lesson in Latin," which is clearly intended to be used by an
English-literate audience. In this case, the Content-Language would
properly only include "en".</t>
<t>Content-Language MAY be applied to any media type -- it is not
limited to textual documents.</t>
</section>
<section anchor="sec_Content-Length" title="Content-Length">
<t>The Content-Length message-body header field contains the length of
the message body of the RTSP message (i.e., after the double CRLF
following the last header). Unlike HTTP, it MUST be included in all
messages that carry a message body beyond the header portion of the
RTSP message. If it is missing, a default value of zero is assumed.
Any Content-Length greater than or equal to zero is a valid value.</t>
</section>
<section anchor="sec_Content-Location" title="Content-Location">
<t>The Content-Location message-body header field MAY be used to
supply the resource location for the message body enclosed in the
message when that body is accessible from a location separate from the
requested resource's URI. A server SHOULD provide a Content-Location
for the variant corresponding to the response message body; especially
in the case where a resource has multiple variants associated with it,
and those entities actually have separate locations by which they
might be individually accessed, the server SHOULD provide a
Content-Location for the particular variant which is returned.</t>
<t>As example, if an RTSP client performs a DESCRIBE request on a
given resource, e.g.,
"rtsp://a.example.com/movie/Plan9FromOuterSpace", then the server may
use additional information, such as the User-Agent header, to
determine the capabilities of the agent. The server will then return a
media description tailored to that class of RTSP agents. To indicate
which specific description the agent receives the resource identifier
("rtsp://a.example.com/movie/Plan9FromOuterSpace/FullHD.sdp") is
provided in Content-Location, while the description is still a valid
response for the generic resource identifier. Thus enabling both
debugging and cache operation as discussed below.</t>
<t>The Content-Location value is not a replacement for the original
requested URI; it is only a statement of the location of the resource
corresponding to this particular variant at the time of the request.
Future requests MAY specify the Content-Location URI as the request
URI if the desire is to identify the source of that particular
variant. This is useful if the RTSP agent desires to verify if the
resource variant is current through a conditional request.</t>
<t>A cache cannot assume that a message body with a Content-Location
different from the URI used to retrieve it can be used to respond to
later requests on that Content-Location URI. However, the Content-
Location can be used to differentiate between multiple variants
retrieved from a single requested resource.</t>
<t>If the Content-Location is a relative URI, the relative URI is
interpreted relative to the Request-URI.</t>
<t>Note, that Content-Location can be used in some cases to derive the
base-URI for relative URI(s) present in session description formats.
This needs to be taken into account when Content-Location is used. The
easiest way to avoid needing to consider that issue is to include the
Content-Base whenever the Content-Location is included.</t>
<t>Note also, when using Media Tags in conjunction with
Content-Location it is important that the different versions have
different MTags, even if provided under different Content-Location
URIs. This as they have still been provided under the same request
URI.</t>
<t>Note also, as in most cases the URI used in the DESCRIBE and the
SETUP requests are different, the URI provided in a DESCRIBE
Content-Location response can't directly be used in a SETUP request.
Instead the extra step of resolving URIs combined with the media
descriptions indication, like with SDP's a=control attribute.</t>
</section>
<section anchor="sec_Content-Type" title="Content-Type">
<t>The Content-Type message-body header indicates the media type of
the message body sent to the recipient. Note that the content types
suitable for RTSP are likely to be restricted in practice to
presentation descriptions and parameter-value types.</t>
</section>
<section anchor="sec_CSeq" title="CSeq">
<t>The CSeq general-header field specifies the sequence number
(integer) for an RTSP request-response pair. This field MUST be
present in all requests and responses. RTSP agents maintain a sequence
number series for each responder to which they have an open message
transport channel. For each new RTSP request an agent originates on a
particular RTSP message transport the CSeq value MUST be incremented
by one. The initial sequence number MAY be any number, however, it is
RECOMMENDED to start at 0. Each sequence number series is unique
between each requester and responder, i.e., the client has one series
for its requests to a server and the server has another when sending
requests to the client. Each requester and responder is identified by
its socket address (IP address and port number), i.e., per direction
of a TCP connection. Any retransmitted request MUST contain the same
sequence number as the original, i.e., the sequence number is not
incremented for retransmissions of the same request. The RTSP agent
receiving requests MUST process the requests arriving on a particular
transport in the order of the sequence numbers. Responses are sent in
the order that they are generated. The RTSP response MUST have the
same sequence number as was present in the corresponding request. A
RTSP Agent receiving a response MAY receive the responses out of order
compared to the order of the requests it sent. Thus, the agent MUST
use the sequence number in the response to pair it with the
corresponding request.</t>
<t><list style="empty">
<t>The main purpose of the sequence number is to map responses to
requests.</t>
<t>The requirement to use a sequence number increment of one for
each new request is to support any future specification of RTSP
message transport over a protocol that does not provide in order
delivery or is unreliable.</t>
<t>The above rules relating to the initial sequence number may
appear unnecessarily loose. The reason is to cater for some common
behavior of existing implementations: When using multiple reliable
connections in sequence it may still be easiest to use a single
sequence number series for a client connecting with a particular
server. Thus, the initial sequence number may be arbitrary
depending on the number of previous requests. For any unreliable
transport a stricter definition or other solution will be required
to enable detection of any loss of the first request.</t>
<t>When using multiple sequential transport connections, there is
no protocol mechanism to ensure in order processing as the
sequence number is scoped on the individual transport connection
and its five tuple. Thus, there are potential issues with opening
a new transport connection to the same host for which there
already exists a transport connection with outstanding requests
and previously despatched requests related to the same RTSP
session.</t>
</list></t>
<t>RTSP Proxies also need to follow the above rules. This implies that
proxies that aggregate requests from multiple clients onto a single
transport towards a server or a next hop proxy need to renumber these
requests to form a unified sequence on that transport, fulfilling the
above rules. A proxy capable of fulfilling some agent's request
without emitting its own request (e.g., a caching proxy that fulfils a
request from its cache), also causes a need to renumber as the number
of received requests with a particular target, may not be the same as
the number of emitted requests towards that target agent. A proxy that
needs to renumber, needs to perform the corresponding renumbering back
to the original sequence number for any received response before
forwarding it back to the originator of the request.</t>
<t><list style="empty">
<t>A client connected to a proxy, and using that transport to send
requests to multiple servers creates a situation where it is quite
likely to receive the responses out of order. This is because the
proxy will establish separate transports from the proxy to the
servers on which to forward the client's requests. When the
responses arrive from the different servers they will be forwarded
to the client in the order they arrive at the proxy and can be
processed, not the order of the client's original sequence
numbers. This is intentional to avoid some session's requests
being blocked by another server's slow processing of requests.</t>
</list></t>
</section>
<section anchor="sec_Date" title="Date">
<t>The Date general-header field represents the date and time at which
the message was originated. The inclusion of the Date header in RTSP
message follows these rules:</t>
<t><list style="symbols">
<t>An RTSP message, sent either by the client or the server,
containing a body MUST include a Date header, if the sending host
has a clock;</t>
<t>Clients and servers are RECOMMENDED to include a Date header in
all other RTSP messages, if the sending host has a clock;</t>
<t>If the server does not have a clock that can provide a
reasonable approximation of the current time, its responses MUST
NOT include a Date header field. In this case, this rule MUST be
followed: Some origin server implementations might not have a
clock available. An origin server without a clock MUST NOT assign
Expires or Last-Modified values to a response, unless these values
were associated with the resource by a system or user with a
reliable clock. It MAY assign an Expires value that is known, at
or before server configuration time, to be in the past (this
allows "pre-expiration" of responses without storing separate
Expires values for each resource).</t>
</list></t>
<t>A received message that does not have a Date header field MUST be
assigned one by the recipient if the message will be cached by that
recipient. An RTSP implementation without a clock MUST NOT cache
responses without revalidating them on every use. An RTSP cache,
especially a shared cache, SHOULD use a mechanism, such as <xref
target="RFC5905">Network Time Protocol (NTP)</xref>, to synchronize
its clock with a reliable external standard.</t>
<t>The RTSP-date, a full date as specified by Section 3.3 of <xref
target="RFC5322"/>, sent in a Date header SHOULD NOT represent a date
and time subsequent to the generation of the message. It SHOULD
represent the best available approximation of the date and time of
message generation, unless the implementation has no means of
generating a reasonably accurate date and time. In theory, the date
ought to represent the moment just before the message body is
generated. In practice, the date can be generated at any time during
the message origination without affecting its semantic value.</t>
<t><list style="empty">
<t>Note: The RTSP 2.0 date format is defined to be the RFC 5322
full date format. This format is more flexible than the RFC 1123
date format used by RTSP 1.0. Thus implementations should use
single spaces as recommended by RFC 5322 as separators and support
receiving the obsolete format.</t>
<t>[[RFC Editor please remove this note: Prior to version 37 of
the draft, rfc2326bis envisaged sticking with the RFC 1123
format.]]</t>
</list></t>
</section>
<section anchor="sec_Expires" title="Expires">
<t>The Expires message-body header field gives a date and time after
which the description or media-stream should be considered stale. The
interpretation depends on the method: <list hangIndent="6"
style="hanging">
<t hangText="DESCRIBE response:">The Expires header indicates a
date and time after which the presentation description (body)
SHOULD be considered stale.</t>
<t hangText="SETUP response:">The Expires header indicate a date
and time after which the media stream SHOULD be considered
stale.</t>
</list></t>
<t>A stale cache entry may not normally be returned by a cache (either
a proxy cache or an user agent cache) unless it is first validated
with the origin server (or with an intermediate cache that has a fresh
copy of the message body). See <xref target="sec_caching"/> for
further discussion of the expiration model.</t>
<t>The presence of an Expires field does not imply that the original
resource will change or cease to exist at, before, or after that
time.</t>
<t>The format is an absolute date and time as defined by RTSP-date. An
example of its use is</t>
<figure>
<artwork><![CDATA[
Expires: Thu, 01 Dec 1994 16:00:00 GMT]]></artwork>
</figure>
<t>RTSP/2.0 clients and caches MUST treat other invalid date formats,
especially including the value "0", as having occurred in the past
(i.e., already expired).</t>
<t>To mark a response as "already expired," an origin server should
use an Expires date that is equal to the Date header value. To mark a
response as "never expires," an origin server SHOULD use an Expires
date approximately one year from the time the response is sent.
RTSP/2.0 servers SHOULD NOT send Expires dates more than one year in
the future.</t>
</section>
<section anchor="sec_From" title="From">
<t>The From request-header field, if given, SHOULD contain an Internet
e-mail address for the human user who controls the requesting user
agent. The address SHOULD be machine-usable, as defined by "mailbox"
in <xref target="RFC1123"/>.</t>
<t>This header field MAY be used for logging purposes and as a means
for identifying the source of invalid or unwanted requests. It SHOULD
NOT be used as an insecure form of access protection. The
interpretation of this field is that the request is being performed on
behalf of the person given, who accepts responsibility for the method
performed. In particular, robot agents SHOULD include this header so
that the person responsible for running the robot can be contacted if
problems occur on the receiving end.</t>
<t>The Internet e-mail address in this field MAY be separate from the
Internet host which issued the request. For example, when a request is
passed through a proxy the original issuer's address SHOULD be
used.</t>
<t>The client SHOULD NOT send the From header field without the user's
approval, as it might conflict with the user's privacy interests or
their site's security policy. It is strongly recommended that the user
be able to disable, enable, and modify the value of this field at any
time prior to a request.</t>
</section>
<section anchor="sec_If-Match" title="If-Match">
<t>The If-Match request-header field is especially useful for ensuring
the integrity of the presentation description, independent of how the
presentation description was received. The presentation description
can be fetched via means external to RTSP (such as HTTP) or via the
DESCRIBE message. In the case of retrieving the presentation
description via RTSP, the server implementation is guaranteeing the
integrity of the description between the time of the DESCRIBE message
and the SETUP message. By including the MTag given in or with the
session description in an If-Match header part of the SETUP request,
the client ensures that resources set up are matching the description.
A SETUP request with the If-Match header for which the MTag validation
check fails, MUST generate a response using 412 (Precondition
Failed).</t>
<t>This validation check is also very useful if a session has been
redirected from one server to another.</t>
</section>
<section anchor="sec_If-Modified-Since" title="If-Modified-Since">
<t>The If-Modified-Since request-header field is used with the
DESCRIBE and SETUP methods to make them conditional. If the requested
variant has not been modified since the time specified in this field,
a description will not be returned from the server (DESCRIBE) or a
stream will not be set up (SETUP). Instead, a 304 (Not Modified)
response MUST be returned without any message-body.</t>
<t>An example of the field is:</t>
<figure>
<artwork><![CDATA[
If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT]]></artwork>
</figure>
</section>
<section anchor="sec_If-None-Match" title="If-None-Match">
<t>This request-header can be used with one or several message body
tags to make DESCRIBE requests conditional. A client that has one or
more message bodies previously obtained from the resource, can verify
that none of those entities is current by including a list of their
associated message body tags in the If-None-Match header field. The
purpose of this feature is to allow efficient updates of cached
information with a minimum amount of transaction overhead. As a
special case, the value "*" matches any current entity of the
resource.</t>
<t>If any of the message body tags match the message body tag of the
message body that would have been returned in the response to a
similar DESCRIBE request (without the If-None-Match header) on that
resource, or if "*" is given and any current entity exists for that
resource, then the server MUST NOT perform the requested method,
unless required to do so because the resource's modification date
fails to match that supplied in an If-Modified-Since header field in
the request. Instead, if the request method was DESCRIBE, the server
SHOULD respond with a 304 (Not Modified) response, including the
cache-related header fields (particularly MTag) of one of the message
bodies that matched. For all other request methods, the server MUST
respond with a status of 412 (Precondition Failed).</t>
<t>See <xref target="sec.weak_strong_validators"/> for rules on how to
determine if two message body tags match.</t>
<t>If none of the message body tags match, then the server MAY perform
the requested method as if the If-None-Match header field did not
exist, but MUST also ignore any If-Modified-Since header field(s) in
the request. That is, if no message body tags match, then the server
MUST NOT return a 304 (Not Modified) response.</t>
<t>If the request would, without the If-None-Match header field,
result in anything other than a 2xx or 304 status, then the
If-None-Match header MUST be ignored. (See <xref
target="sec.rule_entity_lastmod"/> for a discussion of server behavior
when both If-Modified-Since and If-None-Match appear in the same
request.)</t>
<t>The result of a request having both an If-None-Match header field
and an If-Match header field is unspecified and MUST be considered an
illegal request.</t>
</section>
<section anchor="sec_Last-Modified" title="Last-Modified">
<t>The Last-Modified message-body header field indicates the date and
time at which the origin server believes the presentation description
or media stream was last modified. For the method DESCRIBE, the header
field indicates the last modification date and time of the
description, for SETUP that of the media stream.</t>
<t>An origin server MUST NOT send a Last-Modified date which is later
than the server's time of message origination. In such cases, where
the resource's last modification would indicate some time in the
future, the server MUST replace that date with the message origination
date.</t>
<t>An origin server SHOULD obtain the Last-Modified value of the
message body as close as possible to the time that it generates the
Date value of its response. This allows a recipient to make an
accurate assessment of the message body's modification time,
especially if the message body changes near the time that the response
is generated.</t>
<t>RTSP servers SHOULD send Last-Modified whenever feasible.</t>
</section>
<section anchor="sec_Location" title="Location">
<t>The Location response-header field is used to redirect the
recipient to a location other than the Request-URI for completion of
the request or identification of a new resource. For 3rr responses,
the location SHOULD indicate the server's preferred URI for automatic
redirection to the resource. The field value consists of a single
absolute URI.</t>
<t>Note: The <xref target="sec_Content-Location">Content-Location
header field</xref> differs from Location in that the Content-Location
identifies the original location of the message body enclosed in the
request. It is therefore possible for a response to contain header
fields for both Location and Content-Location. Also, see <xref
target="sec.chache_invalidation"/> for cache requirements of some
methods.</t>
</section>
<section anchor="sec_Media-Properties" title="Media-Properties">
<t>This general-header is used in SETUP response or PLAY_NOTIFY
requests to indicate the media's properties that currently are
applicable to the RTSP session. PLAY_NOTIFY MAY be used to modify
these properties at any point. However, the client SHOULD have
received the update prior to any action related to the new media
properties taking effect. For aggregated sessions, the
Media-Properties header will be returned in each SETUP response. The
header received in the latest response is the one that applies on the
whole session from this point until any future update. The header MAY
be included without value in GET_PARAMETER requests to the server with
a Session header included to query the current Media-Properties for
the session. The responder MUST include the current session's media
properties.</t>
<t>The media properties expressed by this header is the one applicable
to all media in the RTSP session. For aggregated sessions, the header
expressed the combined media-properties. As a result, aggregation of
media MAY result in a change of the media properties, and thus the
content of the Media-Properties header contained in subsequent SETUP
responses.</t>
<t>The header contains a list of property values that are applicable
to the currently setup media or aggregate of media as indicated by the
RTSP URI in the request. No ordering is enforced within the header.
Property values should be grouped into a single group that handles a
particular orthogonal property. Values or groups that express multiple
properties SHOULD NOT be used. The list of properties that can be
expressed MAY be extended at any time. Unknown property values MUST be
ignored.</t>
<t>This specification defines the following 4 groups and their
property values:</t>
<t><list style="hanging">
<t hangText="Random Access:"><list style="hanging">
<t hangText="Random-Access:">Indicates that random access is
possible. May optionally include a floating point value in
seconds indicating the longest duration between any two random
access points in the media.</t>
<t hangText="Beginning-Only:">Seeking is limited to the
beginning only.</t>
<t hangText="No-Seeking:">No seeking is possible.</t>
</list></t>
<t hangText="Content Modifications:"><list style="hanging">
<t hangText="Immutable:">The content will not be changed
during the life-time of the RTSP session.</t>
<t hangText="Dynamic:">The content may be changed based on
external methods or triggers</t>
<t hangText="Time-Progressing:">The media accessible
progresses as wallclock time progresses.</t>
</list></t>
<t hangText="Retention:"><list style="hanging">
<t hangText="Unlimited:">Content will be retained for the
duration of the life-time of the RTSP session.</t>
<t hangText="Time-Limited:">Content will be retained at least
until the specified wallclock time. The time must be provided
in the absolute time format specified in <xref
target="sec_clock"/>.</t>
<t hangText="Time-Duration:">Each individual media unit is
retained for at least the specified time duration. This
definition allows for retaining data with a time based sliding
window. The time duration is expressed as floating point
number in seconds. 0.0 is a valid value as this indicates that
no data is retained in a time-progressing session.</t>
</list></t>
<t hangText="Supported Scale:"><list style="hanging">
<t hangText="Scales:">A quoted comma separated list of one or
more decimal values or ranges of scale values supported by the
content in arbitrary order. A range has a start and stop value
separated by a colon. A range indicates that the content
supports fine grained selection of scale values. Fine grained
allows for steps at least as small as one tenth of a scale
value. A content is considered to support fine grained
selection when the server in response to a given scale value
can produce content with an actual scale that is less than 1
tenth of scale unit, i.e., 0.1, from the requested value.
Negative values are supported. The value 0 has no meaning and
MUST NOT be used.</t>
</list></t>
</list></t>
<t>Examples of this header for on-demand content and a live stream
without recording are:</t>
<figure>
<artwork><![CDATA[
On-demand:
Media-Properties: Random-Access=2.5, Unlimited, Immutable,
Scales="-20, -10, -4, 0.5:1.5, 4, 8, 10, 15, 20"
Live stream without recording/timeshifting:
Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0.0
]]></artwork>
</figure>
<t/>
<t/>
</section>
<section anchor="sec_Media-Range" title="Media-Range">
<t>The Media-Range general-header is used to give the range of the
media at the time of sending the RTSP message. This header MUST be
included in SETUP response, and PLAY and PAUSE response for media that
are Time-Progressing, and PLAY and PAUSE response after any change for
media that are Dynamic, and in PLAY_NOTIFY request that are sent due
to Media-Property-Update. Media-Range header without any range
specifications MAY be included in GET_PARAMETER requests to the server
to request the current range. The server MUST in this case include the
current range at the time of sending the response.</t>
<t>The header MUST include range specifications for all time formats
supported for the media, as indicated in <xref
target="sec_Accept-Ranges">Accept-Ranges header</xref> when setting up
the media. The server MAY include more than one range specification of
any given time format to indicate media that has non-continuous range.
The range specifications SHALL be ordered with the range with the
lowest value or earliest start time first, followed by ranges with
increasingly higher values or later start time.</t>
<t>For media that has the Time-Progressing property, the Media-Range
values will only be valid for the particular point in time when it was
issued. As wallclock progresses so will also the media range. However,
it shall be assumed that media time progresses in direct relationship
to wallclock time (with the exception of clock skew) so that a
reasonably accurate estimation of the media range can be
calculated.</t>
</section>
<section anchor="sec_MTag" title="MTag">
<t>The MTag response-header MAY be included in DESCRIBE, GET_PARAMETER
or SETUP responses. The message body tags (<xref
target="sec_message-tags"/>) returned in a DESCRIBE response, and the
one in SETUP refers to the presentation, i.e., both the returned
session description and the media stream. This allows for verification
that one has the right session description to a media resource at the
time of the SETUP request. However, it has the disadvantage that a
change in any of the parts results in invalidation of all the
parts.</t>
<t>If the MTag is provided both inside the message body, e.g., within
the "a=mtag" attribute in SDP, and in the response message, then both
tags MUST be identical. It is RECOMMENDED that the MTag is primarily
given in the RTSP response message, to ensure that caches can use the
MTag without requiring content inspection. However, for session
descriptions that are distributed outside of RTSP, for example using
HTTP, etc. it will be necessary to include the message body tag in the
session description as specified in <xref target="sec_sdp-mtag"/>.</t>
<t>SETUP and DESCRIBE requests can be made conditional upon the MTag
using the headers If-Match (<xref target="sec_If-Match"/>) and
If-None-Match ( <xref target="sec_If-None-Match"/>).</t>
</section>
<section anchor="sec_Notify-Reason" title="Notify-Reason">
<t>The Notify-Reason response-header is solely used in the PLAY_NOTIFY
method. It indicates the reason why the server has sent the
asynchronous PLAY_NOTIFY request (see <xref
target="sec_PLAY_NOTIFY"/>).</t>
</section>
<section anchor="sec_Pipelined-Requests" title="Pipelined-Requests">
<t>The Pipelined-Requests general-header is used to indicate that a
request is to be executed in the context created by a previous
request(s). The primary usage of this header is to allow pipelining of
SETUP requests so that any additional SETUP request after the first
one does not need to wait for the session ID to be sent back to the
requesting agent. The header contains a unique identifier that is
scoped by the persistent connection used to send the requests.</t>
<t>Upon receiving a request with the Pipelined-Requests the responding
agent MUST look up if there exists a binding between this
Pipelined-Requests identifier for the current persistent connection
and an RTSP session ID. If that exists then the received request is
processed the same way as if it contained the Session header with the
found session ID. If there does not exist a mapping and no Session
header is included in the request, the responding agent MUST create a
binding upon the successful completion of a session creating request,
i.e., SETUP. A binding MUST NOT be created, if the request failed to
create an RTSP session. In case the request contains both a Session
header and the Pipelined-Requests header the Pipelined-Requests MUST
be ignored.</t>
<t>Note: Based on the above definition at least the first request
containing a new unique Pipelined-Requests will be required to be a
SETUP request (unless the protocol is extended with new methods of
creating a session). After that first one, additional SETUP requests
or requests of any type using the RTSP session context may include the
Pipelined-Requests header.</t>
<t>When responding to any request that contained the
Pipelined-Requests header the server MUST also include the Session
header when a binding to a session context exists. An RTSP agent that
knows the session identifier SHOULD NOT use the Pipelined-Requests
header in any request and only use the Session header. This as the
Session identifier is persistent across transport contexts, like TCP
connections, which the Pipelined-Requests identifier is not.</t>
<t>The RTSP agent sending the request with a Pipelined-Requests header
has the responsibility for using a unique and previously unused
identifier within the transport context. Currently only a TCP
connection is defined as such transport context. A server MUST delete
the Pipelined-Requests identifier and its binding to a session upon
the termination of that session. Despite the previous mandate, RTSP
agents are RECOMMENDED to not reuse identifiers to allow for better
error handling and logging.</t>
<t>RTSP Proxies may need to translate Pipelined-Requests identifier
values from incoming requests to outgoing to allow for aggregation of
requests onto a persistent connection.</t>
</section>
<section anchor="sec_Proxy-Authenticate" title="Proxy-Authenticate">
<t>The Proxy-Authenticate response-header field MUST be included as
part of a 407 (Proxy Authentication Required) response. The field
value consists of a challenge that indicates the authentication scheme
and parameters applicable to the proxy for this Request-URI.</t>
<t>The HTTP access authentication process is described in <xref
target="RFC2617"/>. Unlike WWW-Authenticate, the Proxy-Authenticate
header field applies only to the current connection and SHOULD NOT be
passed on to downstream agents. This header MUST only be used in
response messages related to client to server requests.</t>
</section>
<section anchor="sec_Proxy-Authentication-Info"
title="Proxy-Authentication-Info">
<t>The Proxy-Authentication-Info response-header is used by the proxy
to communicate some information regarding the successful
authentication to the proxy in the message response. The content and
usage of this header is described in the <xref target="RFC2617">HTTP
access authentication</xref> that is also used by RTSP and clarified
in <xref target="sec_RTSP-HTTP-Authentication"/>. This header MUST
only be used in response messages related to client to server
requests. This header has hop by hop scope.</t>
</section>
<section anchor="sec_Proxy-Authorization" title="Proxy-Authorization">
<t>The Proxy-Authorization request-header field allows the client to
identify itself (or its user) to a proxy which requires
authentication. The Proxy-Authorization field value consists of
credentials containing the authentication information of the user
agent for the proxy and/or realm of the resource being requested.</t>
<t>The HTTP access authentication process is described in <xref
target="RFC2617"/>. Unlike Authorization, the Proxy-Authorization
header field applies only to the next hop proxy. This header MUST only
be used in client to server requests.</t>
</section>
<section anchor="sec_Proxy-Require" title="Proxy-Require">
<t>The Proxy-Require request-header field is used to indicate
proxy-sensitive features that MUST be supported by the proxy. Any
Proxy-Require header features that are not supported by the proxy MUST
be negatively acknowledged by the proxy to the client using the
Unsupported header. The proxy MUST use the 551 (Option Not Supported)
status code in the response. Any feature-tag included in the
Proxy-Require does not apply to the end-point (server or client). To
ensure that a feature is supported by both proxies and servers the tag
needs to be included in also a Require header.</t>
<t>See <xref target="sec_Require"/> for more details on the mechanics
of this message and a usage example. See discussion in the <xref
target="sec_proxies_ext">proxies section</xref> about when to consider
that a feature requires proxy support.</t>
<t>Example of use:</t>
<figure>
<artwork><![CDATA[
Proxy-Require: play.basic]]></artwork>
</figure>
</section>
<section anchor="sec_Proxy-Supported" title="Proxy-Supported">
<t>The Proxy-Supported general-header field enumerates all the
extensions supported by the proxy using feature-tags. The header
carries the intersection of extensions supported by the forwarding
proxies. The Proxy-Supported header MAY be included in any request by
a proxy. It MUST be added by any proxy if the Supported header is
present in a request. When present in a request, the receiver MUST in
the response copy the received Proxy-Supported header.</t>
<t>The Proxy-Supported header field contains a list of feature-tags
applicable to proxies, as described in <xref
target="sec_feature_tags"/>. The list is the intersection of all
feature-tags understood by the proxies. To achieve an intersection,
the proxy adding the Proxy-Supported header includes all proxy
feature-tags it understands. Any proxy receiving a request with the
header, MUST check the list and removes any feature-tag(s) it does not
support. A Proxy-Supported header present in the response MUST NOT be
modified by the proxies. These feature tags are the ones the proxy
chain support in general, and is not specific to the request
resource.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
C->P1: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
User-Agent: PhonyClient/1.2
P1->P2: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
Proxy-Supported: proxy-foo, proxy-bar, proxy-blech
Via: 2.0 pro.example.com
P2->S: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
Proxy-Supported: proxy-foo, proxy-blech
Via: 2.0 pro.example.com, 2.0 prox2.example.com
S->C: RTSP/2.0 200 OK
Supported: foo, bar, baz
Proxy-Supported: proxy-foo, proxy-blech
Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
Via: 2.0 pro.example.com, 2.0 prox2.example.com]]></artwork>
</figure>
</section>
<section anchor="sec_Public" title="Public">
<t>The Public response-header field lists the set of methods supported
by the response sender. This header applies to the general
capabilities of the sender and its only purpose is to indicate the
sender's capabilities to the recipient. The methods listed may or may
not be applicable to the Request-URI; the <xref
target="sec_Allow">Allow header field</xref> MAY be used to indicate
methods allowed for a particular URI.</t>
<t>Example of use:</t>
<figure>
<artwork><![CDATA[
Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN]]></artwork>
</figure>
<t>In the event that there are proxies between the sender and the
recipient of a response, each intervening proxy MUST modify the Public
header field to remove any methods that are not supported via that
proxy. The resulting Public header field will contain an intersection
of the sender's methods and the methods allowed through by the
intervening proxies.</t>
<t><list style="hanging">
<t>In general, proxies should allow all methods to transparently
pass through from the sending RTSP agent to the receiving RTSP
agent, but there may be cases where this is not desirable for a
given proxy. Modification of the Public response-header field by
the intervening proxies ensures that the request sender gets an
accurate response indicating the methods that can be used on the
target agent via the proxy chain.</t>
</list></t>
</section>
<section anchor="sec_Range" title="Range">
<t>The Range general-header specifies a time range in PLAY (<xref
target="sec_PLAY"/>), PAUSE (<xref target="sec_PAUSE"/>), SETUP (<xref
target="sec_SETUP"/>), REDIRECT (<xref target="sec_REDIRECT"/>), and
PLAY_NOTIFY (<xref target="sec_PLAY_NOTIFY"/>) requests and responses.
It MAY be included in GET_PARAMETER requests from the client to the
server with only a Range format and no value to request the current
media position, whether the session is in Play or Ready state in the
included format. The server SHALL, if supporting the range format,
respond with the current playing point or pause point as the start of
the range. If an explicit stop point was used in the previous PLAY
request, then that value shall be included as stop point. Note that if
the server is currently under any type of media playback manipulation
affecting the interpretation of Range, like Scale, that is also
required to be included in any GET_PARAMETER response to provide
complete information.</t>
<t>The range can be specified in a number of units. This specification
defines smpte (<xref target="sec_smpte"/>), npt (<xref
target="sec_npt"/>), and clock (<xref target="sec_clock"/>) range
units. While octet ranges (Byte Ranges) [H14.35.1] and other extended
units MAY be used, their behavior is unspecified since they are not
normally meaningful in RTSP. Servers supporting the Range header MUST
understand the NPT range format and SHOULD understand the SMPTE range
format. If the Range header is sent in a time format that is not
understood, the recipient SHOULD return 456 (Header Field Not Valid
for Resource) and include an Accept-Ranges header indicating the
supported time formats for the given resource.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
Range: clock=19960213T143205Z-]]></artwork>
</figure>
<t>The Range header contains a range of one single range format. A
range is a half-open interval with a start and an end point, including
the start point, but excluding the end point. A range may either be
fully specified with explicit values for start point and end point, or
have either start or end point be implicit. An implicit start point
indicates the session's pause point, and if no pause point is set the
start of the content. An implicit end point indicates the end of the
content. The usage of both implicit start and end point is not allowed
in the same range header, however, the exclusion of the range header
has that meaning, i.e., from pause point (or start) until end of
content.</t>
<t><list style="empty">
<t>Regarding the half-open intervals; a range of A-B starts
exactly at time A, but ends just before B. Only the start time of
a media unit such as a video or audio frame is relevant. For
example, assume that video frames are generated every 40 ms. A
range of 10.0-10.1 would include a video frame starting at 10.0 or
later time and would include a video frame starting at 10.08, even
though it lasted beyond the interval. A range of 10.0-10.08, on
the other hand, would exclude the frame at 10.08.</t>
<t>Please note the difference between NPT time scales' "now" and
an implicit start value. Implicit value reference the current
pause-point. While "now" is the currently ongoing time. In a
time-progressing session with recording (retention for some or
full time) the pause point may be 2 min into the session while now
could be 1 hour into the session.</t>
</list></t>
<t>By default, range intervals increase, where the second point is
larger than the first point.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
Range: npt=10-15]]></artwork>
</figure>
<t>However, range intervals can also decrease if the Scale header (see
<xref target="sec_Scale"/>) indicates a negative scale value. For
example, this would be the case when a playback in reverse is
desired.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
Scale: -1
Range: npt=15-10]]></artwork>
</figure>
<t>Decreasing ranges are still half open intervals as described above.
Thus, for range A-B, A is closed and B is open. In the above example,
15 is closed and 10 is open. An exception to this rule is the case
when B=0 in a decreasing range. In this case, the range is closed on
both ends, as otherwise there would be no way to reach 0 on a reverse
playback for formats that have such a notion, like NPT and SMPTE.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
Scale: -1
Range: npt=15-0]]></artwork>
</figure>
<t>In this range both 15 and 0 are closed.</t>
<t>A decreasing range interval without a corresponding negative Scale
header is not valid.</t>
</section>
<section anchor="sec_Referrer" title="Referrer">
<t>The Referrer request-header field allows the client to specify, for
the server's benefit, the address (URI) of the resource from which the
Request-URI was obtained. The URI refers to that of the presentation
description, typically retrieved via HTTP. The Referrer request-header
allows a server to generate lists of back-links to resources for
interest, logging, optimized caching, etc. It also allows obsolete or
mistyped links to be traced for maintenance. The Referrer field MUST
NOT be sent if the Request-URI was obtained from a source that does
not have its own URI, such as input from the user keyboard.</t>
<t>If the field value is a relative URI, it SHOULD be interpreted
relative to the Request-URI. The URI MUST NOT include a fragment
identifier.</t>
<t>Because the source of a link might be private information or might
reveal an otherwise private information source, it is strongly
recommended that the user be able to select whether or not the
Referrer field is sent. For example, a streaming client could have a
toggle switch for openly/anonymously, which would respectively
enable/disable the sending of Referrer and From information.</t>
<t>Clients SHOULD NOT include a Referrer header field in a
(non-secure) RTSP request if the referring page was transferred with a
secure protocol.</t>
</section>
<section anchor="sec_Request-Status" title="Request-Status">
<t>This request-header is used to indicate the end result for requests
that take time to complete, such as <xref
target="sec_PLAY">PLAY</xref>. It is sent in <xref
target="sec_PLAY_NOTIFY">PLAY_NOTIFY</xref> with the end-of-stream
reason to report how the PLAY request concluded, either in success or
in failure. The header carries a reference to the request it reports
on using the CSeq number for the session indicated by the Session
header in the request. It provides both a numerical status code
(according to <xref target="sec_status-code"/>) and a human readable
reason phrase.</t>
<figure>
<artwork><![CDATA[
Example:
Request-Status: cseq=63 status=500 reason="Media data unavailable"]]></artwork>
</figure>
<t/>
<t/>
</section>
<section anchor="sec_Require" title="Require">
<t>The Require request-header field is used by agents to ensure that
the other end-point supports features that are required in respect to
this request. It can also be used to query if the other end-point
supports certain features, however, the use of the Supported
general-header (<xref target="sec_Supported"/>) is much more effective
in this purpose. In case any of the feature-tags listed by the Require
header are not supported by the server or client receiving the
request, it MUST respond to the request using the error code 551
(Option Not Supported) and include the Unsupported header listing
those feature-tags which are NOT supported. This header does not apply
to proxies, for the same functionality in respect to proxies see
Proxy-Require header (<xref target="sec_Proxy-Require"/>) with the
exception of media modifying proxies. Media modifying proxies, due to
their nature of handling media in a way that is very similar to a
server, do need to understand also the server's features to correctly
serve the client.</t>
<t><list style="hanging">
<t>This is to make sure that the client-server interaction will
proceed without delay when all features are understood by both
sides, and only slow down if features are not understood (as in
the example below). For a well-matched client-server pair, the
interaction proceeds quickly, saving a round-trip often required
by negotiation mechanisms. In addition, it also removes state
ambiguity when the client requires features that the server does
not understand.</t>
</list></t>
<t>Example (Not complete):</t>
<figure>
<artwork><![CDATA[
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/2.0
CSeq: 302
Require: funky-feature
Funky-Parameter: funkystuff
S->C: RTSP/2.0 551 Option not supported
CSeq: 302
Unsupported: funky-feature]]></artwork>
</figure>
<t>In this example, "funky-feature" is the feature-tag which indicates
to the client that the fictional Funky-Parameter field is required.
The relationship between "funky-feature" and Funky-Parameter is not
communicated via the RTSP exchange, since that relationship is an
immutable property of "funky-feature" and thus should not be
transmitted with every exchange.</t>
<t>Proxies and other intermediary devices MUST ignore this header. If
a particular extension requires that intermediate devices support it,
the extension should be tagged in the Proxy-Require field instead (see
<xref target="sec_Proxy-Require"/>). See discussion in the <xref
target="sec_proxies_ext">proxies section</xref> about when to consider
that a feature requires proxy support.</t>
</section>
<section anchor="sec_Retry-After" title="Retry-After">
<t>The Retry-After response-header field can be used with a 503
(Service Unavailable) or 553 (Proxy Unavailable) response to indicate
how long the service is expected to be unavailable to the requesting
client. This field MAY also be used with any 3rr (Redirection)
response to indicate the minimum time the user-agent is asked to wait
before issuing the redirected request. The value of this field can be
either an RTSP-date or an integer number of seconds (in decimal) after
the time of the response.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
Retry-After: Fri, 31 Dec 1999 23:59:59 GMT
Retry-After: 120]]></artwork>
</figure>
<t>In the latter example, the delay is 2 minutes.</t>
</section>
<section anchor="sec_RTP-Info" title="RTP-Info">
<t>The RTP-Info general-header field is used to set RTP-specific
parameters in the PLAY and GET_PARAMETER responses or a PLAY_NOTIFY
and GET_PARAMETER requests. For streams using RTP as transport
protocol the RTP-Info header SHOULD be part of a 200 response to
PLAY.</t>
<t><list style="hanging">
<t>The exclusion of the RTP-Info in a PLAY response for RTP
transported media will result in a client needing to synchronize
the media streams using RTCP. This may have negative impact as the
RTCP can be lost, and does not need to be particularly timely in
its arrival. Also functionality that informs the client from which
packet a seek has occurred is affected.</t>
</list></t>
<t>The RTP-Info MAY be included in SETUP responses to provide
synchronization information when changing transport parameters, see
<xref target="sec_SETUP"/>. The RTP-Info header and the Range header
MAY be included in a GET_PARAMETER request from client to server
without any values to request the current playback point and
corresponding RTP synchronization information. When the RTP-Info
header is included in a Request the Range header MUST also be included
(Note, Range header only MAY be used). The server response SHALL
include both the Range header and the RTP-Info header. If the session
is in Play state, then the value of the Range header SHALL be filled
in with the current playback point and with the corresponding RTP-Info
values. If the server is another state, no values are included in the
RTP-Info header. The header is included in PLAY_NOTIFY requests with
the Notify-Reason of end-of-stream to provide RTP information about
the end of the stream.</t>
<t>The header can carry the following parameters: <list hangIndent="6"
style="hanging">
<t hangText="url:">Indicates the stream URI for which the
following RTP parameters correspond, this URI MUST be the same as
used in the SETUP request for this media stream. Any relative URI
MUST use the Request-URI as base URI. This parameter MUST be
present.</t>
<t hangText="ssrc:">The Synchronization source (SSRC) that the RTP
timestamp and sequence number provided applies to. This parameter
MUST be present.</t>
<t hangText="seq:">Indicates the sequence number of the first
packet of the stream that is direct result of the request. This
allows clients to gracefully deal with packets when seeking. The
client uses this value to differentiate packets that originated
before the seek from packets that originated after the seek. Note
that a client may not receive the packet with the expressed
sequence number, and instead packets with a higher sequence
number, due to packet loss or reordering. This parameter is
RECOMMENDED to be present.</t>
<t hangText="rtptime:">MUST indicate the RTP timestamp value
corresponding to the start time value in the Range
response-header, or if not explicitly given the implied start
point. The client uses this value to calculate the mapping of RTP
time to NPT or other media timescale. This parameter SHOULD be
present to ensure inter-media synchronization is achieved. There
exists no requirement that any received RTP packet will have the
same RTP timestamp value as the one in the parameter used to
establish synchronization.</t>
</list></t>
<t><list style="hanging">
<t>A mapping from RTP timestamps to Network Time Protocol (NTP)
format timestamps (wallclock) is available via RTCP. However, this
information is not sufficient to generate a mapping from RTP
timestamps to media clock time (NPT, etc.). Furthermore, in order
to ensure that this information is available at the necessary time
(immediately at startup or after a seek), and that it is delivered
reliably, this mapping is placed in the RTSP control channel.</t>
</list> <list style="hanging">
<t>In order to compensate for drift for long, uninterrupted
presentations, RTSP clients should additionally map NPT to NTP,
using initial RTCP sender reports to do the mapping, and later
reports to check drift against the mapping.</t>
</list></t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
Range:npt=3.25-15
RTP-Info:url="rtsp://example.com/foo/audio" ssrc=0A13C760:seq=45102;
rtptime=12345678,url="rtsp://example.com/foo/video"
ssrc=9A9DE123:seq=30211;rtptime=29567112
Lets assume that Audio uses a 16kHz RTP timestamp clock and Video
a 90kHz RTP timestamp clock. Then the media synchronization is
depicted in the following way.
NPT 3.0---3.1---3.2-X-3.3---3.4---3.5---3.6
Audio PA A
Video V PV
X: NPT time value = 3.25, from Range header.
A: RTP timestamp value for Audio from RTP-Info header (12345678).
V: RTP timestamp value for Video from RTP-Info header (29567112).
PA: RTP audio packet carrying an RTP timestamp of 12344878. Which
corresponds to NPT = (12344878 - A) / 16000 + 3.25 = 3.2
PV: RTP video packet carrying an RTP timestamp of 29573412. Which
corresponds to NPT = (29573412 - V) / 90000 + 3.25 = 3.32]]></artwork>
</figure>
</section>
<section anchor="sec_Scale" title="Scale">
<t>The Scale general-header indicates the requested or used view rate
for the media resource being played back. A scale value of 1 indicates
normal play at the normal forward viewing rate. If not 1, the value
corresponds to the rate with respect to normal viewing rate. For
example, a ratio of 2 indicates twice the normal viewing rate ("fast
forward") and a ratio of 0.5 indicates half the normal viewing rate.
In other words, a ratio of 2 has content time increase at twice the
playback time. For every second of elapsed (wallclock) time, 2 seconds
of content time will be delivered. A negative value indicates reverse
direction. For certain media transports this may require certain
considerations to work consistent, see <xref target="sec_rtp"/> for
description on how RTP handles this.</t>
<t>The transmitted data rate SHOULD NOT be changed by selection of a
different scale value. The resulting bit-rate should be reasonably
close to the nominal bit-rate of the content for Scale = 1. The server
has to actively manipulate the data when needed to meet the bitrate
constraints. Implementation of scale changes depends on the server and
media type. For video, a server may, for example, deliver only key
frames or selected frames. For audio, it may time-scale the audio
while preserving pitch or, less desirably, deliver fragments of audio,
or completely mute the audio.</t>
<t>The server and content may restrict the range of scale values that
it supports. The supported values are indicated by the <xref
target="sec_Media-Properties">Media-Properties header</xref>. The
client SHOULD only indicate request values to be supported. However,
as the values may change as the content progresses a requested value
may no longer be valid when the request arrives. Thus, a non-supported
value in a request does not generate an error, only forces the server
to choose the closest value. The response MUST always contain the
actual scale value chosen by the server.</t>
<t>If the server does not implement the possibility to scale, it will
not return a Scale header. A server supporting Scale operations for
PLAY MUST indicate this with the use of the "play.scale"
feature-tag.</t>
<t>When indicating a negative scale for a reverse playback, the Range
header MUST indicate a decreasing range as described in <xref
target="sec_Range"/>.</t>
<t>Example of playing in reverse at 3.5 times normal rate:</t>
<figure>
<artwork><![CDATA[
Scale: -3.5
Range: npt=15-10]]></artwork>
</figure>
</section>
<section anchor="sec_Seek-Style" title="Seek-Style">
<t>When a client sends a PLAY request with a Range header to perform a
random access to the media, the client does not know if the server
will pick the first media samples or the first random access point
prior to the request range. Depending on use case, the client may have
a strong preference. To express this preference and provide the client
with information on how the server actually acted on that preference
the Seek-Style general-header is defined.</t>
<t>Seek-Style is a general-header that MAY be included in any PLAY
request to indicate the client's preference for any media stream that
has random access properties. The server MUST always include the
header in any PLAY response for media with random access properties to
indicate what policy was applied. A server that receives an unknown
Seek-Style policy MUST ignore it and select the server default policy.
A client receiving an unknown policy MUST ignore it and use the Range
header and any media synchronization information as basis to determine
what the server did.</t>
<t>This specification defines the following seek policies that may be
requested (see also <xref target="sec_random_access_seek"/>):</t>
<t><list style="hanging">
<t hangText="RAP:">Random Access Point (RAP) is the behavior of
requesting the server to locate the closest previous random access
point that exists in the media aggregate and deliver from that. By
requesting a RAP, media quality will be the best possible as all
media will be delivered from a point where full media state can be
established in the media decoder.</t>
<t hangText="CoRAP:">Conditional Random Access Point (CoRAP) is a
variant of the above RAP behavior. This policy is primarily
intended for cases where there is larger distance between the
random access points in the media. CoRAP is conditioned on that
there is a Random Access Point closer to the requested start point
than to the current pause point. This policy assumes that the
media state existing prior to the pause is usable if delivery is
continued. If the client or server knows that this is not the fact
the RAP policy should be used. In other words: in most cases when
the client requests a start point prior to the current pause
point, a valid decoding dependency chain from the media delivered
prior to the pause and to the requested media unit will not exist.
If the server searched to a random access point the server MUST
return the CoRAP policy in the Seek-Style header and adjust the
Range header to reflect the position of the picked RAP. In case
the random access point is further away and the server selects to
continue from the current pause point it MUST include the "Next"
policy in the Seek-Style header and adjust the Range header start
point to the current pause point.</t>
<t hangText="First-Prior:">The first-prior policy will start
delivery with the media unit that has a playout time first prior
to the requested time. For discrete media that would only include
media units that would still be rendered at the request time. For
continuous media that is media that will be rendered during the
requested start time of the range.</t>
<t hangText="Next:">The next media units after the provided start
time of the range. For continuous framed media that would mean the
first next frame after the provided time. For discrete media the
first unit that is to be rendered after the provided time. The
main usage for this case is when the client knows it has all media
up to a certain point and would like to continue delivery so that
a complete non-interrupted media playback can be achieved. Example
of such scenarios include switching from a broadcast/multicast
delivery to a unicast based delivery. This policy MUST only be
used on the client's explicit request.</t>
</list>Please note that these expressed preferences exist for
optimizing the startup time or the media quality. The "Next" policy
breaks the normal definition of the Range header to enable a client to
request media with minimal overlap, although some may still occur for
aggregated sessions. RAP and First-Prior both fulfill the requirement
of providing media from the requested range and forward. However,
unless RAP is used, the media quality for many media codecs using
predictive methods can be severely degraded unless additional data is
available as, for example, already buffered, or through other side
channels.</t>
</section>
<section anchor="sec_Server" title="Server">
<t>The Server general-header field contains information about the
software used by the origin server to create or handle the request.
The field can contain multiple product tokens and comments identifying
the server and any significant subproducts. The product tokens are
listed in order of their significance for identifying the
application.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
Server: PhonyServer/1.0]]></artwork>
</figure>
<t>If the response is being forwarded through a proxy, the proxy
application MUST NOT modify the Server response-header. Instead, it
SHOULD include a <xref target="sec_Via">Via field</xref>. If the
response is generated by the proxy, the proxy application MUST return
the Server response-header as previously returned by the server.</t>
</section>
<section anchor="sec_Session" title="Session">
<t>The Session general-header field identifies an RTSP session. An
RTSP session is created by the server as a result of a successful
SETUP request and in the response the session identifier is given to
the client. The RTSP session exists until destroyed by a TEARDOWN,
REDIRECT or timed out by the server.</t>
<t>The session identifier is chosen by the server (see <xref
target="sec_session-id"/>) and MUST be returned in the SETUP response.
Once a client receives a session identifier, it MUST be included in
any request related to that session. This means that the Session
header MUST be included in a request, using the following methods:
PLAY, PAUSE, and TEARDOWN, and MAY be included in SETUP, OPTIONS,
SET_PARAMETER, GET_PARAMETER, and REDIRECT, and MUST NOT be included
in DESCRIBE. The Session header MUST NOT be included in the following
methods, if these requests are pipelined and if the session identifier
is not yet known: PLAY, PAUSE, TEARDOWN, SETUP, OPTIONS SET_PARAMETER,
and GET_PARAMETER.</t>
<t>In an RTSP response the session header MUST be included in methods,
SETUP, PLAY, and PAUSE, and MAY be included in methods, TEARDOWN, and
REDIRECT, and if included in the request of the following methods it
MUST also be included in the response, OPTIONS, GET_PARAMETER, and
SET_PARAMETER, and MUST NOT be included in DESCRIBE responses.</t>
<t>Note that a session identifier identifies an RTSP session across
transport sessions or connections. RTSP requests for a given session
can use different URIs (Presentation and media URIs). Note, that there
are restrictions depending on the session which URIs that are
acceptable for a given method. However, multiple "user" sessions for
the same URI from the same client will require use of different
session identifiers.</t>
<t><list style="hanging">
<t>The session identifier is needed to distinguish several
delivery requests for the same URI coming from the same
client.</t>
</list></t>
<t>The response 454 (Session Not Found) MUST be returned if the
session identifier is invalid.</t>
<t>The header MAY include a parameter for session timeout period. If
not explicitly provided this value is set to 60 seconds. As this
affects how often session keep-alives are needed values smaller than
30 seconds are not recommended. However, larger than default values
can be useful in applications of RTSP that have inactive but
established sessions for longer time periods.<list style="hanging">
<t>60 seconds was chosen as session timeout value due to:
Resulting in not too frequent keep-alive messages and having low
sensitivity to variations in request response timing. If one
reduces the timeout value to below 30 seconds the corresponding
request response timeout becomes a significant part of the session
timeout. 60 seconds also allows for reasonably rapid recovery of
committed server resources in case of client failure.</t>
</list></t>
</section>
<section anchor="sec_Speed" title="Speed">
<t>The Speed general-header field requests the server to deliver
specific amounts of nominal media time per unit of delivery time,
contingent on the server's ability and desire to serve the media
stream at the given speed. The client requests the delivery speed to
be within a given range with a lower and upper bound. The server SHALL
deliver at the highest possible speed within the range, but not faster
than the upper-bound, for which the underlying network path can
support the resulting transport data rates. As long as any speed value
within the given range can be provided the server SHALL NOT modify the
media quality. Only if the server is unable to deliver media at the
speed value provided by the lower bound shall it reduce the media
quality.</t>
<t>Implementation of the Speed functionality by the server is
OPTIONAL. The server can indicate its support through a feature-tag,
play.speed. The lack of a Speed header in the response is an
indication of lack of support of this functionality.</t>
<t>The speed parameter values are expressed as a positive decimal
value, e.g., a value of 2.0 indicates that data is to be delivered
twice as fast as normal. A speed value of zero is invalid. The range
is specified in the form "lower bound - upper bound". The lower bound
value may be smaller or equal to the upper bound. All speeds may not
be possible to support. Therefore the server MAY modify the requested
values to the closest supported. The actual supported speed MUST be
included in the response. Note, however, that the use cases may vary
and that Speed value ranges such as 0.7 - 0.8, 0.3-2.0, 1.0-2.5,
2.5-2.5 all have their usage.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
Speed: 1.0-2.5
]]></artwork>
</figure>
<t>Use of this header changes the bandwidth used for data delivery. It
is meant for use in specific circumstances where delivery of the
presentation at a higher or lower rate is desired. The main use cases
are buffer operations or local scale operations. Implementors should
keep in mind that bandwidth for the session may be negotiated
beforehand (by means other than RTSP), and therefore re-negotiation
may be necessary. To perform Speed operations the server needs to
ensure that the network path can support the resulting bit-rate. Thus
the media transport needs to support feedback so that the server can
react and adapt to the available bitrate.</t>
</section>
<section anchor="sec_Supported" title="Supported">
<t>The Supported general-header enumerates all the extensions
supported by the client or server using feature tags. The header
carries the extensions supported by the message sending client or
server. The Supported header MAY be included in any request. When
present in a request, the receiver MUST respond with its corresponding
Supported header. Note that the Supported header is also included in
4xx and 5xx responses.</t>
<t>The Supported header contains a list of feature-tags, described in
<xref target="sec_feature_tags"/>, that are understood by the client
or server. These feature tags are the ones the server or client
support in general, and is not specific to the request resource.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
C->S: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
Supported: bar, blech, baz]]></artwork>
</figure>
</section>
<section anchor="sec_Terminate-Reason" title="Terminate-Reason">
<t>The Terminate-Reason request-header allows the server when sending
a REDIRECT or TEARDOWN request to provide a reason for the session
termination and any additional information. This specification
identifies three reasons for Redirections and may be extended in the
future:</t>
<t><list style="hanging">
<t hangText="Server-Admin:">The server needs to be shutdown for
some administrative reason.</t>
<t hangText="Session-Timeout:">A client's session has been kept
alive for extended periods of time and the server has determined
that it needs to reclaim the resources associated with this
session.</t>
<t hangText="Internal-Error">An internal error that is impossible
to recover from has occurred forcing the server to terminate the
session.</t>
</list>The Server may provide additional parameters containing
information around the redirect. This specification defines the
following ones.</t>
<t><list style="hanging">
<t hangText="time:">Provides a wallclock time when the server will
stop providing any service.</t>
<t hangText="user-msg:">An UTF-8 text string with a message from
the server to the user. This message SHOULD be displayed to the
user.</t>
</list></t>
<t/>
</section>
<section anchor="sec_Timestamp" title="Timestamp">
<t>The Timestamp general-header describes when the agent sent the
request. The value of the timestamp is of significance only to the
agent and may use any timescale. The responding agent MUST echo the
exact same value and MAY, if it has accurate information about this,
add a floating point number indicating the number of seconds that has
elapsed since it has received the request. The timestamp can be used
by the agent to compute the round-trip time to the responding agent so
that it can adjust the timeout value for retransmissions when running
over an unreliable protocol. It also resolves retransmission
ambiguities for unreliable transport of RTSP.</t>
<t>Note that the present specification provides only for reliable
transport of RTSP messages. The Timestamp general-header is specified
in case the protocol is extended in the future to use unreliable
transport.</t>
</section>
<section anchor="sec_Transport" title="Transport">
<t>The Transport general-header indicates which transport protocol is
to be used and configures its parameters such as destination address,
compression, multicast time-to-live and destination port for a single
stream. It sets those values not already determined by a presentation
description.</t>
<t>A Transport request-header MAY contain a list of transport options
acceptable to the client, in the form of multiple transport
specification entries. Transport specifications are comma separated,
listed in decreasing order of preference. Each transport specification
consists of a transport protocol identifier, followed by any number of
parameters, each parameter separated by a semicolon. A Transport
request-header MAY contain multiple transport specifications using the
same transport protocol Identifier. The server MUST return a Transport
response-header in the response to indicate the values actually chosen
if any. If no transport specification is supported, no transport
header is returned and the response MUST use the status code <xref
target="sec_error461">461 (Unsupported Transport)</xref>. In case more
than one transport specification was present in the request, the
server MUST return the single transport specification (transport-spec)
which was actually chosen, if any. The number of transport-spec
entries is expected to be limited as the client will receive guidance
on what configurations that are possible from the presentation
description.</t>
<t>The Transport header MAY also be used in subsequent SETUP requests
to change transport parameters. A server MAY refuse to change
parameters of an existing stream.</t>
<t>The transport protocol identifier defines for each transport
specification which transport protocol to use and any related rules.
Each transport protocol identifier defines the parameters that are
required to occur; additional optional parameters MAY occur. This
flexibility is provided as parameters may be different and provide
different options to the RTSP Agent. A transport specification may
only contain one of any given parameter within it. A parameter
consists of a name and optionally a value string. Parameters MAY be
given in any order. Additionally, a transport specification may only
contain either of the unicast or the multicast transport type
parameter. The transport protocol identifier and all parameters need
to be understood in a transport specification; if not, the transport
specification MUST be ignored. An RTSP proxy of any type that uses or
modifies the transport specification, e.g., access proxy or security
proxy, MUST remove specifications with unknown parameters before
forwarding the RTSP message. If that results in no remaining transport
specification the proxy SHALL send a <xref target="sec_error461">461
(Unsupported Transport)</xref> response without any Transport
header.</t>
<t><list style="hanging">
<t>The Transport header is restricted to describing a single media
stream. (RTSP can also control multiple streams as a single
entity.) Making it part of RTSP rather than relying on a multitude
of session description formats greatly simplifies designs of
firewalls.</t>
</list></t>
<t>The general syntax for the transport protocol identifier is a list
of slash separated tokens:</t>
<figure>
<artwork><![CDATA[
Value1/Value2/Value3...]]></artwork>
</figure>
<t>Which for RTP transports take the form:</t>
<figure>
<artwork><![CDATA[
RTP/profile/lower-transport.]]></artwork>
</figure>
<t>The default value for the "lower-transport" parameters is specific
to the profile. For RTP/AVP, the default is UDP.</t>
<t>There are two different methods for how to specify where the media
should be delivered for unicast transport: <list hangIndent="6"
style="hanging">
<t hangText="dest_addr:">The presence of this parameter and its
values indicates the destination address or addresses (host
address and port pairs for IP flows) necessary for the media
transport.</t>
<t hangText="No dest_addr:">The lack of the dest_addr parameter
indicates that the server MUST send media to the same address from
which the RTSP messages originates.</t>
</list></t>
<t>The choice of method for indicating where the media is to be
delivered depends on the use case. In some cases the only allowed
method will be to use no explicit address indication and have the
server deliver media to the source of the RTSP messages.</t>
<t>For Multicast there is several methods for specifying addresses but
they are different in how they work compared with unicast:</t>
<t><list hangIndent="6" style="hanging">
<t hangText="dest_addr with client picked address:">The address
and relevant parameters, like TTL (scope), for the actual
multicast group to deliver the media to. There are <xref
target="sec_security">security implications</xref> with this
method that need to be addressed if using this method because a
RTSP server can be used as a Denial of Service (DoS) attacker on
an existing multicast group.</t>
<t hangText="dest_addr using Session Description Information:">The
information included in the transport header can all be coming
from the session description, e.g., the SDP c= and m= line. This
mitigates some of the security issues of the previous methods as
it is the session provider that picks the multicast group and
scope. The client MUST include the information if it is available
in the session description.</t>
<t hangText="No dest_addr:">The behavior when no explicit
multicast group is present in a request is not defined.</t>
</list>An RTSP proxy will need to take care. If the media is not
desired to be routed through the proxy, the proxy will need to
introduce the destination indication.</t>
<t>Below are the configuration parameters associated with transport:
<vspace blankLines="1"/> General parameters: <list hangIndent="6"
style="hanging">
<t hangText="unicast / multicast:">This parameter is a mutually
exclusive indication of whether unicast or multicast delivery will
be attempted. One of the two values MUST be specified. Clients
that are capable of handling both unicast and multicast
transmission need to indicate such capability by including two
full transport-specs with separate parameters for each.</t>
<t hangText="layers:">The number of multicast layers to be used
for this media stream. The layers are sent to consecutive
addresses starting at the dest_addr address. If the parameter is
not included, it defaults to a single layer.</t>
<t hangText="dest_addr:">A general destination address parameter
that can contain one or more address specifications. Each
combination of protocol/profile/lower transport needs to have the
format and interpretation of its address specification defined.
For RTP/AVP/UDP and RTP/AVP/TCP, the address specification is a
tuple containing a host address and port. Note, only a single
destination parameter per transport spec is intended. The usage of
multiple destinations to distribute a single media to multiple
entities is unspecified. <vspace blankLines="1"/> The client
originating the RTSP request MAY specify the destination address
of the stream recipient with the host address part of the tuple.
When the destination address is specified, the recipient may be a
different party than the originator of the request. To avoid
becoming the unwitting perpetrator of a remote-controlled
denial-of-service attack, a server MUST perform security checks
(see <xref target="sec-dos"/>) and SHOULD log such attempts before
allowing the client to direct a media stream to a recipient
address not chosen by the server. Implementations cannot rely on
TCP as reliable means of client identification. If the server does
not allow the host address part of the tuple to be set, it MUST
return 463 (Destination Prohibited). <vspace blankLines="1"/> The
host address part of the tuple MAY be empty, for example ":58044",
in cases when it is desired to specify only the destination port.
Responses to requests including the Transport header with a
dest_addr parameter SHOULD include the full destination address
that is actually used by the server. The server MUST NOT remove
address information present already in the request when responding
unless the protocol requires it.</t>
<t hangText="src_addr:">A general source address parameter that
can contain one or more address specifications. Each combination
of protocol/profile/lower transport needs to have the format and
interpretation of its address specification defined. For
RTP/AVP/UDP and RTP/AVP/TCP, the address specification is a tuple
containing a host address and port. <vspace blankLines="1"/> This
parameter MUST be specified by the server if it transmits media
packets from another address than the one RTSP messages are sent
to. This will allow the client to verify source address and give
it a destination address for its RTCP feedback packets, if RTP is
used. The address or addresses indicated in the src_addr parameter
SHOULD be used both for sending and receiving of the media
stream's data packets. The main reasons are threefold: First,
indicating the port and source address(s) lets the receiver know
where from the packets is expected to originate. Secondly,
traversal of NATs is greatly simplified when traffic is flowing
symmetrically over a NAT binding. Thirdly, certain NAT traversal
mechanisms, needs to know to which address and port to send so
called "binding packets" from the receiver to the sender, thus
creating an address binding in the NAT that the sender to receiver
packet flow can use. <vspace blankLines="1"/> <list
style="hanging">
<t>This information may also be available through SDP.
However, since this is more a feature of transport than media
initialization, the authoritative source for this information
should be in the SETUP response.</t>
</list></t>
<t hangText="mode:">The mode parameter indicates the methods to be
supported for this session. Currently defined valid values are
"PLAY". If not provided, the default is "PLAY". The "RECORD" value
was defined in RFC 2326 and is in this specification unspecified
but reserved. RECORD and other values may be specified in the
future.</t>
<t hangText="interleaved:">The interleaved parameter implies
mixing the media stream with the control stream in whatever
protocol is being used by the control stream, using the mechanism
defined in <xref target="sec_binary"/>. The argument provides the
channel number to be used in the $ block (see <xref
target="sec_binary"/>) and MUST be present. This parameter MAY be
specified as an interval, e.g., interleaved=4-5 in cases where the
transport choice for the media stream requires it, e.g., for RTP
with RTCP. The channel number given in the request is only a
guidance from the client to the server on what channel number(s)
to use. The server MAY set any valid channel number in the
response. The declared channel(s) are bi-directional, so both
end-parties MAY send data on the given channel. One example of
such usage is the second channel used for RTCP, where both server
and client send RTCP packets on the same channel. <vspace
blankLines="1"/> <list style="hanging">
<t>This allows RTP/RTCP to be handled similarly to the way
that it is done with UDP, i.e., one channel for RTP and the
other for RTCP.</t>
</list></t>
<t hangText="MIKEY:">This parameter is used in conjunction with
transport specifications that can utilize <xref
target="RFC3830">MIKEY</xref> for security context establishment.
So far only the SRTP based RTP profiles SAVP and SAVPF can utilize
MIKEY and this is defined in <xref target="sec-mikey"/>. This
parameter can be included both in request and response messages.
The binary MIKEY message SHALL be <xref
target="RFC4648">BASE64</xref> encoded before being included in
the value part of the parameter, where the encoding adheres to the
definition in Section 4 of RFC 4648 and where the padding bits are
set to zero.</t>
</list></t>
<t>Multicast-specific: <list hangIndent="6" style="hanging">
<t hangText="ttl:">multicast time-to-live for IPv4. When included
in requests the value indicate the TTL value that the client
requests the server to use. In a response, the value actually
being used by the server is returned. A server will need to
consider what values that are reasonable and also the authority of
the user to set this value. Corresponding functions are not needed
for IPv6 as the scoping is part of the <xref target="RFC4291">IPv6
multicast address</xref>.</t>
</list></t>
<t>RTP-specific: <vspace blankLines="1"/> These parameters MAY only be
used if the media transport protocol is RTP. <list hangIndent="6"
style="hanging">
<t hangText="ssrc:">The ssrc parameter, if included in a SETUP
response, indicates the RTP SSRC <xref target="RFC3550"/> value(s)
that will be used by the media server for RTP packets within the
stream. It is expressed as an eight digit hexadecimal value.
<vspace blankLines="1"/> The ssrc parameter MUST NOT be specified
in requests. The functionality of specifying the ssrc parameter in
a SETUP request is deprecated as it is incompatible with the
specification of RTP in RFC 3550<xref target="RFC3550"/>. If the
parameter is included in the Transport header of a SETUP request,
the server SHOULD ignore it, and choose appropriate SSRCs for the
stream. The server SHOULD set the ssrc parameter in the Transport
header of the response.</t>
<t hangText="RTCP-mux:">Use to negotiate the usage of <xref
target="RFC5761">RTP and RTCP multiplexing</xref> on a single
underlying transport stream / flow. The presence of this parameter
in a SETUP request indicates the client's support and requires the
server to use RTP and RTCP multiplexing. The client SHALL only
include one transport stream in the Transport header
specification. To provide the server with a choice between using
RTP/RTCP multiplexing or not, two different transport header
specifications must be included.</t>
</list></t>
<t>The parameters setup and connection defined below MAY only be used
if the media transport protocol of the lower-level transport is
connection-oriented (such as TCP). However, these parameters MUST NOT
be used when interleaving data over the RTSP connection.<list
hangIndent="6" style="hanging">
<t hangText="setup:">Clients use the setup parameter on the
Transport line in a SETUP request, to indicate the roles it wishes
to play in a TCP connection. This parameter is adapted from <xref
target="RFC4145"/>. We discuss the use of this parameter in
RTP/AVP/TCP non-interleaved transport in <xref
target="sec_media-tcp-contrans"/>; the discussion below is limited
to syntactic issues. Clients may specify the following values for
the setup parameter: <list style="hanging">
<t hangText="active:">The client will initiate an outgoing
connection.</t>
<t hangText="passive:">The client will accept an incoming
connection.</t>
<t hangText="actpass:">The client is willing to accept an
incoming connection or to initiate an outgoing connection.</t>
</list> <vspace blankLines="1"/> If a client does not specify a
setup value, the "active" value is assumed. <vspace
blankLines="1"/> In response to a client SETUP request where the
setup parameter is set to "active", a server's 2xx reply MUST
assign the setup parameter to "passive" on the Transport header
line. <vspace blankLines="1"/> In response to a client SETUP
request where the setup parameter is set to "passive", a server's
2xx reply MUST assign the setup parameter to "active" on the
Transport header line. <vspace blankLines="1"/> In response to a
client SETUP request where the setup parameter is set to
"actpass", a server's 2xx reply MUST assign the setup parameter to
"active" or "passive" on the Transport header line. <vspace
blankLines="1"/> Note that the "holdconn" value for setup is not
defined for RTSP use, and MUST NOT appear on a Transport line.</t>
<t hangText="connection:">Clients use the connection parameter in
a transport specification part of the Transport header in a SETUP
request, to indicate the client's preference for either reusing an
existing connection between client and server (in which case the
client sets the "connection" parameter to "existing"), or
requesting the creation of a new connection between client and
server (in which cast the client sets the "connection" parameter
to "new"). Typically, clients use the "new" value for the first
SETUP request for a URL, and "existing" for subsequent SETUP
requests for a URL. <vspace blankLines="1"/> If a client SETUP
request assigns the "new" value to "connection", the server
response MUST also assign the "new" value to "connection" on the
Transport line. <vspace blankLines="1"/> If a client SETUP request
assigns the "existing" value to "connection", the server response
MUST assign a value of "existing" or "new" to "connection" on the
Transport line, at its discretion. <vspace blankLines="1"/> The
default value of "connection" is "existing", for all SETUP
requests (initial and subsequent).</t>
</list></t>
<t>The combination of transport protocol, profile and lower transport
needs to be defined. A number of combinations are defined in the <xref
target="sec_mediatran"/>.</t>
<t>Below is a usage example, showing a client advertising the
capability to handle multicast or unicast, preferring multicast. Since
this is a unicast-only stream, the server responds with the proper
transport parameters for unicast.</t>
<figure>
<artwork><![CDATA[
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0
CSeq: 302
Transport: RTP/AVP;multicast;mode="PLAY",
RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
"192.0.2.5:3457";mode="PLAY"
Accept-Ranges: npt, smpte, clock
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 302
Date: Thu, 23 Jan 1997 15:35:06 GMT
Session: 47112344
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
"192.0.2.5:3457";src_addr="192.0.2.224:6256"/
"192.0.2.224:6257";mode="PLAY"
Accept-Ranges: npt
Media-Properties: Random-Access=0.6, Dynamic,
Time-Limited=20081128T165900
]]></artwork>
</figure>
</section>
<section anchor="sec_Unsupported" title="Unsupported">
<t>The Unsupported response-header lists the features not supported by
the responding RTSP agent. In the case where the feature was specified
via the Proxy-Require field (<xref target="sec_Proxy-Require"/>), if
there is a proxy on the path between the client and the server, the
proxy MUST send a response message with a status code of 551 (Option
Not Supported). The request MUST NOT be forwarded.</t>
<t>See <xref target="sec_Require"/> for a usage example.</t>
</section>
<section anchor="sec_User-Agent" title="User-Agent">
<t>The User-Agent general-header field contains information about the
user agent originating the request or producing a response. This is
for statistical purposes, the tracing of protocol violations, and
automated recognition of user agents for the sake of tailoring
responses to avoid particular user agent limitations. User agents
SHOULD include this field with requests. The field can contain
multiple product tokens and comments identifying the agent and any
subproducts which form a significant part of the user agent. By
convention, the product tokens are listed in order of their
significance for identifying the application.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[
User-Agent: PhonyClient/1.2]]></artwork>
</figure>
</section>
<!--
<section anchor="sec_Vary" title="Vary">
<t>The Vary field value indicates the set of request-header fields
that fully determines, while the response is fresh, whether a cache is
permitted to use the response to reply to a subsequent request without
revalidation. For uncacheable or stale responses, the Vary field value
advises the user agent about the criteria that were used to select the
representation. A Vary field value of "*" implies that a cache cannot
determine from the request-headers of a subsequent request whether
this response is the appropriate representation.</t>
<t>An RTSP server SHOULD include a Vary header field with any
cacheable response that is subject to server-driven negotiation. Doing
so allows a cache to properly interpret future requests on that
resource and informs the user agent about the presence of negotiation
on that resource. A server MAY include a Vary header field with a
non-cacheable response that is subject to server-driven negotiation,
since this might provide the user agent with useful information about
the dimensions over which the response varies at the time of the
response.</t>
<t>A Vary field value consisting of a list of field-names signals that
the representation selected for the response is based on a selection
algorithm which considers ONLY the listed request-header field values
in selecting the most appropriate representation. A cache MAY assume
that the same selection will be made for future requests with the same
values for the listed field names, for the duration of time for which
the response is fresh.</t>
<t>The field-names given are not limited to the set of standard
request-header fields defined by this specification. Field names are
case-insensitive.</t>
<t>A Vary field value of "*" signals that unspecified parameters not
limited to the request-headers (e.g., the network address of the
client), play a role in the selection of the response representation.
The "*" value MUST NOT be generated by a proxy server; it may only be
generated by an origin server.</t>
</section>
-->
<section anchor="sec_Via" title="Via">
<t>The Via general-header field MUST be used by proxies to indicate
the intermediate protocols and recipients between the user agent and
the server on requests, and between the origin server and the client
on responses. The field is intended to be used for tracking message
forwards, avoiding request loops, and identifying the protocol
capabilities of all senders along the request/response chain.</t>
<t>Multiple Via field values represents each proxy that has forwarded
the message. Each recipient MUST append its information such that the
end result is ordered according to the sequence of forwarding
applications.</t>
<t>Proxies (e.g., Access Proxy or Translator Proxy) SHOULD NOT, by
default, forward the names and ports of hosts within the
private/protected region. This information SHOULD only be propagated
if explicitly enabled. If not enabled, the via-received of any host
behind the firewall/NAT SHOULD be replaced by an appropriate pseudonym
for that host.</t>
<t>For organizations that have strong privacy requirements for hiding
internal structures, a proxy MAY combine an ordered subsequence of Via
header field entries with identical sent-protocol values into a single
such entry. Applications MUST NOT combine entries which have different
received-protocol values.</t>
</section>
<section anchor="sec_WWW-Authenticate" title="WWW-Authenticate">
<t>The WWW-Authenticate response-header field MUST be included in 401
(Unauthorized) response messages. The field value consists of at least
one challenge that indicates the authentication scheme(s) and
parameters applicable to the Request-URI. This header MUST only be
used in response messages related to client to server requests.</t>
<t>The HTTP access authentication process is described in <xref
target="RFC2617"/> with some clarification in <xref
target="sec_RTSP-HTTP-Authentication"/>. User agents are advised to
take special care in parsing the WWW-Authenticate field value as it
might contain more than one challenge, or if more than one
WWW-Authenticate header field is provided, the contents of a challenge
itself can contain a comma-separated list of authentication
parameters.</t>
</section>
</section>
<section anchor="sec_security-framework" title="Security Framework">
<t>The RTSP security framework consists of two high level components:
the pure authentication mechanisms based on HTTP authentication, and the
message transport protection based on TLS, which is independent of RTSP.
Because of the similarity in syntax and usage between RTSP servers and
HTTP servers, the security for HTTP is re-used to a large extent.</t>
<section anchor="sec_RTSP-HTTP-Authentication"
title="RTSP and HTTP Authentication">
<t>RTSP and HTTP share common authentication schemes, and thus follow
the same usage guidelines as specified in <xref target="RFC2617"/>
with the additions for digest authentication specified below in <xref
target="sec_Digest-Authentication"/>. Servers SHOULD implement both
basic and digest <xref target="RFC2617"/> authentication. Clients MUST
implement both basic and digest authentication <xref
target="RFC2617"/> so that a server that requires the client to
authenticate can trust that the capability is present.</t>
<t>It should be stressed that using the HTTP authentication alone does
not provide full control message security. Therefore, in environments
requiring tighter security for the control messages, TLS SHOULD be
used, see <xref target="sec_sec-frame-TLS"/>. Any RTSP message
containing an Authorization header using basic authorization MUST be
using a TLS connection with confidentiality protection enabled, i.e.,
no NULL encryption.</t>
<t>In cases where there is a chain of proxies between the client and
the server, each proxy may individually request the client or previous
proxy to authenticate itself. This is done using the <xref
target="sec_Proxy-Authenticate">Proxy-Authenticate</xref>, the <xref
target="sec_Proxy-Authorization">Proxy-Authorization</xref> and the
<xref
target="sec_Proxy-Authentication-Info">Proxy-Authentication-Info</xref>
headers. These headers are hop-by-hop headers and are only scoped to
the current connection and hop. Thus if a proxy chain exists, a proxy
connecting to another proxy will have to act as a client to authorize
itself towards the next proxy. The <xref
target="sec_WWW-Authenticate">WWW-Authenticate</xref>, <xref
target="sec_Authorization">Authorization</xref> and <xref
target="sec_Authentication-Info">Authentication-Info</xref> headers
are end-to-end and must not be modified by proxies.</t>
<t>This authentication mechanism works only for client to server
requests as currently defined. This leaves server to client request
outside of the context of TLS based communication more vulnerable to
message injection attacks on the client. Based on the server to client
methods that exist, the potential risks are various; hijacking
(REDIRECT), denial of service (TEARDOWN and PLAY_NOTIFY) or attacks
with uncertain results (SET_PARAMETER).</t>
<section anchor="sec_Digest-Authentication"
title="Digest Authentication">
<t>This section describes the modifications and clarifications
required to apply the HTTP Digest authentication scheme to RTSP. The
RTSP scheme usage is almost completely identical to that for <xref
target="RFC2617">HTTP</xref>. These are based on the procedures
defined for <xref target="RFC3261">SIP 2.0</xref>.</t>
<t>The rules for Digest authentication follow those defined in <xref
target="RFC2617"/>, with "HTTP/1.1" replaced by "RTSP/2.0" in
addition to the following differences:<list style="numbers">
<t>Use the ABNF specified in this document, rather than the one
in <xref target="RFC2617"/>. Consequently the following is
assured: <list style="symbols">
<t>Using the right RTSP URIs allowed in the challenge as
well as in the digest.</t>
<t>Resolved the error in the "uri" parameter of the
Authorization header in <xref target="RFC2617"/>.</t>
</list></t>
<t>If MTags are used then the example procedure for choosing a
nonce based on Etag can work based on replacing ETag with the
MTag.</t>
<t>As a clarification to the calculation of the A2 value for
message integrity assurance in the Digest authentication scheme,
implementers should assume, when the entity-body is empty (that
is, when the RTSP messages have no message body) that the hash
of the message-body resolves to the MD5 hash of an empty string,
or: H(entity-body) = MD5("") =
"d41d8cd98f00b204e9800998ecf8427e".</t>
<t>RFC 2617 notes that a cnonce value MUST NOT be sent in an
Authorization (and by extension Proxy-Authorization) header
field if no qop directive has been sent. Therefore, any
algorithms that have a dependency on the cnonce (including
"MD5-Sess") require that the qop directive be sent. Use of the
"qop" parameter is optional in RFC 2617 for the purposes of
backwards compatibility with RFC 2069; since this specification
defines RTSP 2.0 there is no backwards compatibility issue with
mandating. Thus, all RTSP agents MUST implement qop-options.</t>
</list></t>
<t/>
</section>
</section>
<section anchor="sec_sec-frame-TLS" title="RTSP over TLS">
<t>RTSP agents MUST implement RTSP over TLS as defined in this section
and the next <xref target="sec_sec-frame-proxy"/>. RTSP MUST follow
the same guidelines with regards to TLS <xref target="RFC5246"/> usage
as specified for HTTP, see <xref target="RFC2818"/>. RTSP over TLS is
separated from unsecured RTSP both on the URI level and the port
level. Instead of using the "rtsp" scheme identifier in the URI, the
"rtsps" scheme identifier MUST be used to signal RTSP over TLS. If no
port is given in a URI with the "rtsps" scheme, port 322 MUST be used
for TLS over TCP/IP.</t>
<t>When a client tries to setup an insecure channel to the server
(using the "rtsp" URI), and the policy for the resource requires a
secure channel, the server MUST redirect the client to the secure
service by sending a 301 redirect response code together with the
correct Location URI (using the "rtsps" scheme). A user or client MAY
upgrade a non secured URI to a secured by changing the scheme from
"rtsp" to "rtsps". A server implementing support for "rtsps" MUST
allow this.</t>
<t>It should be noted that TLS allows for mutual authentication (when
using both server and client certificates). Still, one of the more
common ways TLS is used is to only provide server side authentication
(often to avoid client certificates). TLS is then used in addition to
HTTP authentication, providing transport security and server
authentication, while HTTP Authentication is used to authenticate the
client.</t>
<t>RTSP includes the possibility to keep a TCP session up between the
client and server, throughout the RTSP session lifetime. It may be
convenient to keep the TCP session, not only to save the extra setup
time for TCP, but also the extra setup time for TLS (even if TLS uses
the resume function, there will be almost two extra round trips).
Still, when TLS is used, such behavior introduces extra active state
in the server, not only for TCP and RTSP, but also for TLS. This may
increase the vulnerability to DoS attacks.</t>
<t>There exists a potential security vulnerability when reusing TCP
and TLS state for different resources (URIs). If two different host
names point at the same IP address it can be desirable to re-use the
TCP/TLS connection to that server. In that case the RTSP agent having
the TCP/TLS connection MUST verify that the server certificate
associated with the connection has a SubjectAltName matching the host
name present in the URI for the resource an RTSP request is to be
issued for.</t>
<t>In addition to these recommendations, <xref
target="sec_sec-frame-proxy"/> gives further recommendations of TLS
usage with proxies.</t>
</section>
<section anchor="sec_sec-frame-proxy" title="Security and Proxies">
<t>The nature of a proxy is often to act as a "man-in-the-middle",
while security is often about preventing the existence of a
"man-in-the-middle". This section provides clients with the
possibility to use proxies even when applying secure transports (TLS)
between the RTSP agents. The TLS proxy mechanism allows for server and
proxy identification using certificates. However, the client cannot be
identified based on certificates. The client needs to select between
using the procedure specified below or using a TLS connection directly
(by-passing any proxies) to the server. The choice may be dependent on
policies.</t>
<t>There are in general two categories of proxies, the transparent
proxies (of which the client is not aware) and the non-transparent
proxies (of which the client is aware). This memo specifies only
non-transparent RTSP proxies, i.e., proxies visible to the RTSP client
and RTSP server. An infrastructure based on proxies requires that the
trust model is such that both client and servers can trust the proxies
to handle the RTSP messages correctly. To be able to trust a proxy,
the client and server also need to be aware of the proxy. Hence,
transparent proxies cannot generally be seen as trusted and will not
work well with security (unless they work only at the transport
layer). In the rest of this section any reference to proxy will be to
a non-transparent proxy, which inspects or manipulates the RTSP
messages.</t>
<t>HTTP Authentication is built on the assumption of proxies and can
provide user-proxy authentication and proxy-proxy/server
authentication in addition to the client-server authentication.</t>
<t>When TLS is applied and a proxy is used, the client will connect to
the proxy's address when connecting to any RTSP server. This implies
that for TLS, the client will authenticate the proxy server and not
the end server. Note that when the client checks the server
certificate in TLS, it MUST check the proxy's identity (URI or
possibly other known identity) against the proxy's identity as
presented in the proxy's Certificate message.</t>
<t>The problem is that for a proxy accepted by the client, the proxy
needs to be provided information on which grounds it should accept the
next-hop certificate. Both the proxy and the user may have rules for
this, and the user should have the possibility to select the desired
behavior. To handle this case, the Accept-Credentials header (See
<xref target="sec_Accept-Credentials"/>) is used, where the client can
request the proxy/proxies to relay back the chain of certificates used
to authenticate any intermediate proxies as well as the server. The
assumption that the proxies are viewed as trusted, gives the user a
possibility to enforce policies to each trusted proxy of whether it
should accept the next agent in the chain. However, it should be noted
that not all deployments will return the chain of certificates used to
authenticate any intermediate proxies as well as the server. An
operator of such a deployment may want to hide its topology from the
client. It should be noted well that the client does not have any
insight into the proxy's operation. Even if the proxy is trusted, it
can still return an incomplete chain of certificates.</t>
<t>A proxy MUST use TLS for the next hop if the RTSP request includes
a "rtsps" URI. TLS MAY be applied on intermediate links (e.g., between
client and proxy, or between proxy and proxy), even if the resource
and the end server are not required to use it. The proxy MUST, when
initiating the next hop TLS connection, use the incoming TLS
connections cipher suite list, only modified by removing any cipher
suites that the proxy does not support. In case a proxy fails to
establish a TLS connection due to cipher suite mismatch between proxy
and next hop proxy or server, this is indicated using error code 472
(Failure to establish secure connection).</t>
<section anchor="sec-frame-accept-cred" title="Accept-Credentials">
<t>The Accept-Credentials header can be used by the client to
distribute simple authorization policies to intermediate proxies.
The client includes the Accept-Credentials header to dictate how the
proxy treats the server/next proxy certificate. There are currently
three methods defined: <list hangIndent="6" style="hanging">
<t hangText="Any:">which means that the proxy (or proxies) MUST
accept whatever certificate is presented. This is of course not
a recommended option to use, but may be useful in certain
circumstances (such as testing).</t>
<t hangText="Proxy:">which means that the proxy (or proxies)
MUST use its own policies to validate the certificate and decide
whether to accept it or not. This is convenient in cases where
the user has a strong trust relation with the proxy. Reasons why
a strong trust relation may exist are: personal/company proxy,
proxy has a out-of-band policy configuration mechanism.</t>
<t hangText="User:">which means that the proxy (or proxies) MUST
send credential information about the next hop to the client for
authorization. The client can then decide whether the proxy
should accept the certificate or not. See <xref
target="sec_security-tls-proxy"/> for further details.</t>
</list></t>
<t>If the Accept-Credentials header is not included in the RTSP
request from the client, then the "Proxy" method MUST be used as
default. If another method than the "Proxy" is to be used, then the
Accept-Credentials header MUST be included in all of the RTSP
requests from the client. This is because it cannot be assumed that
the proxy always keeps the TLS state or the user's previous
preference between different RTSP messages (in particular if the
time interval between the messages is long).</t>
<t>With the "Any" and "Proxy" methods the proxy will apply the
policy as defined for each method. If the policy does not accept the
credentials of the next hop, the proxy MUST respond with a message
using status code 471 (Connection Credentials not accepted).</t>
<t>An RTSP request in the direction server to client MUST NOT
include the Accept-Credentials header. As for the non-secured
communication, the possibility for these requests depends on the
presence of a client established connection. However, if the server
to client request is in relation to a session established over a TLS
secured channel, it MUST be sent in a TLS secured connection. That
secured connection MUST also be the one used by the last client to
server request. If no such transport connection exists at the time
when the server desires to send the request, the server MUST discard
the message.</t>
<t>Further policies MAY be defined and registered, but should be
done so with caution.</t>
</section>
<section anchor="sec_security-tls-proxy"
title="User approved TLS procedure">
<t>For the "User" method, each proxy MUST perform the following
procedure for each RTSP request: <list hangIndent="3"
style="symbols">
<t>Setup the TLS session to the next hop if not already present
(i.e., run the TLS handshake, but do not send the RTSP
request).</t>
<t>Extract the peer certificate chain for the TLS session.</t>
<t>Check if a matching identity and hash of the peer certificate
is present in the Accept-Credentials header. If present, send
the message to the next hop, and conclude these procedures. If
not, go to the next step.</t>
<t>The proxy responds to the RTSP request with a 470 or 407
response code. The 407 response code MAY be used when the proxy
requires both user and connection authorization from user or
client. In this message the proxy MUST include a
Connection-Credentials header, see <xref
target="sec_Connection-Credentials"/> with the next hop's
identity and certificate.</t>
</list></t>
<t>The client MUST upon receiving a 470 or 407 response with
Connection-Credentials header take the decision on whether to accept
the certificate or not (if it cannot do so, the user SHOULD be
consulted). If the certificate is accepted, the client has to again
send the RTSP request. In that request the client has to include the
Accept-Credentials header including the hash over the DER encoded
certificate for all trusted proxies in the chain.</t>
<t>Example: <figure>
<artwork><![CDATA[
C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
CSeq: 2
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
"192.0.2.5:4589"
Accept-Ranges: npt, smpte, clock
Accept-Credentials: User
P->C: RTSP/2.0 470 Connection Authorization Required
CSeq: 2
Connection-Credentials: "rtsps://test.example.org";
MIIDNTCCAp...
C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
CSeq: 3
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
"192.0.2.5:4589"
Accept-Credentials: User "rtsps://test.example.org";sha-256;
dPYD7txpoGTbAqZZQJ+vaeOkyH4=
Accept-Ranges: npt, smpte, clock
P->S: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
CSeq: 3
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
"192.0.2.5:4589"
Via: RTSP/2.0 proxy.example.org
Accept-Credentials: User "rtsps://test.example.org";sha-256;
dPYD7txpoGTbAqZZQJ+vaeOkyH4=
Accept-Ranges: npt, smpte, clock
]]></artwork>
</figure></t>
<t>One implication of this process is that the connection for
secured RTSP messages may take significantly more round-trip times
for the first message. A complete extra message exchange between the
proxy connecting to the next hop and the client results because of
the process for approval for each hop. However, if each message
contains the chain of proxies that the requester accepts, the
remaining message exchange should not be delayed. The procedure of
including the credentials in each request rather than building state
in each proxy, avoids the need for revocation procedures.</t>
</section>
</section>
</section>
<!-- title="Security Framework" -->
<section anchor="sec_syntax" title="Syntax">
<t>The RTSP syntax is described in an Augmented Backus-Naur Form (ABNF)
as defined in RFC 5234 <xref target="RFC5234"/>. It uses the basic
definitions present in RFC 5234.</t>
<t>Please note that ABNF strings, e.g., "Accept", are case insensitive
as specified in section 2.3 of RFC 5234.</t>
<t>The RTSP syntax makes use of the ISO 10646 character set in UTF-8
encoding RFC 3629 <xref target="RFC3629"/>.</t>
<section title="Base Syntax">
<t>RTSP header values can be folded onto multiple lines if the
continuation line begins with a space or horizontal tab. All linear
white space, including folding, has the same semantics as SP. A
recipient MAY replace any linear white space with a single SP before
interpreting the field value or forwarding the message downstream.
This is intended to behave exactly as HTTP/1.1 as described in RFC
2616 <xref target="RFC2616"/>. The SWS construct is used when linear
white space is optional, generally between tokens and separators.</t>
<t>To separate the header name from the rest of value, a colon is
used, which, by the above rule, allows whitespace before, but no line
break, and whitespace after, including a line break. The HCOLON
defines this construct. <figure>
<artwork><![CDATA[
OCTET = %x00-FF ; any 8-bit sequence of data
CHAR = %x01-7F ; any US-ASCII character (octets 1 - 127)
UPALPHA = %x41-5A ; any US-ASCII uppercase letter "A".."Z"
LOALPHA = %x61-7A ;any US-ASCII lowercase letter "a".."z"
ALPHA = UPALPHA / LOALPHA
DIGIT = %x30-39 ; any US-ASCII digit "0".."9"
CTL = %x00-1F / %x7F ; any US-ASCII control character
; (octets 0 - 31) and DEL (127)
CR = %x0D ; US-ASCII CR, carriage return (13)
LF = %x0A ; US-ASCII LF, linefeed (10)
SP = %x20 ; US-ASCII SP, space (32)
HT = %x09 ; US-ASCII HT, horizontal-tab (9)
BACKSLASH = %x5C ; US-ASCII backslash (92)
CRLF = CR LF]]></artwork>
</figure> <figure>
<artwork><![CDATA[
LWS = [CRLF] 1*( SP / HT ) ; Line-breaking White Space
SWS = [LWS] ; Separating White Space
HCOLON = *( SP / HT ) ":" SWS
TEXT = %x20-7E / %x80-FF ; any OCTET except CTLs
tspecials = "(" / ")" / "<" / ">" / "@"
/ "," / ";" / ":" / BACKSLASH / DQUOTE
/ "/" / "[" / "]" / "?" / "="
/ "{" / "}" / SP / HT
token = 1*(%x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39
/ %x41-5A / %x5E-7A / %x7C / %x7E)
; 1*<any CHAR except CTLs or tspecials>
quoted-string = ( DQUOTE *qdtext DQUOTE )
qdtext = %x20-21 / %x23-5B / %x5D-7E / quoted-pair
/ UTF8-NONASCII
; No DQUOTE and no "\"
quoted-pair = "\\" / ( "\" DQUOTE )
ctext = %x20-27 / %x2A-7E
/ %x80-FF ; any OCTET except CTLs, "(" and ")"
generic-param = token [ EQUAL gen-value ]
gen-value = token / host / quoted-string]]></artwork>
</figure> <figure>
<artwork><![CDATA[
safe = "$" / "-" / "_" / "." / "+"
extra = "!" / "*" / "'" / "(" / ")" / ","
rtsp-extra = "!" / "*" / "'" / "(" / ")"
HEX = DIGIT / "A" / "B" / "C" / "D" / "E" / "F"
/ "a" / "b" / "c" / "d" / "e" / "f"
LHEX = DIGIT / "a" / "b" / "c" / "d" / "e" / "f"
; lowercase "a-f" Hex
reserved = ";" / "/" / "?" / ":" / "@" / "&" / "="
unreserved = ALPHA / DIGIT / safe / extra
rtsp-unreserved = ALPHA / DIGIT / safe / rtsp-extra
base64 = *base64-unit [base64-pad]
base64-unit = 4base64-char
base64-pad = (2base64-char "==") / (3base64-char "=")
base64-char = ALPHA / DIGIT / "+" / "/"]]></artwork>
</figure> <figure>
<artwork><![CDATA[
SLASH = SWS "/" SWS ; slash
EQUAL = SWS "=" SWS ; equal
LPAREN = SWS "(" SWS ; left parenthesis
RPAREN = SWS ")" SWS ; right parenthesis
COMMA = SWS "," SWS ; comma
SEMI = SWS ";" SWS ; semicolon
COLON = SWS ":" SWS ; colon
MINUS = SWS "-" SWS ; minus/dash
LDQUOT = SWS DQUOTE ; open double quotation mark
RDQUOT = DQUOTE SWS ; close double quotation mark
RAQUOT = ">" SWS ; right angle quote
LAQUOT = SWS "<" ; left angle quote
TEXT-UTF8char = %x21-7E / UTF8-NONASCII
UTF8-NONASCII = UTF8-1 / UTF8-2 / UTF8-3 / UTF8-4
UTF8-1 = <As defined in RFC 3629>
UTF8-2 = <As defined in RFC 3629>
UTF8-3 = <As defined in RFC 3629>
UTF8-4 = <As defined in RFC 3629>
UTF8-CONT = %x80-BF
POS-FLOAT = 1*12DIGIT ["." 1*9DIGIT]
FLOAT = ["-"] POS-FLOAT]]></artwork>
</figure></t>
</section>
<section anchor="sec_syntax-prot" title="RTSP Protocol Definition">
<section anchor="sec_syntax-prot-generic"
title="Generic Protocol elements">
<t><figure>
<artwork><![CDATA[
RTSP-IRI = schemes ":" IRI-rest
IRI-rest = ihier-part [ "?" iquery ]
ihier-part = "//" iauthority ipath-abempty
RTSP-IRI-ref = RTSP-IRI / irelative-ref
irelative-ref = irelative-part [ "?" iquery ]
irelative-part = "//" iauthority ipath-abempty
/ ipath-absolute
/ ipath-noscheme
/ ipath-empty
iauthority = < As defined in RFC 3987>
ipath = ipath-abempty ; begins with "/" or is empty
/ ipath-absolute ; begins with "/" but not "//"
/ ipath-noscheme ; begins with a non-colon segment
/ ipath-rootless ; begins with a segment
/ ipath-empty ; zero characters
ipath-abempty = *( "/" isegment )
ipath-absolute = "/" [ isegment-nz *( "/" isegment ) ]
ipath-noscheme = isegment-nz-nc *( "/" isegment )
ipath-rootless = isegment-nz *( "/" isegment )
ipath-empty = 0<ipchar>
isegment = *ipchar [";" *ipchar]
isegment-nz = 1*ipchar [";" *ipchar]
/ ";" *ipchar
isegment-nz-nc = (1*ipchar-nc [";" *ipchar-nc])
/ ";" *ipchar-nc
; non-zero-length segment without any colon ":"
; No parameter (; delimited) inside path.
ipchar = iunreserved / pct-encoded / sub-delims / ":" / "@"
ipchar-nc = iunreserved / pct-encoded / sub-delims / "@"
; sub-delims is different from RFC 3987
; not including ";"
iquery = < As defined in RFC 3987>
iunreserved = < As defined in RFC 3987>
pct-encoded = < As defined in RFC 3987>]]></artwork>
</figure></t>
<t><figure>
<artwork><![CDATA[
RTSP-URI = schemes ":" URI-rest
RTSP-REQ-URI = schemes ":" URI-req-rest
RTSP-URI-Ref = RTSP-URI / RTSP-Relative
RTSP-REQ-Ref = RTSP-REQ-URI / RTSP-REQ-Rel
schemes = "rtsp" / "rtsps" / scheme
scheme = < As defined in RFC 3986>
URI-rest = hier-part [ "?" query ]
URI-req-rest = hier-part [ "?" query ]
; Note fragment part not allowed in requests
hier-part = "//" authority path-abempty
RTSP-Relative = relative-part [ "?" query ]
RTSP-REQ-Rel = relative-part [ "?" query ]
relative-part = "//" authority path-abempty
/ path-absolute
/ path-noscheme
/ path-empty
authority = < As defined in RFC 3986>
query = < As defined in RFC 3986>
path = path-abempty ; begins with "/" or is empty
/ path-absolute ; begins with "/" but not "//"
/ path-noscheme ; begins with a non-colon segment
/ path-rootless ; begins with a segment
/ path-empty ; zero characters
path-abempty = *( "/" segment )
path-absolute = "/" [ segment-nz *( "/" segment ) ]
path-noscheme = segment-nz-nc *( "/" segment )
path-rootless = segment-nz *( "/" segment )
path-empty = 0<pchar>
segment = *pchar [";" *pchar]
segment-nz = ( 1*pchar [";" *pchar]) / (";" *pchar)
segment-nz-nc = ( 1*pchar-nc [";" *pchar-nc]) / (";" *pchar-nc)
; non-zero-length segment without any colon ":"
; No parameter (; delimited) inside path.
pchar = unreserved / pct-encoded / sub-delims / ":" / "@"
pchar-nc = unreserved / pct-encoded / sub-delims / "@"
sub-delims = "!" / "$" / "&" / "'" / "(" / ")"
/ "*" / "+" / "," / "="
; sub-delims is different from RFC 3986/3987
; not including ";"
]]></artwork>
</figure> <figure>
<artwork><![CDATA[
smpte-range = smpte-type [EQUAL smpte-range-spec]
; See section 4.4
smpte-range-spec = ( smpte-time "-" [ smpte-time ] )
/ ( "-" smpte-time )
smpte-type = "smpte" / "smpte-30-drop"
/ "smpte-25" / smpte-type-extension
; other timecodes may be added
smpte-type-extension = "smpte" token
smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT
[ ":" 1*2DIGIT [ "." 1*2DIGIT ] ]
]]></artwork>
</figure> <figure>
<artwork><![CDATA[
npt-range = "npt" [EQUAL npt-range-spec]
npt-range-spec = ( npt-time "-" [ npt-time ] ) / ( "-" npt-time )
npt-time = "now" / npt-sec / npt-hhmmss
npt-sec = 1*19DIGIT [ "." 1*9DIGIT ]
npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." 1*9DIGIT ]
npt-hh = 1*19DIGIT ; any positive number
npt-mm = 1*2DIGIT ; 0-59
npt-ss = 1*2DIGIT ; 0-59
]]></artwork>
</figure> <figure>
<artwork><![CDATA[
utc-range = "clock" [EQUAL utc-range-spec]
utc-range-spec = ( utc-time "-" [ utc-time ] ) / ( "-" utc-time )
utc-time = utc-date "T" utc-clock "Z"
utc-date = 8DIGIT
utc-clock = 6DIGIT [ "." 1*9DIGIT ]
]]></artwork>
</figure> <figure>
<artwork><![CDATA[
feature-tag = token
session-id = 1*256( ALPHA / DIGIT / safe )
extension-header = header-name HCOLON header-value
header-name = token
header-value = *(TEXT-UTF8char / UTF8-CONT / LWS)
]]></artwork>
</figure></t>
</section>
<section anchor="sec_syntax-prot-message" title="Message Syntax">
<t><figure>
<artwork><![CDATA[
RTSP-message = Request / Response ; RTSP/2.0 messages
Request = Request-Line
*((general-header
/ request-header
/ message-body-header) CRLF)
CRLF
[ message-body-data ]
Response = Status-Line
*((general-header
/ response-header
/ message-body-header) CRLF)
CRLF
[ message-body-data ] ]]></artwork>
</figure> <figure>
<artwork><![CDATA[
Request-Line = Method SP Request-URI SP RTSP-Version CRLF
Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF]]></artwork>
</figure> <figure>
<artwork><![CDATA[Method = "DESCRIBE"
/ "GET_PARAMETER"
/ "OPTIONS"
/ "PAUSE"
/ "PLAY"
/ "PLAY_NOTIFY"
/ "REDIRECT"
/ "SETUP"
/ "SET_PARAMETER"
/ "TEARDOWN"
/ extension-method
extension-method = token]]></artwork>
</figure> <figure>
<artwork><![CDATA[
Request-URI = "*" / RTSP-REQ-URI
RTSP-Version = "RTSP/" 1*DIGIT "." 1*DIGIT
message-body-data = 1*OCTET]]></artwork>
</figure> <figure>
<artwork><![CDATA[
Status-Code = "100" ; Continue
/ "200" ; OK
/ "301" ; Moved Permanently
/ "302" ; Found
/ "303" ; See Other
/ "304" ; Not Modified
/ "305" ; Use Proxy
/ "400" ; Bad Request
/ "401" ; Unauthorized
/ "402" ; Payment Required
/ "403" ; Forbidden
/ "404" ; Not Found
/ "405" ; Method Not Allowed
/ "406" ; Not Acceptable
/ "407" ; Proxy Authentication Required
/ "408" ; Request Time-out
/ "410" ; Gone
/ "412" ; Precondition Failed
/ "413" ; Request Message Body Too Large
/ "414" ; Request-URI Too Large
/ "415" ; Unsupported Media Type
/ "451" ; Parameter Not Understood
/ "452" ; reserved
/ "453" ; Not Enough Bandwidth
/ "454" ; Session Not Found
/ "455" ; Method Not Valid in This State
/ "456" ; Header Field Not Valid for Resource
/ "457" ; Invalid Range
/ "458" ; Parameter Is Read-Only
/ "459" ; Aggregate operation not allowed
/ "460" ; Only aggregate operation allowed
/ "461" ; Unsupported Transport
/ "462" ; Destination Unreachable
/ "463" ; Destination Prohibited
/ "464" ; Data Transport Not Ready Yet
/ "465" ; Notification Reason Unknown
/ "466" ; Key Management Error
/ "470" ; Connection Authorization Required
/ "471" ; Connection Credentials not accepted
/ "472" ; Failure to establish secure connection
/ "500" ; Internal Server Error
/ "501" ; Not Implemented
/ "502" ; Bad Gateway
/ "503" ; Service Unavailable
/ "504" ; Gateway Time-out
/ "505" ; RTSP Version not supported
/ "551" ; Option not supported
/ extension-code
extension-code = 3DIGIT
Reason-Phrase = 1*(TEXT-UTF8char / HT / SP)
]]></artwork>
</figure> <figure>
<artwork><![CDATA[rtsp-header = general-header
/ request-header
/ response-header
/ message-body-header
general-header = Accept-Ranges
/ Cache-Control
/ Connection
/ CSeq
/ Date
/ Media-Properties
/ Media-Range
/ Pipelined-Requests
/ Proxy-Supported
/ Range
/ RTP-Info
/ Scale
/ Seek-Style
/ Server
/ Session
/ Speed
/ Supported
/ Timestamp
/ Transport
/ User-Agent
/ Via
/ extension-header]]></artwork>
</figure> <figure>
<artwork><![CDATA[
request-header = Accept
/ Accept-Credentials
/ Accept-Encoding
/ Accept-Language
/ Authorization
/ Bandwidth
/ Blocksize
/ From
/ If-Match
/ If-Modified-Since
/ If-None-Match
/ Notify-Reason
/ Proxy-Authorization
/ Proxy-Require
/ Referrer
/ Request-Status
/ Require
/ Terminate-Reason
/ extension-header]]></artwork>
</figure> <figure>
<artwork><![CDATA[
response-header = Authentication-Info
/ Connection-Credentials
/ Location
/ MTag
/ Proxy-Authenticate
/ Proxy-Authentication-Info
/ Public
/ Retry-After
/ Unsupported
/ WWW-Authenticate
/ extension-header]]></artwork>
</figure> <figure>
<artwork><![CDATA[
message-body-header = Allow
/ Content-Base
/ Content-Encoding
/ Content-Language
/ Content-Length
/ Content-Location
/ Content-Type
/ Expires
/ Last-Modified
/ extension-header]]></artwork>
</figure></t>
</section>
<section anchor="sec_syntax-prot-header" title="Header Syntax">
<t><figure>
<artwork><![CDATA[
Accept = "Accept" HCOLON
[ accept-range *(COMMA accept-range) ]
accept-range = media-type-range [SEMI accept-params]
media-type-range = ( "*/*"
/ ( m-type SLASH "*" )
/ ( m-type SLASH m-subtype )
) *( SEMI m-parameter )
accept-params = "q" EQUAL qvalue *(SEMI generic-param )
qvalue = ( "0" [ "." *3DIGIT ] )
/ ( "1" [ "." *3("0") ] )
Accept-Credentials = "Accept-Credentials" HCOLON cred-decision
cred-decision = ("User" [LWS cred-info])
/ "Proxy"
/ "Any"
/ (token [LWS 1*header-value])
; For future extensions
cred-info = cred-info-data *(COMMA cred-info-data)
cred-info-data = DQUOTE RTSP-REQ-URI DQUOTE SEMI hash-alg
SEMI base64
hash-alg = "sha-256" / extension-alg
extension-alg = token
Accept-Encoding = "Accept-Encoding" HCOLON
[ encoding *(COMMA encoding) ]
encoding = codings [SEMI accept-params]
codings = content-coding / "*"
content-coding = "identity" / token
Accept-Language = "Accept-Language" HCOLON
language *(COMMA language)
language = language-range [SEMI accept-params]
language-range = language-tag / "*"
language-tag = primary-tag *( "-" subtag )
primary-tag = 1*8ALPHA
subtag = 1*8ALPHA
Accept-Ranges = "Accept-Ranges" HCOLON acceptable-ranges
acceptable-ranges = (range-unit *(COMMA range-unit))
range-unit = "npt" / "smpte" / "smpte-30-drop" / "smpte-25"
/ "clock" / extension-format
extension-format = token
Allow = "Allow" HCOLON Method *(COMMA Method)
Authentication-Info = "Authentication-Info" HCOLON auth-info
auth-info = auth-info-entry *(COMMA auth-info-entry)
auth-info-entry = nextnonce
/ message-qop
/ response-auth
/ cnonce
/ nonce-count
nextnonce = "nextnonce" EQUAL nonce-value
response-auth = "rspauth" EQUAL response-digest
response-digest = DQUOTE *LHEX DQUOTE
Authorization = "Authorization" HCOLON credentials
credentials = basic-credential
/ digest-credential
/ other-response
basic-credential = "Basic" LWS basic-credentials
basic-credentials = base64 ; Base64 encoding of user-password
user-password = basic-username ":" password
basic-username = *CF-TEXT
CF-TEXT = %x20-39 / %x3B-7E / %x80-FF ; TEXT without :
password = *TEXT
digest-credential = ("Digest" LWS digest-response)
digest-response = dig-resp *(COMMA dig-resp)
dig-resp = username / realm / nonce / digest-uri
/ dresponse / algorithm / cnonce
/ opaque / message-qop
/ nonce-count / auth-param
username = "username" EQUAL username-value
username-value = quoted-string
digest-uri = "uri" EQUAL LDQUOT digest-uri-value RDQUOT
digest-uri-value = RTSP-REQ-URI
message-qop = "qop" EQUAL qop-value
cnonce = "cnonce" EQUAL cnonce-value
cnonce-value = nonce-value
nonce-count = "nc" EQUAL nc-value
nc-value = 8LHEX
dresponse = "response" EQUAL request-digest
request-digest = LDQUOT 32LHEX RDQUOT
auth-param = auth-param-name EQUAL
( token / quoted-string )
auth-param-name = token
other-response = auth-scheme LWS auth-param
*(COMMA auth-param)
auth-scheme = token
Bandwidth = "Bandwidth" HCOLON 1*19DIGIT
Blocksize = "Blocksize" HCOLON 1*9DIGIT ]]></artwork>
</figure><figure>
<artwork><![CDATA[
Cache-Control = "Cache-Control" HCOLON cache-directive
*(COMMA cache-directive)
cache-directive = cache-rqst-directive
/ cache-rspns-directive
cache-rqst-directive = "no-cache"
/ "max-stale" [EQUAL delta-seconds]
/ "min-fresh" EQUAL delta-seconds
/ "only-if-cached"
/ cache-extension
cache-rspns-directive = "public"
/ "private"
/ "no-cache"
/ "no-transform"
/ "must-revalidate"
/ "proxy-revalidate"
/ "max-age" EQUAL delta-seconds
/ cache-extension
cache-extension = token [EQUAL (token / quoted-string)]
delta-seconds = 1*19DIGIT]]></artwork>
</figure><figure>
<artwork><![CDATA[
Connection = "Connection" HCOLON connection-token
*(COMMA connection-token)
connection-token = "close" / token
Connection-Credentials = "Connection-Credentials" HCOLON cred-chain
cred-chain = DQUOTE RTSP-REQ-URI DQUOTE SEMI base64
Content-Base = "Content-Base" HCOLON RTSP-URI
Content-Encoding = "Content-Encoding" HCOLON
content-coding *(COMMA content-coding)
Content-Language = "Content-Language" HCOLON
language-tag *(COMMA language-tag)
Content-Length = "Content-Length" HCOLON 1*19DIGIT
Content-Location = "Content-Location" HCOLON RTSP-REQ-Ref
Content-Type = "Content-Type" HCOLON media-type
media-type = m-type SLASH m-subtype *(SEMI m-parameter)
m-type = discrete-type / composite-type
discrete-type = "text" / "image" / "audio" / "video"
/ "application" / extension-token
composite-type = "message" / "multipart" / extension-token
extension-token = ietf-token / x-token
ietf-token = token
x-token = "x-" token
m-subtype = extension-token / iana-token
iana-token = token
m-parameter = m-attribute EQUAL m-value
m-attribute = token
m-value = token / quoted-string
CSeq = "CSeq" HCOLON cseq-nr
cseq-nr = 1*9DIGIT
Date = "Date" HCOLON RTSP-date
RTSP-date = date-time ;
date-time = <As defined in RFC 5322>
Expires = "Expires" HCOLON RTSP-date
From = "From" HCOLON from-spec
from-spec = ( name-addr / addr-spec ) *( SEMI from-param )
name-addr = [ display-name ] LAQUOT addr-spec RAQUOT
addr-spec = RTSP-REQ-URI / absolute-URI
absolute-URI = < As defined in RFC 3986>
display-name = *(token LWS) / quoted-string
from-param = tag-param / generic-param
tag-param = "tag" EQUAL token
If-Match = "If-Match" HCOLON ("*" / message-tag-list)
message-tag-list = message-tag *(COMMA message-tag)
message-tag = [ weak ] opaque-tag
weak = "W/"
opaque-tag = quoted-string
If-Modified-Since = "If-Modified-Since" HCOLON RTSP-date
If-None-Match = "If-None-Match" HCOLON ("*" / message-tag-list)
Last-Modified = "Last-Modified" HCOLON RTSP-date
Location = "Location" HCOLON RTSP-REQ-URI
Media-Properties = "Media-Properties" HCOLON [media-prop-list]
media-prop-list = media-prop-value *(COMMA media-prop-value)
media-prop-value = ("Random-Access" [EQUAL POS-FLOAT])
/ "Beginning-Only"
/ "No-Seeking"
/ "Immutable"
/ "Dynamic"
/ "Time-Progressing"
/ "Unlimited"
/ ("Time-Limited" EQUAL utc-time)
/ ("Time-Duration" EQUAL POS-FLOAT)
/ ("Scales" EQUAL scale-value-list)
/ media-prop-ext
media-prop-ext = token [EQUAL (1*rtsp-unreserved / quoted-string)]
scale-value-list = DQUOTE scale-entry *(COMMA scale-entry) DQUOTE
scale-entry = scale-value / (scale-value COLON scale-value)
scale-value = FLOAT
Media-Range = "Media-Range" HCOLON [ranges-list]
ranges-list = ranges-spec *(COMMA ranges-spec)
MTag = "MTag" HCOLON message-tag
Notify-Reason = "Notify-Reason" HCOLON Notify-Reas-val
Notify-Reas-val = "end-of-stream"
/ "media-properties-update"
/ "scale-change"
/ Notify-Reason-extension
Notify-Reason-extension = token
Pipelined-Requests = "Pipelined-Requests" HCOLON startup-id
startup-id = 1*8DIGIT]]></artwork>
</figure></t>
<t><figure>
<artwork><![CDATA[Proxy-Authenticate = "Proxy-Authenticate" HCOLON challenge-list
challenge-list = challenge *(COMMA challenge)
challenge = ("Digest" LWS digest-cln *(COMMA digest-cln))
/ ("Basic" LWS realm)
/ other-challenge
other-challenge = auth-scheme LWS auth-param
*(COMMA auth-param)
digest-cln = realm / domain / nonce
/ opaque / stale / algorithm
/ qop-options / auth-param
realm = "realm" EQUAL realm-value
realm-value = quoted-string
domain = "domain" EQUAL LDQUOT RTSP-REQ-Ref
*(1*SP RTSP-REQ-Ref ) RDQUOT
nonce = "nonce" EQUAL nonce-value
nonce-value = quoted-string
opaque = "opaque" EQUAL quoted-string
stale = "stale" EQUAL ( "true" / "false" )
algorithm = "algorithm" EQUAL ("MD5" / "MD5-sess" / token)
qop-options = "qop" EQUAL LDQUOT qop-value
*("," qop-value) RDQUOT
qop-value = "auth" / "auth-int" / token
Proxy-Authentication-Info = "Proxy-Authentication-Info" HCOLON auth-info
Proxy-Authorization = "Proxy-Authorization" HCOLON credentials
Proxy-Require = "Proxy-Require" HCOLON feature-tag-list
feature-tag-list = feature-tag *(COMMA feature-tag)
Proxy-Supported = "Proxy-Supported" HCOLON [feature-tag-list]
Public = "Public" HCOLON Method *(COMMA Method)
Range = "Range" HCOLON ranges-spec
ranges-spec = npt-range / utc-range / smpte-range
/ range-ext
range-ext = extension-format [EQUAL range-value]
range-value = 1*(rtsp-unreserved / quoted-string / ":" )
Referrer = "Referrer" HCOLON (absolute-URI / RTSP-URI-Ref)
Request-Status = "Request-Status" HCOLON req-status-info
req-status-info = cseq-info LWS status-info LWS reason-info
cseq-info = "cseq" EQUAL cseq-nr
status-info = "status" EQUAL Status-Code
reason-info = "reason" EQUAL DQUOTE Reason-Phrase DQUOTE
Require = "Require" HCOLON feature-tag-list
]]></artwork>
</figure><figure>
<artwork><![CDATA[RTP-Info = "RTP-Info" HCOLON [rtsp-info-spec
*(COMMA rtsp-info-spec)]
rtsp-info-spec = stream-url 1*ssrc-parameter
stream-url = "url" EQUAL DQUOTE RTSP-REQ-Ref DQUOTE
ssrc-parameter = LWS "ssrc" EQUAL ssrc HCOLON
ri-parameter *(SEMI ri-parameter)
ri-parameter = ("seq" EQUAL 1*5DIGIT)
/ ("rtptime" EQUAL 1*10DIGIT)
/ generic-param
Retry-After = "Retry-After" HCOLON (RTSP-date / delta-seconds)
Scale = "Scale" HCOLON scale-value
Seek-Style = "Seek-Style" HCOLON Seek-S-values
Seek-S-values = "RAP"
/ "CoRAP"
/ "First-Prior"
/ "Next"
/ Seek-S-value-ext
Seek-S-value-ext = token
Server = "Server" HCOLON ( product / comment )
*(LWS (product / comment))
product = token [SLASH product-version]
product-version = token
comment = LPAREN *( ctext / quoted-pair) RPAREN
Session = "Session" HCOLON session-id
[ SEMI "timeout" EQUAL delta-seconds ]
Speed = "Speed" HCOLON lower-bound MINUS upper-bound
lower-bound = POS-FLOAT
upper-bound = POS-FLOAT
Supported = "Supported" HCOLON [feature-tag-list]
]]></artwork>
</figure> <figure>
<artwork><![CDATA[Terminate-Reason = "Terminate-Reason" HCOLON TR-Info
TR-Info = TR-Reason *(SEMI TR-Parameter)
TR-Reason = "Session-Timeout"
/ "Server-Admin"
/ "Internal-Error"
/ token
TR-Parameter = TR-time / TR-user-msg / generic-param
TR-time = "time" EQUAL utc-time
TR-user-msg = "user-msg" EQUAL quoted-string
Timestamp = "Timestamp" HCOLON timestamp-value [LWS delay]
timestamp-value = *19DIGIT [ "." *9DIGIT ]
delay = *9DIGIT [ "." *9DIGIT ]
Transport = "Transport" HCOLON transport-spec
*(COMMA transport-spec)
transport-spec = transport-id *trns-parameter
transport-id = trans-id-rtp / other-trans
trans-id-rtp = "RTP/" profile ["/" lower-transport]
; no LWS is allowed inside transport-id
other-trans = token *("/" token)]]></artwork>
</figure> <figure>
<artwork><![CDATA[
profile = "AVP" / "SAVP" / "AVPF" / "SAVPF" / token
lower-transport = "TCP" / "UDP" / token
trns-parameter = (SEMI ( "unicast" / "multicast" ))
/ (SEMI "interleaved" EQUAL channel ["-" channel])
/ (SEMI "ttl" EQUAL ttl)
/ (SEMI "layers" EQUAL 1*DIGIT)
/ (SEMI "ssrc" EQUAL ssrc *(SLASH ssrc))
/ (SEMI "mode" EQUAL mode-spec)
/ (SEMI "dest_addr" EQUAL addr-list)
/ (SEMI "src_addr" EQUAL addr-list)
/ (SEMI "setup" EQUAL contrans-setup)
/ (SEMI "connection" EQUAL contrans-con)
/ (SEMI "RTCP-mux")
/ (SEMI "MIKEY" EQUAL MIKEY-Value)
/ (SEMI trn-param-ext)
contrans-setup = "active" / "passive" / "actpass"
contrans-con = "new" / "existing"
trn-param-ext = par-name [EQUAL trn-par-value]
par-name = token
trn-par-value = *(rtsp-unreserved / quoted-string)
ttl = 1*3DIGIT ; 0 to 255
ssrc = 8HEX
channel = 1*3DIGIT ; 0 to 255
MIKEY-Value = base64
mode-spec = ( DQUOTE mode *(COMMA mode) DQUOTE )
mode = "PLAY" / token
addr-list = quoted-addr *(SLASH quoted-addr)
quoted-addr = DQUOTE (host-port / extension-addr) DQUOTE
host-port = ( host [":" port] )
/ ( ":" port )
extension-addr = 1*qdtext
host = < As defined in RFC 3986>
port = < As defined in RFC 3986>
]]></artwork>
</figure> <figure>
<artwork><![CDATA[
Unsupported = "Unsupported" HCOLON feature-tag-list
User-Agent = "User-Agent" HCOLON ( product / comment )
*(LWS (product / comment))
Via = "Via" HCOLON via-parm *(COMMA via-parm)
via-parm = sent-protocol LWS sent-by *( SEMI via-params )
via-params = via-ttl / via-maddr
/ via-received / via-extension
via-ttl = "ttl" EQUAL ttl
via-maddr = "maddr" EQUAL host
via-received = "received" EQUAL (IPv4address / IPv6address)
IPv4address = < As defined in RFC 3986>
IPv6address = < As defined in RFC 3986>
via-extension = generic-param
sent-protocol = protocol-name SLASH protocol-version
SLASH transport-prot
protocol-name = "RTSP" / token
protocol-version = token
transport-prot = "UDP" / "TCP" / "TLS" / other-transport
other-transport = token
sent-by = host [ COLON port ]
WWW-Authenticate = "WWW-Authenticate" HCOLON challenge-list
]]></artwork>
</figure></t>
<!-- Vary = "Vary" HCOLON ( "*" / field-name-list) -->
</section>
</section>
<section anchor="sec_sdp-syntax" title="SDP extension Syntax">
<t>This section defines in ABNF the SDP extensions defined for RTSP.
See <xref target="sec_sdpusage"/> for the definition of the extensions
in text.</t>
<figure>
<artwork><![CDATA[
control-attribute = "a=control:" *SP RTSP-REQ-Ref CRLF
a-range-def = "a=range:" ranges-spec CRLF
a-mtag-def = "a=mtag:" message-tag CRLF
]]></artwork>
</figure>
</section>
</section>
<!-- title="Syntax" -->
<section anchor="sec_security" title="Security Considerations">
<t>The security considerations and threats around RTSP and its usage can
be divided into considerations around the signaling protocol itself and
the issues related to the media stream delivery. However, when it comes
to mitigations of security threats, a threat depending on the media
stream delivery may in fact be mitigated by a mechanism in the signaling
protocol.</t>
<t>There are several chapters and an appendix in this document that
define security solutions for the protocol. We will reference them when
discussing the threats below. But the reader should take special notice
of the <xref target="sec_security-framework">Security Framework</xref>
and the specification of how to use <xref target="sec-savp">SRTP and its
key-mangement</xref> to achieve certain aspects of the media
security.</t>
<section title="Signaling Protocol Threats">
<t>This section focuses on issues related to the signaling protocol.
Because of the similarity in syntax and usage between RTSP servers and
HTTP servers, the security considerations outlined in [H15] apply
also.</t>
<t>Specifically, please note the following: <list hangIndent="6"
style="hanging">
<t hangText="Abuse of Server Log Information:">A server is in the
position to save personal data about a user's requests which might
identify their media consumption patterns or subjects of interest.
This information is clearly confidential in nature and its
handling can be constrained by law in certain countries. RTSP
servers will presumably have similar logging mechanisms to HTTP,
and thus should be equally guarded in protecting the contents of
those logs, thus protecting the privacy of the users of the
servers. People using the RTSP protocol to provide media are
responsible for ensuring that logging material is not distributed
without the permission of any individuals that are identifiable by
the published results.</t>
<t hangText="Transfer of Sensitive Information:">There is no
reason to believe that information transferred or controlled via
RTSP may be any less sensitive than that normally transmitted via
HTTP. Therefore, all of the precautions regarding the protection
of data privacy and user privacy apply to implementors of RTSP
clients, servers, and proxies. See [H15.1.2] for further
details.</t>
<t hangText="Attacks Based On File and Path Names:">Though RTSP
URIs are opaque handles that do not necessarily have file system
semantics, it is anticipated that many implementations will
translate portions of the Request-URIs directly to file system
calls. In such cases, file systems SHOULD follow the precautions
outlined in [H15.2], such as checking for ".." in path
components.</t>
<t hangText="Personal Information:">RTSP clients are often privy
to the same information that HTTP clients are (user name,
location, etc.) and thus should be equally sensitive. See [H15.1]
for further recommendations.</t>
<t hangText="Privacy Issues Connected to Accept Headers:">Since
may of the same "Accept" headers exist in RTSP as in HTTP, the
same caveats outlined in [H15.1.4] with regards to their use
should be followed.</t>
<t hangText="DNS Spoofing:">Presumably, given the longer
connection times typically associated to RTSP sessions relative to
HTTP sessions, RTSP client DNS optimizations should be less
prevalent. Nonetheless, the recommendations provided in [H15.3]
are still relevant to any implementation which attempts to rely on
a DNS-to-IP mapping to hold beyond a single use of the
mapping.</t>
<t hangText="Location Headers and Spoofing:">If a single server
supports multiple organizations that do not trust each another,
then it MUST check the values of Location and Content-Location
header fields in responses that are generated under control of
said organizations to make sure that they do not attempt to
invalidate resources over which they have no authority.
([H15.4])</t>
</list></t>
<t>In addition to the recommendations in the current HTTP
specification (RFC 2616 <xref target="RFC2616"/>, as of this writing)
and also of the previous RFC 2068 <xref target="RFC2068"/>, future
HTTP specifications may provide additional guidance on security
issues.</t>
<t>The following are added considerations for RTSP implementations.
<list hangIndent="6" style="hanging">
<t hangText="Session hijacking:">Since there is no or little
relation between a transport layer connection and an RTSP session,
it is possible for a malicious client to issue requests with
random session identifiers which could affect other clients of an
unsuspecting server. To mitigate this the server SHALL use a
large, random and non-sequential session identifier to minimize
the possibility of this kind of attack. However, unless the RTSP
signaling is always confidentiality protected, e.g., using TLS, an
on-path attacker will be able to hijack a session. Another choice
for preventing session hijacking is to use client authentication
and only allow the authenticated client creating the session to
access that session.</t>
<t hangText="Authentication:">Servers SHOULD implement both basic
and digest <xref target="RFC2617"/> authentication. In
environments requiring tighter security for the control messages,
the transport layer mechanism <xref target="RFC5246">TLS</xref>
SHOULD be used.</t>
<t hangText="Suspicious behavior:">RTSP servers upon detecting
instances of behavior which is deemed a security risk SHOULD
return error code 403 (Forbidden). RTSP servers SHOULD also be
aware of attempts to probe the server for weaknesses and entry
points and MAY arbitrarily disconnect and ignore further requests
from clients which are deemed to be in violation of local security
policy.</t>
<t hangText="TLS through proxies:">If one uses the possibility to
connect TLS in multiple legs (<xref
target="sec_sec-frame-proxy"/>) one really needs to be aware of
the trust model. That procedure requires full faith and trust in
all proxies, which will be identified, that one allows to connect
through. They are men in the middle and have access to all that
goes on over the TLS connection. Thus it is important to consider
if that trust model is acceptable in the actual application.
Further discussion of the actual trust model is in <xref
target="sec_sec-frame-proxy"/>.</t>
<t hangText="Resource Exhaustion:">As RTSP is a stateful protocol
and establishes resource usage on the server there is a clear
possibility to attack the server by trying to overbook these
resources to perform a denial of service attack. This attack can
be both against ongoing sessions and to prevent others from
establishing sessions. RTSP agents will need to have mechanisms to
prevent single peers from consuming extensive amounts of
resources. The methods for guarding against this are varied and
depends on the agent's role and capabilities and policies. Each
implementation has to carefully consider their methods and
policies to mitigate this threat. For example regarding handling
of connections there are recommendations in <xref
target="sec-overload-control"/>.</t>
</list>The above threats and considerations have resulted in a set
of security functions and mechanisms built into or used by the
protocol. The signaling protocol relies on two security features
defined in the <xref target="sec_security-framework">Security
Framework</xref> namely client authentication using HTTP
authentication and TLS based transport protection of the signaling
messages. Both of these mechanisms are required to be implemented by
any RTSP agent.</t>
<t>A number of different security mitigations have been designed into
the protocol and will be instantiated if the specification is
implemented as written, for example by ensuring sufficient amount of
entropy in the randomly generated session identifiers when not using
client authentication to minimize the risk of session hijacking. When
client authentication is used the protection against hijacking will be
greatly improved by scoping the accessible sessions to the one this
client identity has created. Some of the above threats are such that
the implementation of the RTSP functionality itself needs to consider
which policy and strategy it uses to mitigate them.</t>
</section>
<section anchor="sec_media_sec_threats"
title="Media Stream Delivery Threats">
<t>The fact that RTSP establishes and controls a media stream delivery
results in a set of security issues related to the media streams. This
section will attempt to analyze general threats, however the choice of
media stream transport protocol, such as RTP will result in some
differences in threats and what mechanisms exist to mitigate them.
Thus it becomes important that each specification of a new media
stream transport and delivery protocol usable by RTSP requires its own
security analysis. This section includes one for RTP.</t>
<t>The set of general threats from or by the media stream delivery
itself are:<list style="hanging">
<t hangText="Concentrated denial-of-service attack:">The protocol
offers the opportunity for a remote-controlled denial-of-service
(DoS) attack, where the media stream is the hammer in that DoS
attack. See <xref target="sec-dos"/>.</t>
<t hangText="Media Confidentiality:">The media delivery may
contain content of any type and it is not possible in general to
determine how sensitive this content is from a confidentiality
point. Thus it is a strong requirement that any media delivery
protocol provides a method for providing confidentiality of the
actual media content. In addition to the media level
confidentiality it becomes critical that no resource identifiers
used in the signaling are exposed to an attacker as they may have
human understandable names, or may be also available to the
attacker so they can determine the content the user was delivered.
Thus the signaling protocol must also provide confidentiality
protection of any information related to the media resource.</t>
<t hangText="Media Integrity and Authentication:">There are
several reasons, such as discrediting the target, misinformation
of the target, why an attacker will be interested in substituting
the media stream sent out from the RTSP server with one of the
attacker's creation or selection. Therefore it is important that
the media protocol provides mechanisms to verify the source
authentication, integrity and prevent replay attacks on the media
stream.</t>
<t hangText="Scope of Multicast:">If RTSP is used to control the
transmission of media onto a multicast network the scope of the
delivery must be considered. RTSP supports the TTL Transport
header parameter to indicate this scope for IPv4. IPv6 has a
different mechanism for scope boundary. However, such scope
control has risks, as it may be set too large and distribute media
beyond the intended scope.</t>
</list></t>
<t><xref target="sec-rtp-sec">Below</xref> we do a protocol specific
analysis of security considerations for RTP based media transport. In
that section we also make clear the requirements on implementing
security functions for RTSP agents supporting media delivery over
RTP.</t>
<section anchor="sec-dos" title="Remote Denial of Service Attack">
<t>The attacker may initiate traffic flows to one or more IP
addresses by specifying them as the destination in SETUP requests.
While the attacker's IP address may be known in this case, this is
not always useful in prevention of more attacks or ascertaining the
attacker's identity. Thus, an RTSP server MUST only allow
client-specified destinations for RTSP-initiated traffic flows if
the server has ensured that the specified destination address
accepts receiving media through different security mechanisms.
Security mechanisms that are acceptable in order of increasing
generality are: <list style="symbols">
<t>Verification of the client's identity against a database of
known users using RTSP authentication mechanisms (preferably
digest authentication or stronger)</t>
<t>A list of addresses that have consented to be media
destinations, especially considering user identity</t>
<t>Media path based verification</t>
</list></t>
<t>The server SHOULD NOT allow the destination field to be set
unless a mechanism exists in the system to authorize the request
originator to direct streams to the recipient. It is preferred that
this authorization be performed by the media recipient (destination)
itself and the credentials passed along to the server. However, in
certain cases, such as when the recipient address is a multicast
group, or when the recipient is unable to communicate with the
server in an out-of-band manner, this may not be possible. In these
cases the server may chose another method such as a server-resident
authorization list to ensure that the request originator has the
proper credentials to request stream delivery to the recipient.</t>
<t>One solution that performs the necessary verification of
acceptance of media suitable for unicast based delivery is the <xref
target="RFC5245">Interactive Connectivity Establishment (ICE)</xref>
based NAT traversal method described in <xref
target="I-D.ietf-mmusic-rtsp-nat"/>. This mechanism uses random
passwords and a username so that the probability of unintended
indication as a valid media destination is very low. In addition the
server includes in its <xref target="RFC5389"> Session Traversal
Utilities for NAT (STUN)</xref> requests a cookie (consisting of
random material) that the destination echoes back, thus the solution
also safe-guards against having an off-path attacker being able to
spoof the STUN checks. This leaves this solution vulnerable only to
on-path attackers that can see the STUN requests go to the target of
attack and thus forge a response.</t>
<t>For delivery to multicast addresses there is a need for another
solution which is not specified in this memo.</t>
</section>
<section anchor="sec-rtp-sec" title="RTP Security analysis">
<t>RTP is a commonly used media transport protocol and has been the
most common choice for RTSP 1.0 implementations. The core RTP
protocol has been in use for a long time and it has well-known
security properties and the RTP security consideration (Section 9 of
<xref target="RFC3550"/>) needs to be reviewed. In perspective of
the usage of RTP in context of RTSP the following properties should
be noted:<list style="hanging">
<t hangText="Stream Additions:">RTP has support for multiple
simultaneous media streams in each RTP session. As some use
cases require support for non-synchronized adding and removal of
media streams and their identifiers an attacker can easily
insert additional media streams into a session context that
according to protocol design is intended to be played out.
Another threat vector is one of denial of service by exhausting
the resources of the RTP session receiver, for example by using
a large number of SSRC identifiers simultaneously. The strong
mitigation of this is to ensure that one cryptographically
authenticates any incoming packet flow to the RTP session. Weak
mitigations like blocking additional media streams in session
contexts easily lead to a denial of service vulnerability in
addition to preventing certain RTP extensions or use cases which
rely on multiple media streams, such as <xref
target="RFC4588">RTP retransmission</xref> to function.</t>
<t hangText="Forged Feedback:">The built in RTP control Protocol
(RTCP) also offers a large attack surface for a couple of
different types of attacks. One venue is to send RTCP feedback
to the media sender indicating large amounts of packet loss and
thus trigger a media bit-rate adaptation response from the
sender resulting in lowered media quality and potentially shut
down of the media stream. Another attack is to perform a
resource exhaustion attack on the receiver by using many SSRC
identifiers to create large state tables and increase the RTCP
related processing demands.</t>
<t hangText="RTP/RTCP Extensions:">RTP and RTCP extensions
generally provide additional and sometimes extremely powerful
tools to do denial of service or service disruption. For example
the <xref target="RFC5104">Code Control Message</xref> RTCP
extensions enables both locking down the bit-rate to low values
and disruption of video quality by requesting Intra frames.</t>
</list></t>
<t>Taking into account the above general discussion in <xref
target="sec_media_sec_threats"/> and the RTP specific discussion in
this section it is clear that it is necessary that a strong security
mechanism is supported to protect RTP. Therefore this specification
has the following requirements on RTP security functions for all
RTSP agents that handles media streams and where the media stream
transport is done using RTP.</t>
<t>RTSP agents supporting RTP MUST implement <xref target="RFC3711">
Secure RTP (SRTP)</xref> and thus the SAVP profile. In addition the
<xref target="RFC5124">secure AVP profile (SAVPF)</xref> MUST also
be supported if the AVPF profile is implemented. This specification
requires no additional cryptographic transforms or configuration
values beyond those specified as mandatory to implement in RFC3711,
i.e., AES-CM and HMAC-SHA1. The default key-management mechanism
which MUST be implemented is the one defined in the <xref
target="sec-mikey">MIKEY Key Establishment</xref>. The MIKEY
implementation MUST implement the necessary functions for <xref
target="RFC4738">MIKEY-RSA-R mode</xref> and in addition the SRTP
parameter negotiation necessary to negotiate the supported SRTP
transforms and parameters.</t>
</section>
</section>
</section>
<!-- title="Security Considerations" -->
<section anchor="sec_IANA" title="IANA Considerations">
<t>This section sets up a number of registries for RTSP 2.0 that should
be maintained by IANA. These registries are separate from any registries
existing for RTSP 1.0. For each registry there is a description of what
it is required to contain, what specification is needed when adding an
entry with IANA, and finally the entries that this document needs to
register. See also the <xref target="sec_extend-rtsp"/> "Extending
RTSP". There is also an IANA registration of three SDP attributes.</t>
<t>Registries or entries in registries which have been made for RTSP 1.0
are not moved to RTSP 2.0. The registries and entries in registries of
RTSP 1.0 and RTSP 2.0 are independent. If any registry or entry in a
registry is also required in RTSP 2.0, it MUST follow the procedure
defined below to allocate the registry or entry in a registry.</t>
<t>The sections describing how to register an item uses some of the
registration policies described in <xref target="RFC5226">RFC
5226</xref>, namely "First Come, First Served", "Expert Review,
"Specification Required", and "Standards Action".</t>
<t><list style="empty">
<t>RFC-Editor Note: Please replace all occurrences of RFCXXXX with
the RFC number this specification receives when published.</t>
</list>In case a registry requires a contact person, the authors, with
Magnus Westerlund (magnus.westerlund@ericsson.com) as primary, are the
contact persons for any entries created by this document.</t>
<t>A registration request to IANA MUST contain the following
information: <list hangIndent="3" style="symbols">
<t>A name of the item to register according to the rules specified
by the intended registry.</t>
<t>Indication of who has change control over the feature (for
example, IETF, ISO, ITU-T, other international standardization
bodies, a consortium, a particular company or group of companies, or
an individual);</t>
<t>A reference to a further description, if available, for example
(in decreasing order of preference) an RFC, a published standard, a
published paper, a patent filing, a technical report, documented
source code or a computer manual;</t>
<t>For proprietary features, contact information (postal and email
address);</t>
</list></t>
<section title="Feature-tags">
<section title="Description">
<t>When a client and server try to determine what part and
functionality of the RTSP specification and any future extensions
that its counter part implements there is need for a namespace. This
registry contains named entries representing certain
functionality.</t>
<t>The usage of feature-tags is explained in <xref
target="sec_capability"/> and <xref target="sec_OPTIONS"/>.</t>
</section>
<section title="Registering New Feature-tags with IANA">
<t>The registering of feature-tags is done on a first come, first
served basis.</t>
<t>The name of the feature MUST follow these rules: The name may be
of any length, but SHOULD be no more than twenty characters long.
The name MUST NOT contain any spaces, or control characters. The
registration MUST indicate if the feature-tag applies to clients,
servers, or proxies only or any combinations of these. Any
proprietary feature MUST have as the first part of the name a vendor
tag, which identifies the organization. The registry entries consist
of the feature tag, a one paragraph description of what it
represents, its applicability (server, client, proxy, any
combination) and a reference to its specification where
applicable.</t>
<t>Examples for a vendor tag describing a proprietary feature are:
<list hangIndent="6" style="hanging">
<t>vendorA.specfeat01</t>
<t>vendorA.specfeat02</t>
</list></t>
</section>
<section title="Registered entries">
<t>The following feature-tags are defined in this specification and
hereby registered. The change control belongs to the IETF. <list
hangIndent="6" style="hanging">
<t hangText="play.basic:">The implementation for delivery and
playback operations according to the core RTSP specification, as
defined in this memo. Applies for both clients, servers and
proxies. See <xref target="play.basic"/>.</t>
<t hangText="play.scale:">Support of scale operations for media
playback. Applies only for servers. See <xref
target="sec_Scale"/>.</t>
<t hangText="play.speed:">Support of the speed functionality for
media delivery. Applies only for servers. See <xref
target="sec_Speed"/>.</t>
<t hangText="setup.rtp.rtcp.mux">Support of the RTP and RTCP
multiplexing as discussed in <xref target="sec-rtp-rtcp-mux"/>.
Applies for both client and servers and any media caching
proxy.</t>
</list>This should be represented by IANA as a table with the
feature tags, contact person and their references.</t>
</section>
</section>
<section title="RTSP Methods">
<section title="Description">
<t>Methods are described in <xref target="sec_methods"/>. Extending
the protocol with new methods allow for totally new
functionality.</t>
</section>
<section title="Registering New Methods with IANA">
<t>A new method MUST be registered through an IETF Standards Action.
The reason is that new methods may radically change the protocol's
behavior and purpose.</t>
<t>A specification for a new RTSP method MUST consist of the
following items: <list hangIndent="3" style="symbols">
<t>A method name which follows the ABNF rules for methods.</t>
<t>A clear specification what a request using the method does
and what responses are expected. Which directions the method is
used, C->S or S->C or both. How the use of headers, if
any, modifies the behavior and effect of the method.</t>
<t>A list or table specifying which of the IANA registered
headers that are allowed to be used with the method in request
or/and response. The list or table SHOULD follow the format of
tables in <xref target="sec_headers"/>.</t>
<t>Describe how the method relates to network proxies.</t>
</list></t>
</section>
<section title="Registered Entries">
<t>This specification, RFCXXXX, registers 10 methods: DESCRIBE,
GET_PARAMETER, OPTIONS, PAUSE, PLAY, PLAY_NOTIFY, REDIRECT, SETUP,
SET_PARAMETER, and TEARDOWN. The initial table of the registry is
provided below.</t>
<figure>
<artwork><![CDATA[Method Directionality Reference
-----------------------------------------------------
DESCRIBE C->S [RFCXXXX]
GET_PARAMETER C->S, S->C [RFCXXXX]
OPTIONS C->S, S->C [RFCXXXX]
PAUSE C->S [RFCXXXX]
PLAY C->S [RFCXXXX]
PLAY_NOTIFY S->C [RFCXXXX]
REDIRECT S->C [RFCXXXX]
SETUP C->S [RFCXXXX]
SET_PARAMETER C->S, S->C [RFCXXXX]
TEARDOWN C->S, S->C [RFCXXXX] ]]></artwork>
</figure>
<t/>
</section>
</section>
<section title="RTSP Status Codes">
<section title="Description">
<t>A status code is the three digit number used to convey
information in RTSP response messages, see <xref
target="sec_response"/>. The number space is limited and care should
be taken not to fill the space.</t>
</section>
<section title="Registering New Status Codes with IANA">
<t>A new status code registration follows the policy of IETF Review.
New RTSP functionality requiring Status Codes should first be
registered in the range x50-x99. Only when the range is full should
registrations be done in the x00-x49 range, unless it is to adopt an
HTTP extension also to RTSP. The reason is to enable any HTTP
extension to be adopted to RTSP without needing to renumber any
related status codes. A specification for a new status code MUST
specify the following: <list hangIndent="3" style="symbols">
<t>The registered number.</t>
<t>A description of what the status code means and the expected
behavior of the sender and receiver of the code.</t>
</list></t>
</section>
<section title="Registered Entries">
<t>RFCXXXX, registers the numbered status code defined in the ABNF
entry "Status-Code" except "extension-code" (that defines the syntax
allowed for future extensions) in <xref
target="sec_syntax-prot-message"/>.</t>
</section>
</section>
<section title="RTSP Headers">
<section title="Description">
<t>By specifying new headers a method(s) can be enhanced in many
different ways. An unknown header will be ignored by the receiving
agent. If the new header is vital for a certain functionality, a
feature-tag for the functionality can be created and demanded to be
used by the counter-part with the inclusion of a Require header
carrying the feature-tag.</t>
</section>
<section title="Registering New Headers with IANA">
<t>Registrations in the registry can be done following the Expert
Review policy. A specification SHOULD be provided, preferably an
IETF RFC or other Standards Developing Organization specification.
The minimal information in a registration request is the header name
and the contact information.</t>
<t>The specification SHOULD contain the following information: <list
hangIndent="3" style="symbols">
<t>The name of the header.</t>
<t>An ABNF specification of the header syntax.</t>
<t>A list or table specifying when the header may be used,
encompassing all methods, their request or response, the
direction (C->S or S->C).</t>
<t>How the header is to be handled by proxies.</t>
<t>A description of the purpose of the header.</t>
</list></t>
</section>
<section title="Registered entries">
<t>All headers specified in <xref target="sec_headers"/> in RFCXXXX
are to be registered. The Registry is to include header name and
reference.</t>
<t>Furthermore the following legacy RTSP headers defined in other
specifications are registered with header name, reference and
description according to below list. Note: These references may not
fulfill all of the above rules for registrations due to their legacy
status. <list hangIndent="3" style="symbols">
<t>x-wap-profile defined in <xref target="TS-26234"/>. The
x-wap-profile request-header contains one or more absolute URLs
to the requesting agent's device capability profile.</t>
<t>x-wap-profile-diff defined in <xref target="TS-26234"/>. The
x-wap-profile-diff request-header contains a subset of a device
capability profile.</t>
<t>x-wap-profile-warning defined in <xref target="TS-26234"/>.
The x-wap-profile-warning is a response-header that contains
error codes explaining to what extent the server has been able
to match the terminal request in regards to device capability
profile as described using x-wap-profile and x-wap-profile-diff
headers.</t>
<t>x-predecbufsize defined in <xref target="TS-26234"/>. This
response-header provides an RTSP agent with the TS 26.234 Annex
G hypothetical pre-decoder buffer size.</t>
<t>x-initpredecbufperiod defined in <xref target="TS-26234"/>.
This response-header provides an RTSP agent with the TS 26.234
Annex G hypothetical pre-decoder buffering period.</t>
<t>x-initpostdecbufperiod defined in <xref target="TS-26234"/>.
This response-header provides an RTSP agent with the TS 26.234
Annex G post-decoder buffering period.</t>
<t>3gpp-videopostdecbufsize defined in <xref
target="TS-26234"/>. This response-header provides an RTSP agent
with the TS 26.234 defined post-decoder buffer size usable for
H.264 (AVC) video streams.</t>
<t>3GPP-Link-Char defined in <xref target="TS-26234"/>. This
request-header provides the RTSP server with the RTSP client's
link characteristics as determined from the radio interface. The
information that can be provided are guaranteed bit-rate,
maximum bit-rate and maximum transfer delay.</t>
<t>3GPP-Adaptation defined in <xref target="TS-26234"/>. This
general-header is part of the bit-rate adaptation solution
specified for PSS. It provides the RTSP client's buffer sizes
and target buffer levels to the server and responses are used to
acknowledge the support and values.</t>
<t>3GPP-QoE-Metrics defined in <xref target="TS-26234"/>. This
general-header is used by PSS RTSP agents to negotiate the
quality of experience metrics that a client should gather and
report to the server.</t>
<t>3GPP-QoE-Feedback defined in <xref target="TS-26234"/>. This
request-header is used by RTSP clients supporting PSS to report
the actual values of the metrics gathered in its quality of
experience metering.</t>
</list></t>
<t>The use of "x-" is NOT RECOMMENDED but the above headers in the
register list were defined prior to the clarification.</t>
</section>
</section>
<section title="Accept-Credentials">
<t>The security framework's TLS connection mechanism has two
registerable entities.</t>
<section title="Accept-Credentials policies">
<t>In <xref target="sec-frame-accept-cred"/> three policies for how
to handle certificates are specified. Further policies may be
defined and MUST be registered with IANA using the following rules:
<list hangIndent="3" style="symbols">
<t>Registering requires an IETF Standards Action</t>
<t>A registration is required to name a contact person.</t>
<t>Name of the policy.</t>
<t>A describing text that explains how the policy works for
handling the certificates.</t>
</list></t>
<t>This specification registers the following values: <list
hangIndent="6" style="empty">
<t hangText="">Any</t>
<t hangText="">Proxy</t>
<t hangText="">User</t>
</list></t>
</section>
<section title="Accept-Credentials hash algorithms">
<t>The Accept-Credentials header (See <xref
target="sec_Accept-Credentials"/>) allows for the usage of other
algorithms for hashing the DER records of accepted entities. The
registration of any future algorithm is expected to be extremely
rare and could also cause interoperability problems. Therefore the
bar for registering new algorithms is intentionally placed high.</t>
<t>Any registration of a new hash algorithm MUST fulfill the
following requirement: <list hangIndent="3" style="symbols">
<t>Follow the IETF Standards Action policy.</t>
<t>A definition of the algorithm and its identifier meeting the
"token" ABNF requirement.</t>
</list>The registered value is:</t>
<figure>
<artwork><![CDATA[Hash Alg. Id Reference
------------------------
sha-256 [RFCXXXX]]]></artwork>
</figure>
<t/>
</section>
</section>
<section title="Cache-Control Cache Directive Extensions">
<t>There exists a number of cache directives which can be sent in the
Cache-Control header. A registry for these cache directives MUST be
defined with the following rules: <list hangIndent="3" style="symbols">
<t>Registering requires an IETF Standards Action or IESG
Approval.</t>
<t>A registration is required to contain a contact person.</t>
<t>Name of the directive and a definition of the value, if
any.</t>
<t>Specification if it is a request or response directive.</t>
<t>A describing text that explains how the cache directive is used
for RTSP controlled media streams.</t>
</list></t>
<t>This specification registers the following values: <list
hangIndent="6" style="empty">
<t hangText="">no-cache:</t>
<t hangText="">public:</t>
<t hangText="">private:</t>
<t hangText="">no-transform:</t>
<t hangText="">only-if-cached:</t>
<t hangText="">max-stale:</t>
<t hangText="">min-fresh:</t>
<t hangText="">must-revalidate:</t>
<t hangText="">proxy-revalidate:</t>
<t hangText="">max-age:</t>
</list>The registry should be represented as: Name of the directive,
contact person and reference.</t>
</section>
<section title="Media Properties">
<t/>
<section title="Description">
<t>The media streams being controlled by RTSP can have many
different properties. The media properties required to cover the use
cases that were in mind when writing the specification are defined.
However, it can be expected that further innovation will result in
new use cases or media streams with properties not covered by the
ones specified here. Thus new media properties can be specified. As
new media properties may need a substantial amount of new
definitions to correctly specify behavior for this property the bar
is intended to be high.</t>
</section>
<section title="Registration Rules">
<t>Registering a new media property MUST fulfill the following
requirements</t>
<t><list style="symbols">
<t>Follow the Specification Required policy and get the approval
of the designated Expert.</t>
<t>Have an ABNF definition of the media property value name that
meets "media-prop-ext" definition</t>
<t>Define which media property group it belongs to or define a
new group.</t>
<t>A Contact Person for the Registration</t>
<t>Description of all changes to the behavior of the RTSP
protocol as result of these changes.</t>
</list></t>
</section>
<section title="Registered Values">
<t>This specification registers the ten values listed in <xref
target="sec_Media-Properties"/>. The registry should be represented
as: Name of the media property, property group, contact person and
reference.</t>
</section>
</section>
<section anchor="sec_iana_Notify-Reason_header"
title="Notify-Reason header">
<t/>
<section title="Description">
<t>Notify-Reason values are used for indicating the reason the
notification was sent. Each reason has its associated rules on what
headers and information that may or must be included in the
notification. New notification behaviors need to be specified to
enable interoperable usage, thus a specification of each new value
is required.</t>
</section>
<section title="Registration Rules">
<t>Registrations for new Notify-Reason value MUST fulfill the
following requirements</t>
<t><list style="symbols">
<t>Follow the Specification Required policy and get the approval
of the designated Expert.</t>
<t>An ABNF definition of the Notify reason value name that meets
"Notify-Reason-extension" definition</t>
<t>A Contact Person for the Registration</t>
<t>Description of which headers shall be included in the request
and response, when it should be sent, and any effect it has on
the server client state.</t>
</list></t>
</section>
<section title="Registered Values">
<t>This specification registers 3 values defined in the
Notify-Reas-val ABNF, <xref target="sec_syntax-prot-header"/>:</t>
<t><list style="hanging">
<t hangText="end-of-stream:">This Notify-Reason value indicates
the end of a media stream.</t>
<t hangText="media-properties-update:">This Notify-Reason value
allows the server to indicate that the properties of the media
has changed during the playout.</t>
<t hangText="scale-change:">This Notify-Reason value allows the
server to notify the client about a change in the Scale of the
media.</t>
</list>The registry entries should be represented in the registry
as: Name, short description, contact and reference.</t>
</section>
</section>
<section title="Range Header Formats">
<section title="Description">
<t>The <xref target="sec_Range">Range header</xref> allows for
different range formats. These range formats also needs an
identifier to be used the <xref
target="sec_Accept-Ranges">Accept-Ranges header</xref>. New range
formats may be registered, but moderation should be applied as it
makes interoperability more difficult.</t>
</section>
<section title="Registration Rules">
<t>A registration MUST fulfill the following requirements: <list
hangIndent="3" style="symbols">
<t>Follow the Specification Required policy.</t>
<t>An ABNF definition of the range format that fulfills the
"range-ext" definition.</t>
<t>Define the range format identifier used in Accept-Ranges
header according to the "extension-format" definition.</t>
<t>A Contact person for the registration.</t>
<t>Rules for how one handles the range when using a negative
Scale.</t>
</list></t>
</section>
<section title="Registered Values">
<t>The registry should be represented as: Range header format
identifier, Name of the range format, contact person and reference.
This specification registers the following values.</t>
<t><list style="hanging">
<t hangText="npt:">Normal Play Time</t>
<t hangText="clock:">UTC Absolute Time format</t>
<t hangText="smpte:">SMPTE Timestamps</t>
<t hangText="smpte-30-drop:">SMPTE Timestamps 29.97 Frames/sec
(30 Hz with Drop)</t>
<t hangText="smpte-25:">SMPTE Timestamps 25 Frames/sec</t>
</list></t>
</section>
</section>
<section title="Terminate-Reason Header">
<t>The <xref target="sec_Terminate-Reason">Terminate-Reason
header</xref> has two registries for extensions.</t>
<section title="Redirect Reasons">
<t>Registrations are done under the policy of Expert Review. The
registered value needs to follow the Terminate-Reason ABNF, i.e., be
a token. The specification needs to provide a definition of what
procedures are to be followed when a client receives this redirect
reason. This specification registers three values:</t>
<t><list style="symbols">
<t>Session-Timeout</t>
<t>Server-Admin</t>
<t>Internal-Error</t>
</list>The registry should be represented as: Name of the Redirect
Reason, contact person and reference.</t>
</section>
<section title="Terminate-Reason Header Parameters">
<t>Registrations are done under the policy of Specification
Required. The registrations must define a syntax for the parameter
that also follows the syntax allowed by the RTSP 2.0 specification.
A contact person is also required. This specification registers:</t>
<t><list style="symbols">
<t>time</t>
<t>user-msg</t>
</list>The registry should be represented as: Name of the
Terminate Reason, contact person and reference.</t>
</section>
</section>
<section title="RTP-Info header parameters">
<t/>
<section title="Description">
<t>The <xref target="sec_RTP-Info">RTP-Info header</xref> carries
one or more parameter value pairs with information about a
particular point in the RTP stream. RTP extensions or new usages may
need new types of information. As RTP information that could be
needed is likely to be generic enough and to maximize the
interoperability, new registration requires Specification
Required.</t>
</section>
<section title="Registration Rules">
<t>Registrations for new RTP-Info value MUST fulfill the following
requirements</t>
<t><list style="symbols">
<t>Follow the Specification Required policy and get the approval
of the designated Expert.</t>
<t>Have an ABNF definition that meets the "generic-param"
definition</t>
<t>A Contact Person for the Registration</t>
</list></t>
</section>
<section title="Registered Values">
<t>This specification registers the following parameter value
pairs:</t>
<t><list style="symbols">
<t>url</t>
<t>ssrc</t>
<t>seq</t>
<t>rtptime</t>
</list>The registry should be represented as: Name of the
parameter, contact person and reference.</t>
</section>
</section>
<section title="Seek-Style Policies">
<t/>
<section title="Description">
<t>New seek policies may be registered, however, a large number of
these will complicate implementation substantially. The impact of
unknown policies is that the server will not honor the unknown and
use the server default policy instead.</t>
</section>
<section title="Registration Rules">
<t>Registrations of new Seek-Style polices MUST fulfill the
following requirements</t>
<t><list style="symbols">
<t>Follow the Specification Required policy.</t>
<t>Have an ABNF definition of the Seek-Style policy name that
meets "Seek-S-value-ext" definition</t>
<t>A Contact Person for the Registration</t>
<t>Description of which headers shall be included in the request
and response, when it should be sent, and any affect it has on
the server client state.</t>
</list></t>
</section>
<section title="Registered Values">
<t>This specification registers 4 values:</t>
<t><list style="symbols">
<t>RAP</t>
<t>CoRAP</t>
<t>First-Prior</t>
<t>Next</t>
</list>The registry should be represented as: Name of the
Seek-Style Policy, short description, contact person and
reference.</t>
</section>
</section>
<section anchor="sec_IANA-trn" title="Transport Header Registries">
<t>The <xref target="sec_Transport">transport header</xref> contains a
number of parameters which have possibilities for future extensions.
Therefore registries for these need to be defined.</t>
<section title="Transport Protocol Identifier">
<t>A Transport Protocol Specification consists of a Transport
Protocol Identifier, representing some combination of transport
protocols, and any number of transport header parameters required or
optional to use with the identified protocol specification. This
registry contains the identifiers used by registered Transport
Protocol Identifiers.</t>
<t>A registry for the parameter transport protocol identifier MUST
be defined with the following rules: <list hangIndent="3"
style="symbols">
<t>Registering uses the policy of Specification Required.</t>
<t>A contact person or organization with address and email.</t>
<t>A value definition that are following the ABNF syntax
definition of "transport-id" <xref
target="sec_syntax-prot-header"/>.</t>
<t>A descriptive text that explains how the registered value are
used in RTSP, which underlying transport protocols that are
used, and any required Transport header parameters.</t>
</list>The registry should be represented as: The protocol ID
string, contact person and reference.</t>
<t>This specification registers the following values: <list
hangIndent="6" style="hanging">
<t hangText="RTP/AVP:">Use of the <xref
target="RFC3550">RTP</xref> protocol for media transport in
combination with the <xref target="RFC3551">"RTP profile for
audio and video conferences with minimal control"</xref> over
UDP. The usage is explained in RFC XXXX, <xref
target="sec_rtp"/>.</t>
<t hangText="RTP/AVP/UDP:">the same as RTP/AVP.</t>
<t hangText="RTP/AVPF:">Use of the <xref
target="RFC3550">RTP</xref> protocol for media transport in
combination with the <xref target="RFC4585">"Extended RTP
Profile for RTCP-based Feedback (RTP/AVPF)"</xref> over UDP. The
usage is explained in RFC XXXX, <xref target="sec_rtp"/>.</t>
<t hangText="RTP/AVPF/UDP:">the same as RTP/AVPF.</t>
<t hangText="RTP/SAVP:">Use of the <xref
target="RFC3550">RTP</xref> protocol for media transport in
combination with the <xref target="RFC3711">"The Secure
Real-time Transport Protocol (SRTP)"</xref> over UDP. The usage
is explained in RFC XXXX, <xref target="sec_rtp"/>.</t>
<t hangText="RTP/SAVP/UDP:">the same as RTP/SAVP.</t>
<t hangText="RTP/SAVPF:">Use of the RTP<xref target="RFC3550"/>
protocol for media transport in combination with the <xref
target="RFC5124">Extended Secure RTP Profile for Real-time
Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)</xref> over UDP. The usage is explained in RFC XXXX,
<xref target="sec_rtp"/>.</t>
<t hangText="RTP/SAVPF/UDP:">the same as RTP/SAVPF.</t>
<t hangText="RTP/AVP/TCP:">Use of the <xref
target="RFC3550">RTP</xref> protocol for media transport in
combination with the <xref target="RFC3551">"RTP profile for
audio and video conferences with minimal control"</xref> over
TCP. The usage is explained in RFC XXXX, <xref
target="sec_media-tcp-contrans"/>.</t>
<t hangText="RTP/AVPF/TCP:">Use of the <xref
target="RFC3550">RTP</xref> protocol for media transport in
combination with the <xref target="RFC4585">"Extended RTP
Profile for RTCP-based Feedback (RTP/AVPF)"</xref> over TCP. The
usage is explained in RFC XXXX, <xref
target="sec_media-tcp-contrans"/>.</t>
<t hangText="RTP/SAVP/TCP:">Use of the <xref
target="RFC3550">RTP</xref> protocol for media transport in
combination with the <xref target="RFC3711">"The Secure
Real-time Transport Protocol (SRTP)"</xref> over TCP. The usage
is explained in RFC XXXX, <xref
target="sec_media-tcp-contrans"/>.</t>
<t hangText="RTP/SAVPF/TCP:">Use of the <xref
target="RFC3550">RTP</xref> protocol for media transport in
combination with the "<xref target="RFC5124">Extended Secure RTP
Profile for Real-time Transport Control Protocol (RTCP)-Based
Feedback (RTP/SAVPF)"</xref> over TCP. The usage is explained in
RFC XXXX, <xref target="sec_media-tcp-contrans"/>.</t>
</list></t>
</section>
<section title="Transport modes">
<t>The Transport Mode is a <xref target="sec_Transport">Transport
header</xref> parameter, it is used to identify the general mode of
media transport. The PLAY value registered defines a PLAYBACK mode,
where media flows from Server to Client.</t>
<t>A registry for the transport parameter mode MUST be defined with
the following rules: <list hangIndent="3" style="symbols">
<t>Registering requires an IETF Standards Action.</t>
<t>A contact person or organization with address and email.</t>
<t>A value definition that are following the ABNF "token"
definition <xref target="sec_syntax-prot-header"/>.</t>
<t>A describing text that explains how the registered value are
used in RTSP.</t>
</list></t>
<t>This specification registers 1 value: <list hangIndent="6"
style="hanging">
<t hangText="PLAY:">See RFC XXXX.</t>
</list></t>
<t>The registry should be represented as: The Transport Mode value,
contact person and reference.</t>
</section>
<section title="Transport Parameters">
<t>Transport Parameters are different parameters used in a <xref
target="sec_Transport">Transport Header's transport
specification</xref> to provide additional information required
beyond the transport protocol identifier to establish a functioning
transport.</t>
<t>A registry for parameters that may be included in the Transport
header MUST be defined with the following rules: <list
hangIndent="3" style="symbols">
<t>Registering uses the Specification Required policy.</t>
<t>A Transport Parameter Name following the "token" ABNF
definition.</t>
<t>A value definition, if the parameter takes a value, that are
following the ABNF definition "trn-par-value" <xref
target="sec_syntax-prot-header"/>.</t>
<t>A describing text that explains how the registered value are
used in RTSP.</t>
</list> This specification registers all the transport parameters
defined in <xref target="sec_Transport"/>. This is a copy of this
list: <list style="symbols">
<t>unicast</t>
<t>multicast</t>
<t>interleaved</t>
<t>ttl</t>
<t>layers</t>
<t>ssrc</t>
<t>mode</t>
<t>dest_addr</t>
<t>src_addr</t>
<t>setup</t>
<t>connection</t>
<t>RTCP-mux</t>
<t>MIKEY</t>
</list>The registry should be represented as: The transport
parameter name, contact person and reference.</t>
</section>
</section>
<section anchor="sec_iana-uri-schemes" title="URI Schemes">
<t>This specification updates two URI schemes, one previously
registered "rtsp", and one missing in the registry "rtspu", previously
only defined in the <xref target="RFC2326">RTSP 1.0</xref>, one new
URI scheme "rtsps" is also registered. These URI schemes are
registered in an existing registry (Uniform Resource Identifier (URI)
Schemes) which is not created by this memo. Registrations are
following RFC 4395<xref target="RFC4395"/>.</t>
<section title="The rtsp URI Scheme">
<t><list hangIndent="6" style="hanging">
<t hangText="URI scheme name:">rtsp</t>
<t hangText="Status:">Permanent</t>
<t hangText="URI scheme syntax:">See <xref
target="sec_syntax-prot-generic"/> of RFC XXXX.</t>
<t hangText="URI scheme semantics:">The rtsp scheme is used to
indicate resources accessible through the usage of the Real-time
Streaming Protocol (RTSP). RTSP allows different operations on
the resource identified by the URI, but the primary purpose is
the streaming delivery of the resource to a client. However, the
operations that are currently defined are: DESCRIBE,
GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE, REDIRECT,
SETUP, SET_PARAMETER, and TEARDOWN.</t>
<t hangText="Encoding considerations:">IRIs in this scheme are
defined and needs to be encoded as RTSP URIs when used within
the RTSP protocol. That encoding is done according to RFC
3987.</t>
<t
hangText="Applications/protocols that use this URI scheme name:">RTSP
1.0 (RFC 2326), RTSP 2.0 (RFC XXXX)</t>
<t hangText="Interoperability considerations:">The extensions in
the URI syntax performed between RTSP 1.0 and 2.0 can create
interoperability issues. The changes are:<list style="hanging">
<t>Support for IPV6 literal in host part and future IP
literals through RFC 3986 defined mechanism.</t>
<t>A new relative format to use in the RTSP protocol
elements that is not required to start with "/".</t>
</list></t>
<t>The above changes should have no impact on interoperability
as in detail discussed in <xref target="sec_url"/> of
RFCXXXX.</t>
<t hangText="Security considerations:">All the security threats
identified in Section 7 of RFC 3986 apply also to this scheme.
They need to be reviewed and considered in any implementation
utilizing this scheme.</t>
<t hangText="Contact:">Magnus Westerlund,
magnus.westerlund@ericsson.com</t>
<t hangText="Author/Change controller:">IETF</t>
<t hangText="References:">RFC 2326, RFC 3986, RFC 3987, RFC
XXXX</t>
</list></t>
</section>
<section title="The rtsps URI Scheme">
<t><list hangIndent="6" style="hanging">
<t hangText="URI scheme name:">rtsps</t>
<t hangText="Status:">Permanent</t>
<t hangText="URI scheme syntax:">See <xref
target="sec_syntax-prot-generic"/> of RFC XXXX.</t>
<t hangText="URI scheme semantics:">The rtsps scheme is used to
indicate resources accessible through the usage of the Real-time
Streaming Protocol (RTSP) over TLS. RTSP allows different
operations on the resource identified by the URI, but the
primary purpose is the streaming delivery of the resource to a
client. However, the operations that are currently defined are:
DESCRIBE, GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE,
REDIRECT, SETUP, SET_PARAMETER, and TEARDOWN.</t>
<t hangText="Encoding considerations:">IRIs in this scheme are
defined and needs to be encoded as RTSP URIs when used within
the RTSP protocol. That encoding is done according to RFC
3987.</t>
<t
hangText="Applications/protocols that use this URI scheme name:">RTSP
1.0 (RFC 2326), RTSP 2.0 (RFC XXXX)</t>
<t hangText="Interoperability considerations:">The "rtsps"
scheme was never officially defined for RTSP 1.0, however it has
seen widespread use in actual deployments of RTSP 1.0. Therefore
this section discusses the believed changes between the
unspecified RTSP 1.0 "rtsps" scheme and RTSP 2.0 definition. The
extensions in the URI syntax performed between RTSP 1.0 and 2.0
can create interoperability issues. The changes are:<list
style="hanging">
<t>Support for IPV6 literal in host part and future IP
literals through RFC 3986 defined mechanism.</t>
<t>A new relative format to use in the RTSP protocol
elements that is not required to start with "/".</t>
</list></t>
<t>The above changes should have no impact on interoperability
as in detail discussed in <xref target="sec_url"/> of
RFCXXXX.</t>
<t hangText="Security considerations:">All the security threats
identified in Section 7 of RFC 3986 apply also to this scheme.
They need to be reviewed and considered in any implementation
utilizing this scheme.</t>
<t hangText="Contact:">Magnus Westerlund,
magnus.westerlund@ericsson.com</t>
<t hangText="Author/Change controller:">IETF</t>
<t hangText="References:">RFC 2326, RFC 3986, RFC 3987, RFC
XXXX</t>
</list></t>
</section>
<section title="The rtspu URI Scheme">
<t><list hangIndent="6" style="hanging">
<t hangText="URI scheme name:">rtspu</t>
<t hangText="Status:">Permanent</t>
<t hangText="URI scheme syntax:">See Section 3.2 of RFC
2326.</t>
<t hangText="URI scheme semantics:">The rtspu scheme is used to
indicate resources accessible through the usage of the Real-time
Streaming Protocol (RTSP) over unreliable datagram transport.
RTSP allows different operations on the resource identified by
the URI, but the primary purpose is the streaming delivery of
the resource to a client. However, the operations that are
currently defined are: DESCRIBE, GET_PARAMETER, OPTIONS,
REDIRECT,PLAY, PLAY_NOTIFY, PAUSE, SETUP, SET_PARAMETER, and
TEARDOWN.</t>
<t hangText="Encoding considerations:">This scheme is not
intended to be used with characters outside the US-ASCII
repertoire.</t>
<t
hangText="Applications/protocols that use this URI scheme name:">RTSP
1.0 (RFC 2326)</t>
<t hangText="Interoperability considerations:">The definition of
the transport mechanism of RTSP over UDP has interoperability
issues. That makes the usage of this scheme problematic.</t>
<t hangText="Security considerations:">All the security threats
identified in Section 7 of RFC 3986 apply also to this scheme.
They needs to be reviewed and considered in any implementation
utilizing this scheme.</t>
<t hangText="Contact:">Magnus Westerlund,
magnus.westerlund@ericsson.com</t>
<t hangText="Author/Change controller:">IETF</t>
<t hangText="References:">RFC 2326</t>
</list></t>
</section>
</section>
<section title="SDP attributes">
<t>This specification defines three SDP <xref target="RFC4566"/>
attributes that it is requested that IANA register.</t>
<figure>
<artwork><![CDATA[
SDP Attribute ("att-field"):
Attribute name: range
Long form: Media Range Attribute
Type of name: att-field
Type of attribute: Media and session level
Subject to charset: No
Purpose: RFC XXXX
Reference: RFC XXXX, RFC 2326
Values: See ABNF definition.
Attribute name: control
Long form: RTSP control URI
Type of name: att-field
Type of attribute: Media and session level
Subject to charset: No
Purpose: RFC XXXX
Reference: RFC XXXX, RFC 2326
Values: Absolute or Relative URIs.
Attribute name: mtag
Long form: Message Tag
Type of name: att-field
Type of attribute: Media and session level
Subject to charset: No
Purpose: RFC XXXX
Reference: RFC XXXX
Values: See ABNF definition
]]></artwork>
</figure>
</section>
<section anchor="sec_iana_textpar"
title="Media Type Registration for text/parameters">
<t/>
<t><list style="hanging">
<t hangText="Type name:">text</t>
<t hangText="Subtype name:">parameters</t>
<t hangText="Required parameters:"/>
<t hangText="Optional parameters:">charset: The charset parameter
is applicable to the encoding of the parameter values. The default
charset is UTF-8, if the 'charset' parameter is not present.</t>
<t hangText="Encoding considerations:">8bit</t>
<t hangText="Security considerations:">This format may carry any
type of parameters. Some can have security requirements, like
privacy, confidentiality or integrity requirements. The format has
no built in security protection. For the usage it was defined the
transport can be protected between server and client using TLS.
However, care must be taken to consider if also the proxies are
trusted with the parameters in case hop-by-hop security is used.
If stored as a file in file system, the necessary precautions need
to be taken in relation to the parameters requirements including
object security such as S/MIME <xref target="RFC5751"/>.</t>
<t hangText="Interoperability considerations:">This media type was
mentioned as a fictional example in <xref target="RFC2326"/>, but
was not formally specified. This has resulted in usage of this
media type which may not match its formal definition.</t>
<t hangText="Published specification:">RFC XXXX, <xref
target="sec_text-parameters"/>.</t>
<t hangText="Applications that use this media type:">Applications
that use RTSP and have additional parameters they like to read and
set using the RTSP GET_PARAMETER and SET_PARAMETER methods.</t>
<t hangText="Additional information:"/>
<t hangText="Magic number(s):"/>
<t hangText="File extension(s):"/>
<t hangText="Macintosh file type code(s): "/>
<t
hangText="Person & email address to contact for further information:">Magnus
Westerlund (magnus.westerlund@ericsson.com)</t>
<t hangText="Intended usage: ">Common</t>
<t hangText="Restrictions on usage: ">None</t>
<t hangText="Author:">Magnus Westerlund
(magnus.westerlund@ericsson.com)</t>
<t hangText="Change controller:">IETF</t>
<t hangText="Addition Notes:"/>
</list></t>
<t/>
</section>
</section>
<!-- title="IANA Considerations" -->
</middle>
<back>
<references title="Normative References">
<?rfc include="reference.RFC.5646" ?>
<?rfc include="reference.RFC.4566" ?>
<?rfc include="reference.RFC.3551" ?>
<?rfc include="reference.RFC.2616" ?>
<?rfc include="reference.RFC.5234" ?>
<?rfc include="reference.RFC.2119" ?>
<?rfc include="reference.RFC.5124" ?>
<?rfc include="reference.RFC.3987" ?>
<?rfc include="reference.RFC.3986" ?>
<?rfc include="reference.RFC.5246" ?>
<?rfc include="reference.RFC.2617" ?>
<?rfc include="reference.RFC.0768" ?>
<?rfc include="reference.RFC.0793" ?>
<?rfc include="reference.RFC.3629" ?>
<?rfc include="reference.RFC.5280" ?>
<?rfc include="reference.RFC.4648" ?>
<?rfc include='reference.RFC.4086'?>
<?rfc include="reference.RFC.4395" ?>
<reference anchor="TS-26234">
<front>
<title>Transparent end-to-end Packet-switched Streaming Service
(PSS); Protocols and codecs; Technical Specification 26.234</title>
<author initials="" surname="">
<organization>Third Generation Partnership Project
(3GPP)</organization>
</author>
<date month="December" year="2002"/>
</front>
</reference>
<reference anchor="FIPS-pub-180-2">
<front>
<title>Federal Information Processing Standards Publications (FIPS
PUBS) 180-2: Secure Hash Standard</title>
<author initials="" surname="">
<organization>National Institute of Standards and Technology
(NIST)</organization>
</author>
<date month="August" year="2002"/>
</front>
<format target="http://csrc.nist.gov/publications/fips/fips180-2/fips180-2.pdf"
type="PDF"/>
</reference>
<?rfc include="reference.RFC.3550" ?>
<?rfc include="reference.RFC.2818" ?>
<?rfc include="reference.RFC.4585" ?>
<?rfc include="reference.RFC.3711" ?>
<?rfc include="reference.RFC.3830" ?>
<?rfc include="reference.RFC.4571" ?>
<?rfc include='reference.RFC.6838'?>
<?rfc include="reference.RFC.4291" ?>
<?rfc include='reference.RFC.5322'?>
<?rfc include='reference.RFC.5751'?>
<?rfc include='reference.RFC.5226'?>
<?rfc include='reference.RFC.5761'?>
<?rfc include='reference.RFC.4738'?>
<?rfc include='reference.RFC.2460'?>
<?rfc include='reference.RFC.5888'?>
<?rfc include='reference.I-D.ietf-avtcore-rtp-circuit-breakers'?>
<reference anchor="SMPTE_TC">
<front>
<title>SMPTE Standard for Television -- Time and Control Code, ST
12M-1-2008</title>
<author fullname="">
<organization>Society of Motion Picture and Television
Engineers</organization>
</author>
<date/>
</front>
<format target="http://standards.smpte.org/content/978-1-61482-268-4/st-12-1-2008/SEC1.body.pdf+html"
type="HTML"/>
</reference>
</references>
<references title="Informative References">
<?rfc include="reference.RFC.1123" ?>
<?rfc include="reference.RFC.2068" ?>
<?rfc include="reference.RFC.2326" ?>
<?rfc include='reference.RFC.2663'?>
<?rfc include='reference.RFC.2974'?>
<?rfc include="reference.RFC.3261" ?>
<?rfc include='reference.RFC.3264'?>
<?rfc include='reference.RFC.4856'?>
<?rfc include="reference.RFC.4145" ?>
<?rfc include='reference.RFC.4567'?>
<?rfc include='reference.RFC.4588'?>
<?rfc include='reference.RFC.4855'?>
<?rfc include='reference.RFC.5104'?>
<?rfc include='reference.RFC.5245'?>
<?rfc include='reference.RFC.5389'?>
<?rfc include="reference.RFC.5905" ?>
<?rfc include="reference.RFC.6298" ?>
<?rfc include="reference.I-D.ietf-mmusic-rtsp-nat"?>
<reference anchor="ISO.13818-6.1995">
<front>
<title>Information technology - Generic coding of moving pictures
and associated audio information - part 6: Extension for digital
storage media and control</title>
<author>
<organization>International Organization for
Standardization</organization>
</author>
<date month="November" year="1995"/>
</front>
<seriesInfo name="ISO" value="Draft Standard 13818-6"/>
</reference>
<reference anchor="ISO.8601.2000">
<front>
<title>Data elements and interchange formats - Information
interchange - Representation of dates and times</title>
<author>
<organization>International Organization for
Standardization</organization>
</author>
<date month="December" year="2000"/>
</front>
<seriesInfo name="ISO/IEC" value="Standard 8601"/>
</reference>
<reference anchor="Stevens98">
<front>
<title>Unix Networking Programming - Volume 1, second
edition</title>
<author fullname="W. Richard Stevens" initials="W. R."
surname="Stevens">
<organization/>
</author>
<date year="1998"/>
</front>
</reference>
<?rfc include="reference.RFC.5583" ?>
<?rfc include='reference.RFC.3339'?>
<?rfc include='reference.RFC.0791'?>
</references>
<section anchor="sec_examples" title="Examples">
<t>This section contains several different examples trying to illustrate
possible ways of using RTSP. The examples can also help with the
understanding of how functions of RTSP work. However, remember that
these are examples and the normative and syntax description in the other
sections take precedence. Please also note that many of the examples
contain syntax illegal line breaks to accommodate the formatting
restriction that the RFC series impose.</t>
<section anchor="sec-examples-mod-unicast"
title="Media on Demand (Unicast)">
<t>This is an example of media on demand streaming of a media stored
in a container file. For purposes of this example, a container file is
a storage entity in which multiple continuous media types pertaining
to the same end-user presentation are present. In effect, the
container file represents an RTSP presentation, with each of its
components being RTSP controlled media streams. Container files are a
widely used means to store such presentations. While the components
are transported as independent streams, it is desirable to maintain a
common context for those streams at the server end.</t>
<t><list style="hanging">
<t>This enables the server to keep a single storage handle open
easily. It also allows treating all the streams equally in case of
any prioritization of streams by the server.</t>
</list></t>
<t>It is also possible that the presentation author may wish to
prevent selective retrieval of the streams by the client in order to
preserve the artistic effect of the combined media presentation.
Similarly, in such a tightly bound presentation, it is desirable to be
able to control all the streams via a single control message using an
aggregate URI.</t>
<t>The following is an example of using a single RTSP session to
control multiple streams. It also illustrates the use of aggregate
URIs. In a container file it is also desirable to not write any URI
parts which are not kept, when the container is distributed, like the
host and most of the path element. Therefore this example also uses
the "*" and relative URI in the delivered SDP.</t>
<t>Also this presentation description (SDP) is not cacheble, as the
Expires header is set to an equal value with date indicating immediate
expiration of its validity.</t>
<t>Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URI to the
container file.</t>
<figure>
<artwork><![CDATA[
C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Server: PhonyServer/1.0
Date: Thu, 24 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 271
Content-Base: rtsp://example.com/twister.3gp/
Expires: 24 Jan 1997 15:35:06 GMT
v=0
o=- 2890844256 2890842807 IN IP4 198.51.100.5
s=RTSP Session
i=An Example of RTSP Session Usage
e=adm@example.com
c=IN IP4 0.0.0.0
a=control: *
a=range:npt=0-0:10:34.10
t=0 0
m=audio 0 RTP/AVP 0
a=control: trackID=1
m=video 0 RTP/AVP 26
a=control: trackID=4]]></artwork>
</figure>
<figure>
<artwork><![CDATA[C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"
Accept-Ranges: npt, smpte, clock
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast; ssrc=93CB001E;
dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";
src_addr="198.51.100.5:9000"/"198.51.100.5:9001"
Session: 12345678
Expires: 24 Jan 1997 15:35:12 GMT
Date: 24 Jan 1997 15:35:12 GMT
Accept-Ranges: npt
Media-Properties: Random-Access=0.02, Immutable, Unlimited
C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
Session: 12345678
Accept-Ranges: npt, smpte, clock
M->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast; ssrc=A813FC13;
dest_addr="192.0.2.53:8002"/"192.0.2.53:8003";
src_addr="198.51.100.5:9002"/"198.51.100.5:9003";
Session: 12345678
Expires: 24 Jan 1997 15:35:13 GMT
Date: 24 Jan 1997 15:35:13 GMT
Accept-Range: NPT
Media-Properties: Random-Access=0.8, Immutable, Unlimited
]]></artwork>
</figure>
<figure>
<artwork><![CDATA[C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 4
User-Agent: PhonyClient/1.2
Range: npt=30-
Seek-Style: RAP
Session: 12345678
M->C: RTSP/2.0 200 OK
CSeq: 4
Server: PhonyServer/1.0
Date: 24 Jan 1997 15:35:14 GMT
Session: 12345678
Range: npt=30-634.10
Seek-Style: RAP
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12345;rtptime=3450012,
url="rtsp://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=54321;rtptime=2876889
C->M: PAUSE rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 5
User-Agent: PhonyClient/1.2
Session: 12345678
# Pause happens 0.87 seconds after starting to play
M->C: RTSP/2.0 200 OK
CSeq: 5
Server: PhonyServer/1.0
Date: 24 Jan 1997 15:36:01 GMT
Session: 12345678
Range: npt=30.87-634.10
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 6
User-Agent: PhonyClient/1.2
Range: npt=30.87-634.10
Seek-Style: Next
Session: 12345678
M->C: RTSP/2.0 200 OK
CSeq: 6
Server: PhonyServer/1.0
Date: 24 Jan 1997 15:36:01 GMT
Session: 12345678
Range: npt=30.87-634.10
Seek-Style: Next
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12555;rtptime=6330012,
url="rtsp://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=55021;rtptime=3132889
C->M: TEARDOWN rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 7
User-Agent: PhonyClient/1.2
Session: 12345678
M->C: RTSP/2.0 200 OK
CSeq: 7
Server: PhonyServer/1.0
Date: 24 Jan 1997 15:49:34 GMT
]]></artwork>
</figure>
<t/>
<t/>
</section>
<section anchor="sec-examples-mod-unicast-pipelining"
title="Media on Demand using Pipelining">
<t>This example is basically the example above (<xref
target="sec-examples-mod-unicast"/>), but now utilizing pipelining to
speed up the setup. It requires only two round trip times until the
media starts flowing. First of all, the session description is
retrieved to determine what media resources need to be setup. In the
second step, one sends the necessary SETUP requests and the PLAY
request to initiate media delivery.</t>
<t>Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URI to the
container file.</t>
<figure>
<artwork><![CDATA[
C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Server: PhonyServer/1.0
Date: Thu, 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 271
Content-Base: rtsp://example.com/twister.3gp/
Expires: 24 Jan 1997 15:35:06 GMT
v=0
o=- 2890844256 2890842807 IN IP4 192.0.2.5
s=RTSP Session
i=An Example of RTSP Session Usage
e=adm@example.com
c=IN IP4 0.0.0.0
a=control: *
a=range:npt=0-0:10:34.10
t=0 0
m=audio 0 RTP/AVP 0
a=control: trackID=1
m=video 0 RTP/AVP 26
a=control: trackID=4
C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"
Accept-Ranges: npt, smpte, clock
Pipelined-Requests: 7654
C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
Accept-Ranges: npt, smpte, clock
Pipelined-Requests: 7654
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 4
User-Agent: PhonyClient/1.2
Range: npt=0-
Seek-Style: RAP
Pipelined-Requests: 7654
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast;
dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";
src_addr="198.51.100.5:9000"/"198.51.100.5:9001";
ssrc=93CB001E
Session: 12345678
Expires: 24 Jan 1997 15:35:12 GMT
Date: 23 Jan 1997 15:35:12 GMT
Accept-Ranges: npt
Pipelined-Requests: 7654
Media-Properties: Random-Access=0.2, Immutable, Unlimited
M->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast;
dest_addr="192.0.2.53:8002"/"192.0.2.53:8003;
src_addr="198.51.100.5:9002"/"198.51.100.5:9003";
ssrc=A813FC13
Session: 12345678
Expires: 24 Jan 1997 15:35:13 GMT
Date: 23 Jan 1997 15:35:13 GMT
Accept-Range: NPT
Pipelined-Requests: 7654
Media-Properties: Random-Access=0.8, Immutable, Unlimited
M->C: RTSP/2.0 200 OK
CSeq: 4
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:14 GMT
Session: 12345678
Range: npt=0-623.10
Seek-Style: RAP
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12345;rtptime=3450012,
url="rtsp://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=54321;rtptime=2876889
Pipelined-Requests: 7654]]></artwork>
</figure>
</section>
<section anchor="sec-examples-mod-unicast-secured"
title="Secured Media Session for on Demand Content">
<t>This example is basically the above example (<xref
target="sec-examples-mod-unicast-pipelining"/>), but now including
establishment of SRTP crypto contexts to get a secured media delivery.
First of all, the client attempts to fetch this insecurely, but the
server redirects to a URI indicating a requirement on using a secure
connection for the RTSP messages. The client establishes a TCP/TLS
connections and the session description is retrieved to determine what
media resources need to be setup. In the this session description
secure media (SRTP) is indicated. In the next step, the client sends
the necessary SETUP requests including MIKEY messages. This is
pipeline with a PLAY request to initiate media delivery.</t>
<t>Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URI to the
container file.</t>
<t>Note: The MIKEY messages below are not valid MIKEY message and are
BASE64 encoded random data to represent where the MIKEY messages would
go.</t>
<figure>
<artwork><![CDATA[C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 301 Moved Permanently
CSeq: 1
Server: PhonyServer/1.0
Date: Thu, 23 Jan 1997 15:35:06 GMT
Location: rtsps://example.com/twister.3gp
C->M: Establish TCP/TLS connection and verify server's
certificate that is represents example.com.
Used for all below RTSP messages.
C->M: DESCRIBE rtsps://example.com/twister.3gp RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Date: Thu, 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 271
Content-Base: rtsps://example.com/twister.3gp/
Expires: 24 Jan 1997 15:35:06 GMT
v=0
o=- 2890844256 2890842807 IN IP4 192.0.2.5
s=RTSP Session
i=An Example of RTSP Session Usage
e=adm@example.com
c=IN IP4 0.0.0.0
a=control: *
a=range:npt=0-0:10:34.10
t=0 0
m=audio 0 RTP/SAVP 0
a=control: trackID=1
m=video 0 RTP/SAVP 26
a=control: trackID=4
C->M: SETUP rtsps://example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/SAVP;unicast;dest_addr=":8000"/":8001";
MIKEY=VGhpcyBpcyB0aGUgZmlyc3Qgc3RyZWFtcyBNSUtFWSBtZXNzYWdl
Accept-Ranges: npt, smpte, clock
Pipelined-Requests: 7654
C->M: SETUP rtsps://example.com/twister.3gp/trackID=4 RTSP/2.0
CSeq: 4
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/SAVP;unicast;dest_addr=":8002"/":8003";
MIKEY=TUlLRVkgZm9yIHN0cmVhbSB0d2lzdGVyLjNncC90cmFja0lEPTQ=
Accept-Ranges: npt, smpte, clock
Pipelined-Requests: 7654
C->M: PLAY rtsps://example.com/twister.3gp/ RTSP/2.0
CSeq: 5
User-Agent: PhonyClient/1.2
Range: npt=0-
Seek-Style: RAP
Pipelined-Requests: 7654
M->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Transport: RTP/SAVP;unicast;
dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";
src_addr="198.51.100.5:9000"/"198.51.100.5:9001";
ssrc=93CB001E;
MIKEY=TUlLRVkgUmVzcG9uc2UgdHdpc3Rlci4zZ3AvdHJhY2tJRD0x
Session: 12345678
Expires: 24 Jan 1997 15:35:12 GMT
Date: 23 Jan 1997 15:35:12 GMT
Accept-Ranges: npt
Pipelined-Requests: 7654
Media-Properties: Random-Access=0.2, Immutable, Unlimited
M->C: RTSP/2.0 200 OK
CSeq: 4
Server: PhonyServer/1.0
Transport: RTP/SAVP;unicast;
dest_addr="192.0.2.53:8002"/"192.0.2.53:8003;
src_addr="198.51.100.5:9002"/"198.51.100.5:9003";
ssrc=A813FC13;
MIKEY=TUlLRVkgUmVzcG9uc2UgdHdpc3Rlci4zZ3AvdHJhY2tJRD00
Session: 12345678
Expires: 24 Jan 1997 15:35:13 GMT
Date: 23 Jan 1997 15:35:13 GMT
Accept-Range: NPT
Pipelined-Requests: 7654
Media-Properties: Random-Access=0.8, Immutable, Unlimited
M->C: RTSP/2.0 200 OK
CSeq: 5
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:14 GMT
Session: 12345678
Range: npt=0-623.10
Seek-Style: RAP
RTP-Info: url="rtsps://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12345;rtptime=3450012,
url="rtsps://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=54321;rtptime=2876889;
Pipelined-Requests: 7654]]></artwork>
</figure>
</section>
<section title="Media on Demand (Unicast)">
<t>An alternative example of media on demand with a bit more tweaks is
the following. Client C requests a movie distributed from two
different media servers A (audio.example.com) and V (
video.example.com). The media description is stored on a web server W.
The media description contains descriptions of the presentation and
all its streams, including the codecs that are available, dynamic RTP
payload types, the protocol stack, and content information such as
language or copyright restrictions. It may also give an indication
about the timeline of the movie.</t>
<t>In this example, the client is only interested in the last part of
the movie.</t>
<figure>
<artwork><![CDATA[
C->W: GET /twister.sdp HTTP/1.1
Host: www.example.com
Accept: application/sdp
W->C: HTTP/1.1 200 OK
Date: Thu, 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 278
Expires: 23 Jan 1998 15:35:06 GMT
v=0
o=- 2890844526 2890842807 IN IP4 198.51.100.5
s=RTSP Session
e=adm@example.com
c=IN IP4 0.0.0.0
a=range:npt=0-1:49:34
t=0 0
m=audio 0 RTP/AVP 0
a=control:rtsp://audio.example.com/twister/audio.en
m=video 0 RTP/AVP 31
a=control:rtsp://video.example.com/twister/video
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057",
RTP/AVP/TCP;unicast;interleaved=0-1
Accept-Ranges: npt, smpte, clock
A->C: RTSP/2.0 200 OK
CSeq: 1
Session: 12345678
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3056"/"192.0.2.53:3057";
src_addr="198.51.100.5:5000"/"198.51.100.5:5001"
Date: 23 Jan 1997 15:35:12 GMT
Server: PhonyServer/1.0
Expires: 24 Jan 1997 15:35:12 GMT
Cache-Control: public
Accept-Ranges: npt, smpte
Media-Properties: Random-Access=0.02, Immutable, Unlimited
]]></artwork>
</figure>
<figure>
<artwork><![CDATA[C->V: SETUP rtsp://video.example.com/twister/video RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3058"/"192.0.2.53:3059",
RTP/AVP/TCP;unicast;interleaved=0-1
Accept-Ranges: npt, smpte, clock
V->C: RTSP/2.0 200 OK
CSeq: 1
Session: 23456789
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3058"/"192.0.2.53:3059";
src_addr="198.51.100.5:5002"/"198.51.100.5:5003"
Date: 23 Jan 1997 15:35:12 GMT
Server: PhonyServer/1.0
Cache-Control: public
Expires: 24 Jan 1997 15:35:12 GMT
Accept-Ranges: npt, smpte
Media-Properties: Random-Access=1.2, Immutable, Unlimited
C->V: PLAY rtsp://video.example.com/twister/video RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Session: 23456789
Range: smpte=0:10:00-
V->C: RTSP/2.0 200 OK
CSeq: 2
Session: 23456789
Range: smpte=0:10:00-1:49:23
Seek-Style: First-Prior
RTP-Info: url="rtsp://video.example.com/twister/video"
ssrc=A17E189D:seq=12312232;rtptime=78712811
Server: PhonyServer/2.0
Date: 23 Jan 1997 15:35:13 GMT]]></artwork>
</figure>
<figure>
<artwork><![CDATA[C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Session: 12345678
Range: smpte=0:10:00-
A->C: RTSP/2.0 200 OK
CSeq: 2
Session: 12345678
Range: smpte=0:10:00-1:49:23
Seek-Style: First-Prior
RTP-Info: url="rtsp://audio.example.com/twister/audio.en"
ssrc=3D124F01:seq=876655;rtptime=1032181
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:13 GMT
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: 12345678
A->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:36:52 GMT
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: 23456789
V->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/2.0
Date: 23 Jan 1997 15:36:52 GMT
]]></artwork>
</figure>
<t>Even though the audio and video track are on two different servers
that may start at slightly different times and may drift with respect
to each other over time, the client can perform initial
synchronization of the two media using RTP-Info and Range received in
the PLAY responses. If the two servers are time synchronized the RTCP
packets can also be used to maintain synchronization.</t>
</section>
<section title="Single Stream Container Files">
<t>Some RTSP servers may treat all files as though they are "container
files", yet other servers may not support such a concept. Because of
this, clients needs to use the rules set forth in the session
description for Request-URIs, rather than assuming that a consistent
URI may always be used throughout. Below is an example of how a
multi-stream server might expect a single-stream file to be served:
<figure>
<artwork><![CDATA[
C->S: DESCRIBE rtsp://foo.example.com/test.wav RTSP/2.0
Accept: application/x-rtsp-mh, application/sdp
CSeq: 1
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 1
Content-base: rtsp://foo.example.com/test.wav/
Content-type: application/sdp
Content-length: 163
Server: PhonyServer/1.0
Date: Thu, 23 Jan 1997 15:35:06 GMT
Expires: 23 Jan 1997 17:00:00 GMT
v=0
o=- 872653257 872653257 IN IP4 192.0.2.5
s=mu-law wave file
i=audio test
c=IN IP4 0.0.0.0
t=0 0
a=control: *
m=audio 0 RTP/AVP 0
a=control:streamid=0
]]></artwork>
</figure><figure>
<artwork><![CDATA[C->S: SETUP rtsp://foo.example.com/test.wav/streamid=0 RTSP/2.0
Transport: RTP/AVP/UDP;unicast;
dest_addr=":6970"/":6971";mode="PLAY"
CSeq: 2
User-Agent: PhonyClient/1.2
Accept-Ranges: npt, smpte, clock
S->C: RTSP/2.0 200 OK
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:6970"/"192.0.2.53:6971";
src_addr="198.51.100.5:6970"/"198.51.100.5:6971";
mode="PLAY";ssrc=EAB98712
CSeq: 2
Session: 2034820394
Expires: 23 Jan 1997 16:00:00 GMT
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:07 GMT
Accept-Ranges: npt
Media-Properties: Random-Acces=0.5, Immutable, Unlimited]]></artwork>
</figure><figure>
<artwork><![CDATA[
C->S: PLAY rtsp://foo.example.com/test.wav/ RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: 2034820394
S->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:08 GMT
Session: 2034820394
Range: npt=0-600
Seek-Style: RAP
RTP-Info: url="rtsp://foo.example.com/test.wav/streamid=0"
ssrc=0D12F123:seq=981888;rtptime=3781123]]></artwork>
</figure></t>
<t>Note the different URI in the SETUP command, and then the switch
back to the aggregate URI in the PLAY command. This makes complete
sense when there are multiple streams with aggregate control, but is
less than intuitive in the special case where the number of streams is
one. However, the server has declared the aggregated control URI in
the SDP and therefore this is legal.</t>
<t>In this case, it is also required that servers accept
implementations that use the non-aggregated interpretation and use the
individual media URI, like this:</t>
<figure>
<artwork><![CDATA[
C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: 2034820394
]]></artwork>
</figure>
</section>
<section anchor="sec_example-live-multicast"
title="Live Media Presentation Using Multicast">
<t>The media server M chooses the multicast address and port. Here, it
is assumed that the web server only contains a pointer to the full
description, while the media server M maintains the full description.
<figure>
<artwork><![CDATA[
C->W: GET /sessions.html HTTP/1.1
Host: www.example.com
W->C: HTTP/1.1 200 OK
Content-Type: text/html
<html>
...
<a href "rtsp://live.example.com/concert/audio">
Streamed Live Music performance </a>
...
</html>
]]></artwork>
</figure><figure>
<artwork><![CDATA[C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/2.0
CSeq: 1
Supported: play.basic, play.scale
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Content-Type: application/sdp
Content-Length: 183
Server: PhonyServer/1.0
Date: Thu, 23 Jan 1997 15:35:06 GMT
Supported: play.basic
v=0
o=- 2890844526 2890842807 IN IP4 192.0.2.5
s=RTSP Session
t=0 0
m=audio 3456 RTP/AVP 0
c=IN IP4 233.252.0.54/16
a=control: rtsp://live.example.com/concert/audio
a=range:npt=0-]]></artwork>
</figure><figure>
<artwork><![CDATA[C->M: SETUP rtsp://live.example.com/concert/audio RTSP/2.0
CSeq: 2
Transport: RTP/AVP;multicast;
dest_addr="233.252.0.54:3456"/"233.252.0.54:3457";ttl=16
Accept-Ranges: npt, smpte, clock
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Date: Thu, 23 Jan 1997 15:35:06 GMT
Transport: RTP/AVP;multicast;
dest_addr="233.252.0.54:3456"/"233.252.0.54:3457";ttl=16
;ssrc=4D12AB92/0DF876A3
Session: 0456804596
Accept-Ranges: npt, clock
Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0
]]></artwork>
</figure><figure>
<artwork><![CDATA[C->M: PLAY rtsp://live.example.com/concert/audio RTSP/2.0
CSeq: 3
Session: 0456804596
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:07 GMT
Session: 0456804596
Seek-Style: Next
Range:npt=1256-
RTP-Info: url="rtsp://live.example.com/concert/audio"
ssrc=0D12F123:seq=1473; rtptime=80000
]]></artwork>
</figure></t>
</section>
<section anchor="sec_capability-example" title="Capability Negotiation">
<t>This example illustrates how the client and server determine their
capability to support a special feature, in this case "play.scale".
The server, through the clients request and the included Supported
header, learns the client supports RTSP 2.0, and also supports the
playback time scaling feature of RTSP. The server's response contains
the following feature related information to the client; it supports
the basic media delivery functions (play.basic), the extended
functionality of time scaling of content (play.scale), and one
"example.com" proprietary feature (com.example.flight). The client
also learns the methods supported (Public header) by the server for
the indicated resource.</t>
<figure>
<artwork><![CDATA[
C->S: OPTIONS rtsp://media.example.com/movie/twister.3gp RTSP/2.0
CSeq: 1
Supported: play.basic, play.scale
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 1
Public:OPTIONS,SETUP,PLAY,PAUSE,TEARDOWN,DESCRIBE,GET_PARAMETER
Allow: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN, DESCRIBE
Server: PhonyServer/2.0
Supported: play.basic, play.scale, com.example.flight
]]></artwork>
</figure>
<t>When the client sends its SETUP request it tells the server that it
requires support of the play.scale feature for this session by
including the Require header.</t>
<figure>
<artwork><![CDATA[
C->S: SETUP rtsp://media.example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3056"/"192.0.2.53:3057",
RTP/AVP/TCP;unicast;interleaved=0-1
Require: play.scale
Accept-Ranges: npt, smpte, clock
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 3
Session: 12345678
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3056"/"192.0.2.53:3057";
src_addr="198.51.100.5:5000"/"198.51.100.5:5001"
Server: PhonyServer/2.0
Accept-Ranges: npt, smpte
Media-Properties: Random-Access=0.8, Immutable, Unlimited
]]></artwork>
</figure>
</section>
</section>
<!-- title="Examples" -->
<section anchor="sec_machine" title="RTSP Protocol State Machine">
<t>The RTSP session state machine describes the behavior of the protocol
from RTSP session initialization through RTSP session termination. It is
probably easiest to think of this as the server's state and then view
the the client as needing to track what it believes the server's state
will be based on sent or received RTSP messages. Thus in most cases the
state tables below can be read as: If the client does X, and assuming it
fulfills any pre-requisite(s), the (server) state will move to the new
state and the indicated response will returned. However, there are also
server to client notifications or requests, where the action describes
what notification or request will occur, its requisites and what new
state will result after the server has received the response, as well as
describing the client's response to the action.</t>
<t>The State machine is defined on a per session basis which is uniquely
identified by the RTSP session identifier. The session may contain one
or more media streams depending on state. If a single media stream is
part of the session it is in non-aggregated control. If two or more is
part of the session it is in aggregated control.</t>
<t>The below state machine is an informative description of the
protocols behavior. In case of ambiguity with the earlier parts of this
specification, the description in the earlier parts take precedence.</t>
<section title="States">
<t>The state machine contains three states, described below. For each
state there exists a table which shows which requests and events are
allowed and whether they will result in a state change. <list
hangIndent="6" style="hanging">
<t hangText="Init:">Initial state no session exists.</t>
<t hangText="Ready:">Session is ready to start playing.</t>
<t hangText="Play:">Session is playing, i.e., sending media stream
data in the direction S->C.</t>
</list></t>
</section>
<section title="State variables">
<t>This representation of the state machine needs more than its state
to work. A small number of variables are also needed and they are
explained below. <list hangIndent="6" style="hanging">
<t hangText="NRM:">The number of media streams part of this
session.</t>
<t hangText="RP:">Resume point, the point in the presentation time
line at which a request to continue playing will resume from. A
time format for the variable is not mandated.</t>
</list></t>
</section>
<section title="Abbreviations">
<t>To make the state tables more compact a number of abbreviations are
used, which are explained below. <list hangIndent="6" style="hanging">
<t hangText="IFI:">IF Implemented.</t>
<t hangText="md:">Media</t>
<t hangText="PP:">Pause Point, the point in the presentation time
line at which the presentation was paused.</t>
<t hangText="Prs:">Presentation, the complete multimedia
presentation.</t>
<t hangText="RedP:">Redirect Point, the point in the presentation
time line at which a REDIRECT was specified to occur.</t>
<t hangText="SES:">Session.</t>
</list></t>
</section>
<section title="State Tables">
<t>This section contains a table for each state. The table contains
all the requests and events that this state is allowed to act on. The
events which are method names are, unless noted, requests with the
given method in the direction client to server (C->S). In some
cases there exist one or more requisite. The response column tells
what type of response actions should be performed. Possible actions
that are requested for an event include: response codes, e.g., 200,
headers that need to be included in the response, setting of state
variables, or setting of other session related parameters. The new
state column tells which state the state machine changes to.</t>
<t>The response to a valid request meeting the requisites is normally
a 2xx (SUCCESS) unless otherwise noted in the response column. The
exceptions need to be given a response according to the response
column. If the request does not meet the requisite, is erroneous or
some other type of error occurs, the appropriate response code is to
be sent. If the response code is a 4xx the session state is unchanged.
A response code of 3rr will result in that the session is ended and
its state is changed to Init. A response code of 304 results in no
state change. However, there are restrictions to when a 3rr response
may be used. A 5xx response does not result in any change of the
session state, except if the error is not possible to recover from. A
unrecoverable error results in the ending of the session. As it in the
general case can't be determined if it was a unrecoverable error or
not the client will be required to test. In the case that the next
request after a 5xx is responded with 454 (Session Not Found) the
client knows that the session has ended. For any request message that
cannot be responded to within the time defined in <xref
target="sec_connection-timeout"/>, a 100 response must be sent.</t>
<t>The server will timeout the session after the period of time
specified in the SETUP response, if no activity from the client is
detected. Therefore there exists a timeout event for all states except
Init.</t>
<t>In the case that NRM = 1 the presentation URI is equal to the media
URI or a specified presentation URI. For NRM > 1 the presentation
URI needs to be other than any of the medias that are part of the
session. This applies to all states.</t>
<texttable anchor="tab_state-nochange"
title="None state-machine changing events">
<preamble/>
<ttcol align="left">Event</ttcol>
<ttcol align="left">Prerequisite</ttcol>
<ttcol align="left">Response</ttcol>
<c>DESCRIBE</c>
<c>Needs REDIRECT</c>
<c>3rr, Redirect</c>
<c>DESCRIBE</c>
<c/>
<c>200, Session description</c>
<c>OPTIONS</c>
<c>Session ID</c>
<c>200, Reset session timeout timer</c>
<c>OPTIONS</c>
<c/>
<c>200</c>
<c>SET_PARAMETER</c>
<c>Valid parameter</c>
<c>200, change value of parameter</c>
<c>GET_PARAMETER</c>
<c>Valid parameter</c>
<c>200, return value of parameter</c>
</texttable>
<t>The methods in <xref target="tab_state-nochange"/> do not have any
effect on the state machine or the state variables. However, some
methods do change other session related parameters, for example
SET_PARAMETER which will set the parameter(s) specified in its body.
Also all of these methods that allow Session header will also update
the keep-alive timer for the session.</t>
<texttable anchor="tab_state-init" title="State: Init">
<preamble/>
<ttcol align="left">Action</ttcol>
<ttcol align="left">Requisite</ttcol>
<ttcol align="left">New State</ttcol>
<ttcol align="left">Response</ttcol>
<c>SETUP</c>
<c/>
<c>Ready</c>
<c>NRM=1, RP=0.0</c>
<c>SETUP</c>
<c>Needs Redirect</c>
<c>Init</c>
<c>3rr Redirect</c>
<c>S -> C: REDIRECT</c>
<c>No Session hdr</c>
<c>Init</c>
<c>Terminate all SES</c>
</texttable>
<t>The initial state of the state machine, see <xref
target="tab_state-init"/> can only be left by processing a correct
SETUP request. As seen in the table the two state variables are also
set by a correct request. This table also shows that a correct SETUP
can in some cases be redirected to another URI and/or server by a 3rr
response.</t>
<texttable anchor="tab_state-ready" title="State: Ready">
<preamble/>
<ttcol align="left">Action</ttcol>
<ttcol align="left">Requisite</ttcol>
<ttcol align="left">New State</ttcol>
<ttcol align="left">Response</ttcol>
<c>SETUP</c>
<c>New URI</c>
<c>Ready</c>
<c>NRM +=1</c>
<c>SETUP</c>
<c>URI Setup prior</c>
<c>Ready</c>
<c>Change transport param</c>
<c>TEARDOWN</c>
<c>Prs URI,</c>
<c>Init</c>
<c>No session hdr, NRM = 0</c>
<c>TEARDOWN</c>
<c>md URI,NRM=1</c>
<c>Init</c>
<c>No Session hdr, NRM = 0</c>
<c>TEARDOWN</c>
<c>md URI,NRM>1</c>
<c>Ready</c>
<c>Session hdr, NRM -= 1</c>
<c>PLAY</c>
<c>Prs URI, No range</c>
<c>Play</c>
<c>Play from RP</c>
<c>PLAY</c>
<c>Prs URI, Range</c>
<c>Play</c>
<c>According to range</c>
<c>PLAY</c>
<c>md URI, NRM=1, Range</c>
<c>Play</c>
<c>According to range</c>
<c>PLAY</c>
<c>md URI, NRM=1</c>
<c>Play</c>
<c>Play from RP</c>
<c>PAUSE</c>
<c>Prs URI</c>
<c>Ready</c>
<c>Return PP</c>
<c>SC:REDIRECT</c>
<c>Terminate-Reason</c>
<c>Ready</c>
<c>Set RedP</c>
<c>SC:REDIRECT</c>
<c>No Terminate-Reason time parameter</c>
<c>Init</c>
<c>Session is removed</c>
<c>Timeout</c>
<c/>
<c>Init</c>
<c/>
<c>RedP reached</c>
<c/>
<c>Init</c>
<c>TEARDOWN of session</c>
</texttable>
<t>In the Ready state, see <xref target="tab_state-ready"/>, some of
the actions are depending on the number of media streams (NRM) in the
session, i.e., aggregated or non-aggregated control. A SETUP request
in the Ready state can either add one more media stream to the session
or, if the media stream (same URI) already is part of the session,
change the transport parameters. TEARDOWN is depending on both the
Request-URI and the number of media streams within the session. If the
Request-URI is the presentations URI the whole session is torn down.
If a media URI is used in the TEARDOWN request and more than one media
exists in the session, the session will remain and a session header is
returned in the response. If only a single media stream remains in the
session when performing a TEARDOWN with a media URI the session is
removed. The number of media streams remaining after tearing down a
media stream determines the new state.</t>
<texttable anchor="tab_state-play" title="State: Play">
<preamble/>
<ttcol align="left">Action</ttcol>
<ttcol align="left">Requisite</ttcol>
<ttcol align="left">New State</ttcol>
<ttcol align="left">Response</ttcol>
<c>PAUSE</c>
<c>Prs URI</c>
<c>Ready</c>
<c>Set RP to present point</c>
<c>End of media</c>
<c>All media</c>
<c>Play</c>
<c>Set RP = End of media</c>
<c>End of range</c>
<c/>
<c>Play</c>
<c>Set RP = End of range</c>
<c>PLAY</c>
<c>Prs URI, No range</c>
<c>Play</c>
<c>Play from present point</c>
<c>PLAY</c>
<c>Prs URI, Range</c>
<c>Play</c>
<c>According to range</c>
<c>SC:PLAY_NOTIFY</c>
<c/>
<c>Play</c>
<c>200</c>
<c>SETUP</c>
<c>New URI</c>
<c>Play</c>
<c>455</c>
<c>SETUP</c>
<c>Setuped URI</c>
<c>Play</c>
<c>455</c>
<c>SETUP</c>
<c>Setuped URI, IFI</c>
<c>Play</c>
<c>Change transport param.</c>
<c>TEARDOWN</c>
<c>Prs URI</c>
<c>Init</c>
<c>No session hdr</c>
<c>TEARDOWN</c>
<c>md URI,NRM=1</c>
<c>Init</c>
<c>No Session hdr, NRM=0</c>
<c>TEARDOWN</c>
<c>md URI</c>
<c>Play</c>
<c>455</c>
<c>SC:REDIRECT</c>
<c>Terminate Reason with Time parameter</c>
<c>Play</c>
<c>Set RedP</c>
<c>SC:REDIRECT</c>
<c/>
<c>Init</c>
<c>Session is removed</c>
<c>RedP reached</c>
<c/>
<c>Init</c>
<c>TEARDOWN of session</c>
<c>Timeout</c>
<c/>
<c>Init</c>
<c>Stop Media playout</c>
</texttable>
<t>The Play state table, see <xref target="tab_state-play"/>, contains
a number of requests that need a presentation URI (labeled as Prs URI)
to work on (i.e., the presentation URI has to be used as the
Request-URI). This is due to the exclusion of non-aggregated stream
control in sessions with more than one media stream.</t>
<t>To avoid inconsistencies between the client and server, automatic
state transitions are avoided. This can be seen at for example "End of
media" event when all media has finished playing, the session still
remains in Play state. An explicit PAUSE request needs to be sent to
change the state to Ready. It may appear that there exist automatic
transitions in "RedP reached" and "PP reached". However, they are
requested and acknowledged before they take place. The time at which
the transition will happen is known by looking at the range header. If
the client sends a request close in time to these transitions it needs
to be prepared for receiving error messages, as the state may or may
not have changed.</t>
</section>
</section>
<!-- title="RTSP Protocol State Machine" -->
<section anchor="sec_mediatran" title="Media Transport Alternatives">
<t>This section defines how certain combinations of protocols, profiles
and lower transports are used. This includes the usage of the Transport
header's source and destination address parameters "src_addr" and
"dest_addr".</t>
<section anchor="sec_rtp" title="RTP">
<t>This section defines the interaction of RTSP with respect to the
RTP protocol <xref target="RFC3550"/>. It also defines any necessary
media transport signaling with regards to RTP.</t>
<t>The available RTP profiles and lower layer transports are described
below along with rules on signaling the available combinations.</t>
<section anchor="sec_mediatran-rtp-avp" title="AVP">
<t>The usage of the "RTP Profile for Audio and Video Conferences
with Minimal Control" <xref target="RFC3551"/> when using RTP for
media transport over different lower layer transport protocols is
defined below in regards to RTSP.</t>
<t>One such case is defined within this document: the use of
embedded (interleaved) binary data as defined in <xref
target="sec_binary"/>. The usage of this method is indicated by
including the "interleaved" parameter.</t>
<t>When using embedded binary data the "src_addr" and "dest_addr"
MUST NOT be used. This addressing and multiplexing is used as
defined with use of channel numbers and the interleaved
parameter.</t>
</section>
<section title="AVP/UDP">
<t>This part describes sending of RTP <xref target="RFC3550"/> over
lower transport layer UDP <xref target="RFC0768"/> according to the
profile "RTP Profile for Audio and Video Conferences with Minimal
Control" defined in RFC 3551 <xref target="RFC3551"/>.
Implementations of RTP/AVP/UDP MUST implement <xref
target="sec_RTCP-usage-with-RTSP">RTCP</xref>. This profile requires
one or two uni- or bi-directional UDP flows per media stream. The
first UDP flow is for RTP and the second is for RTCP. Multiplexing
of <xref target="sec-rtp-rtcp-mux">RTP and RTCP</xref> MAY be used,
in which case a single UDP flow is used for both parts. Embedding of
RTP data with the RTSP messages, in accordance with <xref
target="sec_binary"/>, SHOULD NOT be performed when RTSP messages
are transported over unreliable transport protocols, like UDP <xref
target="RFC0768"/>.</t>
<t>The RTP/UDP and RTCP/UDP flows can be established using the
Transport header's "src_addr", and "dest_addr" parameters.</t>
<t>In RTSP PLAY mode, the transmission of RTP packets from client to
server is unspecified. The behavior in regards to such RTP packets
MAY be defined in future.</t>
<t>The "src_addr" and "dest_addr" parameters are used in the
following way for media delivery and playback mode, i.e., Mode=PLAY:
<list hangIndent="3" style="symbols">
<t>The "src_addr" and "dest_addr" parameters MUST contain either
1 or 2 address specifications. Note that two address
specifications MAY be provided even if RTP and RTCP multiplexing
is negotiated.</t>
<t>Each address specification for RTP/AVP/UDP or RTP/AVP/TCP
MUST contain either: <list hangIndent="3" style="symbols">
<t>both an address and a port number, or</t>
<t>a port number without an address.</t>
</list></t>
<t>The first address specification given in either of the
parameters applies to the RTP stream. The second specification
if present applies to the RTCP stream, unless in case RTP and
RTCP multiplexing is negotiated where both RTP and RTCP will use
the first specification.</t>
<t>The RTP/UDP packets from the server to the client MUST be
sent to the address and port given by the first address
specification of the "dest_addr" parameter.</t>
<t>The RTCP/UDP packets from the server to the client MUST be
sent to the address and port given by the second address
specification of the "dest_addr" parameter, unless RTP and RTCP
multiplexing has been negotiated, in which case RTCP MUST be
sent to the first address specification. If no second pair is
specified and RTP and RTCP multiplexing has not been negotiated,
RTCP MUST NOT be sent.</t>
<t>The RTCP/UDP packets from the client to the server MUST be
sent to the address and port given by the second address
specification of the "src_addr" parameter, unless RTP and RTCP
multiplexing has been negotiated, in which case RTCP MUST be
sent to the first address specification. If no second pair is
specified and RTP and RTCP multiplexing has not been negotiated,
RTCP MUST NOT be sent.</t>
<t>The RTP/UDP packets from the client to the server MUST be
sent to the address and port given by the first address
specification of the "src_addr" parameter.</t>
<t>RTP and RTCP Packets SHOULD be sent from the corresponding
receiver port, i.e., RTCP packets from the server should be sent
from the "src_addr" parameters second address port pair, unless
RTP and RTCP multiplexing has been negotiated in which case the
first address port pair is used.</t>
</list></t>
</section>
<section title="AVPF/UDP">
<t>The RTP profile <xref target="RFC4585">"Extended RTP Profile for
RTCP-based Feedback (RTP/AVPF)"</xref> MAY be used as RTP profiles
in sessions using RTP. All that is defined for AVP MUST also apply
for AVPF.</t>
<t>The usage of AVPF is indicated by the media initialization
protocol used. In the case of SDP it is indicated by media lines
(m=) containing the profile RTP/AVPF. That SDP MAY also contain
further AVPF related SDP attributes configuring the AVPF session
regarding reporting interval and feedback messages to be used <xref
target="RFC4585"/>. This configuration MUST be followed.</t>
</section>
<section anchor="sec-savp" title="SAVP/UDP">
<t>The RTP profile "The Secure Real-time Transport Protocol (SRTP)"
<xref target="RFC3711"/> is an RTP profile (SAVP) that MAY be used
in RTSP sessions using RTP. All that is defined for AVP MUST also
apply for SAVP.</t>
<t>The usage of SRTP requires that a security context is
established. The default key-management unless otherwise signalled
SHALL be MIKEY in RSA-R mode as defined in <xref
target="sec-mikey"/>, and not according to the procedure defined in
<xref target="RFC4567">"Key Management Extensions for Session
Description Protocol (SDP) and Real Time Streaming Protocol
(RTSP)"</xref>. The reason is that RFC 4567 sends the initial MIKEY
message in SDP, thus both requiring the usage of the DESCRIBE method
and forcing the server to keep state for clients performing DESCRIBE
in anticipation that they might require key management.</t>
<t>MIKEY is selected as default method for establishing SRTP
cryptographic context within an RTSP session as it can be embedded
in the RTSP messages, while still ensuring confidentiality of
content of the keying material, even when using hop-by-hop TLS
security for the RTSP messages. This method does also support
pipelining of the RTSP messages.</t>
<section anchor="sec-mikey" title="MIKEY Key Establishment">
<t>This method for using <xref target="RFC3830">MIKEY</xref> to
establish the SRTP cryptographic context is initiated in the
client's SETUP request, and the server's response to the SETUP
carries the MIKEY response. This ensures that the crypto context
establishment happens simultaneously with the establishment of the
media stream being protected. By using MIKEY's <xref
target="RFC4738">RSA-R mode</xref> the client can be the initiator
and still allow the server to set the parameters in accordance
with the actual media stream.</t>
<t>The SRTP cryptographic context establishment is done according
to the following process:</t>
<t><list style="numbers">
<t>The client determines that SAVP or SAVPF shall be used from
media description format, e.g., SDP. If no other key
management method is explicitly signalled, then MIKEY SHALL be
used as defined herein. The use of SRTP with RTSP is only
defined with MIKEY with keys established as defined in this
Section. Future documents may define how an RTSP
implementation treats SDP that indicates some other key
mechanism to be used. The need for such specification include
<xref target="RFC4567"/> that is not defined for use in RTSP
2.0 within this document.</t>
<t>The client SHALL establish a TLS connection for RTSP
messages, directly or hop by hop with the server. If
hop-by-hop TLS security is used, the User method SHALL be
indicated in the Accept-Credentials header. We do note that
using hop-by-hop does allow the proxy to insert itself as a
man in the middle also in the MIKEY exchange by providing one
of its certificates, rather than the server's in the
Connection-Credentials header. The client SHALL therefore
validate the server certificate.</t>
<t>The client retrieves the server's certificate from a direct
TLS connection, or if hop by hop from Connection-Credentials
header. The client then checks that the server certificate is
valid and belongs to the server.</t>
<t>The client forms the MIKEY Initiator message using RSA-R
mode in unicast mode as specified in <xref target="RFC4738"/>.
The client SHOULD use the same certificate for TLS and in
MIKEY to enable the server to bind the two together. The
client's certificate SHALL be included in the MIKEY message.
The client SHALL indicate its SRTP capabilities in the
message.</t>
<t>The MIKEY message from the previous step is <xref
target="RFC4648">base64</xref> encoded and becomes the value
of the MIKEY parameter that is included in the transport
specification(s) that specifies a SRTP based profile (SAVP,
SAVPF) in the SETUP request.</t>
<t>Any proxy encountering the MIKEY parameter SHALL forward it
without modification. A proxy requiring to understand
transport specification which doesn't support SAVP/SAVPF with
MIKEY will discard the whole transport specification. Most
types of proxies can easily support SAVP and SAVPF with MIKEY.
If possible bypassing the proxy should be tried.</t>
<t>The server upon receiving the SETUP request, will need to
decide upon the transport specification to use, if multiple
are included by the client. In the determination of which
transport specifications that are supported and preferred, the
server SHOULD decode the MIKEY message to take the embedded
SRTP parameters into account. If all transport specs require
SRTP but no MIKEY parameter or other supported keying method
is included, the server SHALL respond with 403.</t>
<t>Upon generating a response the following outcomes can
occur:<list style="symbols">
<t>A transport spec not using SRTP and MIKEY is selected.
Thus the response will not contain any MIKEY
parameter.</t>
<t>A transport spec using SRTP and MIKEY is selected but
an error is encountered in the MIKEY processing. In that
case an RTSP error response code of 466 "Key Management
Error" SHALL be used. A MIKEY message describing the error
MAY be included.</t>
<t>A transport spec using SRTP and MIKEY is selected and a
MIKEY response message can be created. The server SHOULD
use the same certificate for TLS and in MIKEY to enable
client to bind the two together. If a different
certificate is used it SHALL be included in the MIKEY
message. It is RECOMMENDED that the envelope key cache
type is set to ‘Cache’ and that a single
envelope key is reused for all MIKEY messages to the
client. That message is included in the MIKEY parameter
part of the single selected transport specification in the
SETUP response. The server will set the SRTP parameters as
preferred for this media stream within the supported range
by the client.</t>
</list></t>
<t>The server transmits the SETUP response back to the
client.</t>
<t>The client receives the SETUP response and if the response
code indicates a successful request it decodes the MIKEY
message and establishes the SRTP cryptographic context from
the parameters in the MIKEY response.</t>
</list>In the above method the client's certificate may be
self-signed in cases where the client's identity is not necessary
to authenticate and the security goal is only to ensure that the
RTSP signaling client is the same as the one receiving the SRTP
security context.</t>
</section>
</section>
<section title="SAVPF/UDP">
<t>The RTP profile "Extended Secure RTP Profile for RTCP-based
Feedback (RTP/SAVPF)" <xref target="RFC5124"/> is an RTP profile
(SAVPF) that MAY be used in RTSP sessions using RTP. All that is
defined for AVPF MUST also apply for SAVPF.</t>
<t>The usage of SRTP requires that a cryptographic context is
established. The default mechanism for establishing that security
association is to use MIKEY<xref target="RFC3830"/> with RTSP as
defined in <xref target="sec-mikey"/>.</t>
</section>
<section anchor="sec_RTCP-usage-with-RTSP"
title="RTCP usage with RTSP">
<t>RTCP has several usages when RTP is used for media transport as
explained below. Due to that RTCP MUST be supported if an RTSP agent
handles RTP.</t>
<section title="Media synchronization">
<t>RTCP provides media synchronization and clock drift
compensation. The initial media synchronization is available from
RTP-Info header. However, to be able to handle any clock drift
between the media streams, RTCP is needed.</t>
</section>
<section title="RTSP Session keep-alive">
<t>RTCP traffic from the RTSP client to the RTSP server MUST
function as keep-alive. This requires an RTSP server supporting
RTP to use the received RTCP packets as indications that the
client desires the related RTSP session to be kept alive.</t>
</section>
<section title="Bit-rate adaption">
<t>RTCP Receiver reports and any additional feedback from the
client MUST be used to adapt the bit-rate used over the transport
for all cases when RTP is sent over UDP. An RTP sender without
reserved resources MUST NOT use more than its fair share of the
available resources. This can be determined by comparing on short
to medium term (some seconds) the used bit-rate and adapt it so
that the RTP sender sends at a bit-rate comparable to what a TCP
sender would achieve on average over the same path.</t>
<t>To ensure that the implementation's adaptation mechanism has a
well defined outer envelope, all implementations using a
non-congestion controlled unicast transport protocol, like UDP,
MUST implement <xref
target="I-D.ietf-avtcore-rtp-circuit-breakers">Multimedia
Congestion Control: Circuit Breakers for Unicast RTP
Sessions</xref>.</t>
</section>
<section anchor="sec-rtp-rtcp-mux" title="RTP and RTCP Multiplexing">
<t>RTSP can be used to negotiate the usage of RTP and RTCP
multiplexing as described in <xref target="RFC5761"/>. This allows
servers and client to reduce the amount of resources required for
the session by only requiring one underlying transport stream per
media stream instead of two when using RTP and RTCP. This lessens
the server port consumption and also the necessary state and
keep-alive work when operating across <xref
target="RFC2663">Network and Address Translators</xref>.</t>
<t>Content must be prepared with some consideration for RTP and
RTCP multiplexing, mainly ensuring that the RTP payload types used
do not collide with the ones used for RTCP packet types. This
option likely needs explicit support from the content unless the
RTP payload types can be remapped by the server and that is
correctly reflected in the session description. Beyond that
support of this feature should come at little cost and much
gain.</t>
<t>It is recommended that if the content and server support RTP
and RTCP multiplexing that this is indicated in the session
description, for example using the SDP attribute "a=rtcp-mux". If
the SDP message contains the a=rtcp-mux attribute for a media
stream, the server MUST support RTP and RTCP multiplexing. If
indicated or otherwise desired by the client it can include the
Transport parameter "RTCP-mux" in any transport specification
where it desires to use RTCP-mux. The server will indicate if it
supports RTCP-mux. Servers and Clients SHOULD support RTP and RTCP
multiplexing.</t>
<t>For capability exchange, an RTSP feature tag for RTP and RTCP
multiplexing is defined: "setup.rtp.rtcp.mux".</t>
<t>To minimize the risk of negotiation failure while using RTP and
RTCP multiplexing some recommendations are here provided. If the
session description includes explicit indication of support
(a=rtcp-mux in SDP), then a RTSP agent can safely create a SETUP
request with a transport specification with only a single
dest_addr parameter address specification. If no such explicit
indication is provided, then even if the feature tag
"setup.rtp.rtcp.mux" is provided in a Supported header by the RTSP
server or the feature tag included in the Required header in the
SETUP request, the media resource may not support RTP and RTCP
multiplexing. Thus, to maximize the probability of successful
negotiation the RTSP agent is recommended to include two dest_addr
parameter address specifications in the first or first set (if
pipelining is used) of SETUP request(s) for any media resource
aggregate. That way the RTSP server can either accept RTP and RTCP
multiplexing and only use the first address specification, and if
not use both specifications. The RTSP agent after having received
the response for a successful negotiation of the usage of RTP and
RTCP multiplexing, can then release the resources associated with
the second address specification.</t>
</section>
</section>
</section>
<section title="RTP over TCP">
<t>Transport of RTP over TCP can be done in two ways: over independent
TCP connections using RFC 4571 <xref target="RFC4571"/> or interleaved
in the RTSP connection. In both cases the protocol MUST be "rtp" and
the lower layer MUST be TCP. The profile may be any of the above
specified ones; AVP, AVPF, SAVP or SAVPF.</t>
<section title="Interleaved RTP over TCP">
<t>The use of embedded (interleaved) binary data transported on the
RTSP connection is possible as specified in <xref
target="sec_binary"/>. When using this declared combination of
interleaved binary data the RTSP messages MUST be transported over
TCP. TLS may or may not be used. If TLS is used both RTSP messages
and the binary data will be protected by TLS.</t>
<t>One should, however, consider that this will result in all media
streams go through any proxy. Using independent TCP connections can
avoid that issue.</t>
</section>
<section anchor="sec_media-tcp-contrans"
title="RTP over independent TCP">
<t>In this Appendix, we describe the sending of RTP <xref
target="RFC3550"/> over lower transport layer TCP <xref
target="RFC0793"/> according to "Framing Real-time Transport
Protocol (RTP) and RTP Control Protocol (RTCP) Packets over
Connection-Oriented Transport" <xref target="RFC4571"/>. This
Appendix adapts the guidelines for using RTP over TCP within SIP/SDP
<xref target="RFC4145"/> to work with RTSP.</t>
<t>A client codes the support of RTP over independent TCP by
specifying an RTP/AVP/TCP transport option without an interleaved
parameter in the Transport line of a SETUP request. This transport
option MUST include the "unicast" parameter.</t>
<t>If the client wishes to use RTP with RTCP, two address
specifications needs to be included in the dest_addr parameter. If
the client wishes to use RTP without RTCP, one address specification
is included in the dest_addr parameter. If the client wishes to
multiplex RTP and RTCP on a single transport flow (see <xref
target="sec-rtp-rtcp-mux"/>), one or two address specifications are
included in the dest_addr parameter in addition to the RTCP-mux
transport parameter. Two address specifications are allowed to allow
successful negotiation when server or content can't support RTP and
RTCP multiplexing. Ordering rules of dest_addr ports follow the
rules for RTP/AVP/UDP.</t>
<t>If the client wishes to play the active role in initiating the
TCP connection, it MAY set the "setup" parameter (See <xref
target="sec_Transport"/>) on the Transport line to be "active", or
it MAY omit the setup parameter, as active is the default. If the
client signals the active role, the ports in the address
specifications in the dest_addr parameter MUST be set to 9 (the
discard port).</t>
<t>If the client wishes to play the passive role in TCP connection
initiation, it MUST set the "setup" parameter on the Transport line
to be "passive". If the client is able to assume the active or the
passive role, it MUST set the "setup" parameter on the Transport
line to be "actpass". In either case, the dest_addr parameter's
address specification port value for RTP MUST be set to the TCP port
number on which the client is expecting to receive the TCP
connection for RTP, and the dest_addr's address specification port
value for RTCP MUST be set to the TCP port number on which the
client is expecting to receive the TCP connection for RTCP. In the
case that the client wishes to multiplex RTP and RTCP on a single
transport flow, the RTCP-mux parameter is included and one or two
dest_addr parameter address specifications are included, as
mentioned earlier in this section.</t>
<t>If upon receipt of a non-interleaved RTP/AVP/TCP SETUP request, a
server decides to accept this requested option, the 2xx reply MUST
contain a Transport option that specifies RTP/AVP/TCP (without using
the interleaved parameter, and with using the unicast parameter).
The dest_addr parameter value MUST be echoed from the parameter
value in the client request unless the destination address (only
port) was not provided in which case the server MAY include the
source address of the RTSP TCP connection with the port number
unchanged.</t>
<t>In addition, the server reply MUST set the setup parameter on the
Transport line, to indicate the role the server will play in the
connection setup. Permissible values are "active" (if a client set
"setup" to "passive" or "actpass") and "passive" (if a client set
"setup" to "active" or "actpass").</t>
<t>If a server sets "setup" to "passive", the "src_addr" in the
reply MUST indicate the ports the server is willing to receive an
TCP connection for RTP and (if the client requested an TCP
connection for RTCP by specifying two dest_addr address
specifications) an TCP/RTCP connection. If a server sets "setup" to
"active", the ports specified in "src_addr" address specifications
MUST be set to 9. The server MAY use the "ssrc" parameter, following
the guidance in <xref target="sec_Transport"/>. The server sets only
one address specification in the case that the client has indicated
only a single address specification or in case RTP and RTCP
multiplexing was requested and accepted by server. Port ordering for
src_addr follows the rules for RTP/AVP/UDP.</t>
<t>Servers MUST support taking the passive role and MAY support
taking the active role. Servers with a public IP address take the
passive role, thus enabling clients behind NATs and Firewalls a
better chance of successful connect to the server by actively
connecting outwards. Therefore the clients are RECOMMENDED to take
the active role.</t>
<t>After sending (receiving) a 2xx reply for a SETUP method for a
non-interleaved RTP/AVP/TCP media stream, the active party SHOULD
initiate the TCP connection as soon as possible. The client MUST NOT
send a PLAY request prior to the establishment of all the TCP
connections negotiated using SETUP for the session. In case the
server receives a PLAY request in a session that has not yet
established all the TCP connections, it MUST respond using the 464
"Data Transport Not Ready Yet" (<xref target="sec_error464"/>) error
code.</t>
<t>Once the PLAY request for a media resource transported over
non-interleaved RTP/AVP/TCP occurs, media begins to flow from server
to client over the RTP TCP connection, and RTCP packets flow
bidirectionally over the RTCP TCP connection. Unless RTP and RTCP
multiplexing has been negotiated in which case RTP and RTCP will
flow over a common TCP connection. As in the RTP/UDP case, client to
server traffic on a RTP only TCP session is unspecified by this
memo. The packets that travel on these connections MUST be framed
using the protocol defined in <xref target="RFC4571"/>, not by the
framing defined for interleaving RTP over the RTSP connection
defined in <xref target="sec_binary"/>.</t>
<t>A successful PAUSE request for a media being transported over
RTP/AVP/TCP pauses the flow of packets over the connections, without
closing the connections. A successful TEARDOWN request signals that
the TCP connections for RTP and RTCP are to be closed by the RTSP
client as soon as possible.</t>
<t>Subsequent SETUP requests on an already-SETUP RTP/AVP/TCP URI may
be ambiguous in the following way: does the client wish to open up
new TCP connection for RTP or RTCP for the URI, or does the client
wish to continue using the existing TCP connections? The client
SHOULD use the "connection" parameter (defined in <xref
target="sec_Transport"/>) on the Transport line to make its
intention clear (by setting "connection" to "new" if new connections
are needed, and by setting "connection" to "existing" if the
existing connections are to be used). After a 2xx reply for a SETUP
request for a new connection, parties should close the pre-existing
connections, after waiting a suitable period for any stray RTP or
RTCP packets to arrive.</t>
<t>The usage of SRTP, i.e., either SAVP or SAVPF profiles, requires
that a security association is established. The default mechanism
for establishing that security association is to use MIKEY<xref
target="RFC3830"/> with RTSP as defined <xref
target="sec-mikey"/>.</t>
<t>Below, we rewrite part of the example media on demand example
shown in <xref target="sec-examples-mod-unicast"/> to use
RTP/AVP/TCP non-interleaved:</t>
<figure>
<artwork><![CDATA[
C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Server: PhonyServer/1.0
Date: Thu, 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 227
Content-Base: rtsp://example.com/twister.3gp/
Expires: 24 Jan 1997 15:35:06 GMT
v=0
o=- 2890844256 2890842807 IN IP4 198.51.100.34
s=RTSP Session
i=An Example of RTSP Session Usage
e=adm@example.com
c=IN IP4 0.0.0.0
a=control: *
a=range:npt=0-0:10:34.10
t=0 0
m=audio 0 RTP/AVP 0
a=control: trackID=1
C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP/TCP;unicast;dest_addr=":9"/":9";
setup=active;connection=new
Accept-Ranges: npt, smpte, clock]]></artwork>
</figure>
<figure>
<artwork><![CDATA[
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Transport: RTP/AVP/TCP;unicast;
dest_addr=":9"/":9";
src_addr="198.51.100.5:53478"/"198.51.100:54091";
setup=passive;connection=new;ssrc=93CB001E
Session: 12345678
Expires: 24 Jan 1997 15:35:12 GMT
Date: 23 Jan 1997 15:35:12 GMT
Accept-Ranges: npt
Media-Properties: Random-Access=0.8, Immutable, Unlimited
C->M: TCP Connection Establishment x2
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 4
User-Agent: PhonyClient/1.2
Range: npt=30-
Session: 12345678
M->C: RTSP/2.0 200 OK
CSeq: 4
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:14 GMT
Session: 12345678
Range: npt=30-623.10
Seek-Style: First-Prior
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=54321;rtptime=2876889
]]></artwork>
</figure>
</section>
</section>
<section title="Handling Media Clock Time Jumps in the RTP Media Layer">
<t>RTSP allows media clients to control selected, non-contiguous
sections of media presentations, rendering those streams with an <xref
target="RFC3550">RTP media layer</xref>. Two cases occur, the first is
when a new PLAY request replaces an old ongoing request and the new
request results in a jump in the media. This should produce in the RTP
layer a continuous media stream. A client may also directly following
a completed PLAY request perform a new PLAY request. This will result
in some gap in the media layer. The below text will look into both
cases.</t>
<t>A PLAY request that replaces an ongoing request allows the media
layer rendering the RTP stream without being affected by jumps in
media clock time. The RTP timestamps for the new media range is set so
that they become continuous with the previous media range in the
previous request. The RTP sequence number for the first packet in the
new range will be the next following the last packet in the previous
range, i.e., monotonically increasing. The goal is to allow the media
rendering layer to work without interruption or reconfiguration across
the jumps in media clock. This should be possible in all cases of
replaced PLAY requests for media that has random-access properties. In
this case care is needed to align frames or similar media dependent
structures.</t>
<t>In cases where jumps in media clock time are a result of RTSP
signaling operations arriving after a completed PLAY operation, the
request timing will result in that media becomes non-continuous. The
server becomes unable to send the media so that it arrives timely and
still carry timestamps to make the media stream continuous. In these
cases the server will produce RTP streams where there are gaps in the
RTP timeline for the media. In such cases, if the media has frame
structure, aligning the timestamp for the next frame with the previous
structure reduces the burden to render this media. The gap should
represent the time the server hasn't been serving media, e.g., the
time between the end of the media stream or a PAUSE request and the
new PLAY request. In these cases the RTP sequence number would
normally be monotonically increasing across the gap.</t>
<t>For RTSP sessions with media that lacks random access properties,
such as live streams, any media clock jump is commonly the result of a
correspondingly long pause of delivery. The RTP timestamp will have
increased in direct proportion to the duration of the paused delivery.
Note also that in this case the RTP sequence number should be the next
packet number. If not, the RTCP packet loss reporting will indicate as
loss all packets not received between the point of pausing and later
resuming. This may trigger congestion avoidance mechanisms. An allowed
exception from the above recommendation on monotonically increasing
RTP sequence number is live media streams, likely being relayed. In
this case, when the client resumes delivery, it will get the media
that is currently being delivered to the server itself. For this type
of basic delivery of live streams to multiple users over unicast,
individual rewriting of RTP sequence numbers becomes quite a burden.
For solutions that anyway caches media, timeshifts, etc, the rewriting
should be a minor issue.</t>
<t>The goal when handling jumps in media clock time is that the
provided stream is continuous without gaps in RTP timestamp or
sequence number. However, when delivery has been halted for some
reason the RTP timestamp when resuming MUST represent the duration the
delivery was halted. RTP sequence number MUST generally be the next
number, i.e., monotonically increasing modulo 65536. For media
resources with the properties Time-Progressing and Time-Duration=0.0
the server MAY create RTP media streams with RTP sequence number jumps
in them due to the client first halting delivery and later resuming it
(PAUSE and then later PLAY). However, servers utilizing this exception
must take into consideration the resulting RTCP receiver reports that
likely contain loss reports for all the packets part of the
discontinuity. A client cannot rely on that a server will align when
resuming playing even if it is RECOMMENDED. The RTP-Info header will
provide information on how the server acts in each case.</t>
<t><list style="hanging">
<t>We cannot assume that the RTSP client can communicate with the
RTP media agent, as the two may be independent processes. If the
RTP timestamp shows the same gap as the NPT, the media agent will
assume that there is a pause in the presentation. If the jump in
NPT is large enough, the RTP timestamp may roll over and the media
agent may believe later packets to be duplicates of packets just
played out. Having the RTP timestamp jump will also affect the
RTCP measurements based on this.</t>
</list></t>
<t>As an example, assume an RTP timestamp frequency of 8000 Hz, a
packetization interval of 100 ms and an initial sequence number and
timestamp of zero.</t>
<figure>
<artwork><![CDATA[
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 4
Session: abcdefgh
Range: npt=10-15
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 4
Session: abcdefgh
Range: npt=10-15
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=0;rtptime=0
]]></artwork>
</figure>
<t>The ensuing RTP data stream is depicted below:</t>
<figure>
<artwork><![CDATA[
S -> C: RTP packet - seq = 0, rtptime = 0, NPT time = 10s
S -> C: RTP packet - seq = 1, rtptime = 800, NPT time = 10.1s
. . .
S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s
]]></artwork>
</figure>
<t>Upon the completion of the requested delivery the server sends a
PLAY_NOTIFY</t>
<figure>
<artwork><![CDATA[ S->C: PLAY_NOTIFY rtsp://example.com/fizzle RTSP/2.0
CSeq: 5
Notify-Reason: end-of-stream
Request-Status: cseq=4 status=200 reason="OK"
Range: npt=-15
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=49;rtptime=39200
Session: abcdefgh
C->S: RTSP/2.0 200 OK
CSeq: 5
User-Agent: PhonyClient/1.2
]]></artwork>
</figure>
<t>Upon the completion of the play range, the client follows up with a
request to PLAY from a new NPT.</t>
<figure>
<artwork><![CDATA[
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 6
Session: abcdefg
Range: npt=18-20
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 6
Session: abcdefg
Range: npt=18-20
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=50;rtptime=40100]]></artwork>
</figure>
<t>The ensuing RTP data stream is depicted below:</t>
<figure>
<artwork><![CDATA[
S->C: RTP packet - seq = 50, rtptime = 40100, NPT time = 18s
S->C: RTP packet - seq = 51, rtptime = 40900, NPT time = 18.1s
. . .
S->C: RTP packet - seq = 69, rtptime = 55300, NPT time = 19.9s
]]></artwork>
</figure>
<t>In this example, first, NPT 10 through 15 is played, then the
client requests the server to skip ahead and play NPT 18 through 20.
The first segment is presented as RTP packets with sequence numbers 0
through 49 and timestamp 0 through 39,200. The second segment consists
of RTP packets with sequence number 50 through 69, with timestamps
40,100 through 55,200. While there is a gap in the NPT, there is no
gap in the sequence number space of the RTP data stream.</t>
<t>The RTP timestamp gap is present in the above example due to the
time it takes to perform the second play request, in this case 12.5 ms
(100/8000).</t>
</section>
<section title="Handling RTP Timestamps after PAUSE">
<t>During a PAUSE / PLAY interaction in an RTSP session, the duration
of time for which the RTP transmission was halted MUST be reflected in
the RTP timestamp of each RTP stream. The duration can be calculated
for each RTP stream as the time elapsed from when the last RTP packet
was sent before the PAUSE request was received and when the first RTP
packet was sent after the subsequent PLAY request was received. The
duration includes all latency incurred and processing time required to
complete the request.</t>
<t><list style="hanging">
<t>The RTP RFC <xref target="RFC3550"/> states that: The RTP
timestamp for each unit [packet] would be related to the wallclock
time at which the unit becomes current on the virtual presentation
timeline.</t>
<t>In order to satisfy the requirements of <xref
target="RFC3550"/>, the RTP timestamp space needs to increase
continuously with real time. While this is not optimal for stored
media, it is required for RTP and RTCP to function as intended.
Using a continuous RTP timestamp space allows the same timestamp
model for both stored and live media and allows better opportunity
to integrate both types of media under a single control.</t>
</list></t>
<t>As an example, assume a clock frequency of 8000 Hz, a packetization
interval of 100 ms and an initial sequence number and timestamp of
zero.</t>
<figure>
<artwork><![CDATA[
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 4
Session: abcdefg
Range: npt=10-15]]></artwork>
</figure>
<figure>
<artwork><![CDATA[ User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 4
Session: abcdefg
Range: npt=10-15
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=0;rtptime=0
]]></artwork>
</figure>
<t>The ensuing RTP data stream is depicted below:</t>
<figure>
<artwork><![CDATA[
S -> C: RTP packet - seq = 0, rtptime = 0, NPT time = 10s
S -> C: RTP packet - seq = 1, rtptime = 800, NPT time = 10.1s
S -> C: RTP packet - seq = 2, rtptime = 1600, NPT time = 10.2s
S -> C: RTP packet - seq = 3, rtptime = 2400, NPT time = 10.3s
]]></artwork>
</figure>
<t>The client then sends a PAUSE request:</t>
<figure>
<artwork><![CDATA[
C->S: PAUSE rtsp://example.com/fizzle RTSP/2.0
CSeq: 5
Session: abcdefg
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 5
Session: abcdefg
Range: npt=10.4-15]]></artwork>
</figure>
<t>20 seconds elapse and then the client sends a PLAY request. In
addition the server requires 15 ms to process the request:</t>
<figure>
<artwork><![CDATA[
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 6
Session: abcdefg
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 6
Session: abcdefg
Range: npt=10.4-15
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=4;rtptime=164400]]></artwork>
</figure>
<t>The ensuing RTP data stream is depicted below:</t>
<figure>
<artwork><![CDATA[
S -> C: RTP packet - seq = 4, rtptime = 164400, NPT time = 10.4s
S -> C: RTP packet - seq = 5, rtptime = 165200, NPT time = 10.5s
S -> C: RTP packet - seq = 6, rtptime = 166000, NPT time = 10.6s]]></artwork>
</figure>
<t>First, NPT 10 through 10.3 is played, then a PAUSE is received by
the server. After 20 seconds a PLAY is received by the server which
takes 15 ms to process. The duration of time for which the session was
paused is reflected in the RTP timestamp of the RTP packets sent after
this PLAY request.</t>
<t>A client can use the RTSP range header and RTP-Info header to map
NPT time of a presentation with the RTP timestamp.</t>
<t>Note: In RFC 2326 <xref target="RFC2326"/>, this matter was not
clearly defined and was misunderstood commonly. However, for RTSP 2.0
it is expected that this will be handled correctly and no exception
handling will be required.</t>
<t>Note further: It may be required to reset some of the state to
ensure the correct media decoding and the usual jitter-buffer handling
when issuing a PLAY request.</t>
</section>
<section title="RTSP / RTP Integration">
<t>For certain data types, tight integration between the RTSP layer
and the RTP layer will be necessary. This by no means precludes the
above restrictions. Combined RTSP/RTP media clients should use the
RTP-Info field to determine whether incoming RTP packets were sent
before or after a seek or before or after a PAUSE.</t>
</section>
<section title="Scaling with RTP">
<t>For scaling (see <xref target="sec_Scale"/>), RTP timestamps should
correspond to the rendering timing. For example, when playing video
recorded at 30 frames/second at a scale of two and speed (<xref
target="sec_Speed"/>) of one, the server would drop every second frame
to maintain and deliver video packets with the normal timestamp
spacing of 3,000 per frame, but NPT would increase by 1/15 second for
each video frame.</t>
<t><list style="hanging">
<t>Note: The above scaling puts requirements on the media codec or
a media stream to support it. For example motion JPEG or other
non-predictive video coding can easier handle the above
example.</t>
</list></t>
</section>
<section title="Maintaining NPT synchronization with RTP timestamps">
<t>The client can maintain a correct display of NPT (Normal Play Time)
by noting the RTP timestamp value of the first packet arriving after
repositioning. The sequence parameter of the RTP-Info (<xref
target="sec_RTP-Info"/>) header provides the first sequence number of
the next segment.</t>
</section>
<section title="Continuous Audio">
<t>For continuous audio, the server SHOULD set the RTP marker bit at
the beginning of serving a new PLAY request or at jumps in timeline.
This allows the client to perform playout delay adaptation.</t>
</section>
<section title="Multiple Sources in an RTP Session">
<t>Note that more than one SSRC MAY be sent in the media stream. If it
happens all sources are expected to be rendered simultaneously.</t>
</section>
<section title="Usage of SSRCs and the RTCP BYE Message During an RTSP Session">
<t>The RTCP BYE message indicates the end of use of a given SSRC. If
all sources leave an RTP session, it can, in most cases, be assumed to
have ended. Therefore, a client or server MUST NOT send an RTCP BYE
message until it has finished using a SSRC. A server SHOULD keep using
a SSRC until the RTP session is terminated. Prolonging the use of a
SSRC allows the established synchronization context associated with
that SSRC to be used to synchronize subsequent PLAY requests even if
the PLAY response is late.</t>
<t>An SSRC collision with the SSRC that transmits media does also have
consequences, as it will normally force the media sender to change its
SSRC in accordance with the RTP specification <xref
target="RFC3550"/>. However, an RTSP server may wait and see if the
client changes and thus resolve the conflict to minimize the impact.
As media sender SSRC change will result in a loss of synchronization
context, and require any receiver to wait for RTCP sender reports for
all media requiring synchronization before being able to play out
synchronized. Due to these reasons a client joining a session should
take care to not select the same SSRC(s) as the server indicates in
the ssrc Transport header parameter. Any SSRC signalled in the
Transport header MUST be avoided. A client detecting a collision prior
to sending any RTP or RTCP messages SHALL also select a new SSRC.</t>
</section>
<section title="Future Additions">
<t>It is the intention that any future protocol or profile regarding
media delivery and lower transport should be easy to add to RTSP. This
section provides the necessary steps that needs to be meet.</t>
<t>The following things needs to be considered when adding a new
protocol or profile for use with RTSP: <list hangIndent="3"
style="symbols">
<t>The protocol or profile needs to define a name tag representing
it. This tag is required to be an ABNF "token" to be possible to
use in the Transport header specification.</t>
<t>The useful combinations of protocol, profiles and lower layer
transport for this extension needs to be defined. For each
combination declare the necessary parameters to use in the
Transport header.</t>
<t>For new media protocols the interaction with RTSP needs to be
addressed. One important factor will be the media synchronization.
It may be necessary to have new headers similar to RTP info to
carry this information.</t>
<t>Discuss congestion control for media, especially if transport
without built in congestion control is used.</t>
</list></t>
<t>See the IANA section (<xref target="sec_IANA"/>) for information
how to register new attributes.</t>
</section>
</section>
<section anchor="sec_sdpusage"
title="Use of SDP for RTSP Session Descriptions">
<t>The Session Description Protocol (SDP, <xref target="RFC4566"/>) may
be used to describe streams or presentations in RTSP. This description
is typically returned in reply to a DESCRIBE request on a URI from a
server to a client, or received via HTTP from a server to a client.</t>
<t>This appendix describes how an SDP file determines the operation of
an RTSP session. Thus, it is worth pointing out that the interpretation
of the SDP is done in the context of the SDP receiver, which is the one
being configured. This is the same as in <xref
target="RFC2974">SAP</xref>; this differs from <xref
target="RFC3264">SDP Offer/Answer</xref> where each SDP is interpreted
in the context of the agent providing it.</t>
<t>SDP as is provides no mechanism by which a client can distinguish,
without human guidance, between several media streams to be rendered
simultaneously and a set of alternatives (e.g., two audio streams spoken
in different languages). The SDP extension "Grouping of Media Lines in
the Session Description Protocol (SDP)" <xref target="RFC5888"/>
provides such functionality to some degree. <xref
target="sec_sdp_m_grouping"/> describes the usage of SDP media line
grouping for RTSP.</t>
<section title="Definitions">
<t>The terms "session-level", "media-level" and other key/attribute
names and values used in this appendix are to be used as defined in
SDP<xref target="RFC4566"/>:</t>
<section anchor="sec_sdp-control-url" title="Control URI">
<t>The "a=control:" attribute is used to convey the control URI.
This attribute is used both for the session and media descriptions.
If used for individual media, it indicates the URI to be used for
controlling that particular media stream. If found at the session
level, the attribute indicates the URI for aggregate control
(presentation URI). The session level URI MUST be different from any
media level URI. The presence of a session level control attribute
MUST be interpreted as support for aggregated control. The control
attribute MUST be present on media level unless the presentation
only contains a single media stream, in which case the attribute MAY
be present on the session level only and then also apply to that
single media stream.</t>
<t>ABNF for the attribute is defined in <xref
target="sec_sdp-syntax"/>.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[ a=control:rtsp://example.com/foo]]></artwork>
</figure>
<t>This attribute MAY contain either relative or absolute URIs,
following the rules and conventions set out in RFC 3986 <xref
target="RFC3986"/>. Implementations MUST look for a base URI in the
following order: <list hangIndent="3" style="numbers">
<t>the RTSP Content-Base field;</t>
<t>the RTSP Content-Location field;</t>
<t>the RTSP Request-URI.</t>
</list>If this attribute contains only an asterisk (*), then the
URI MUST be treated as if it were an empty embedded URI, and thus
inherit the entire base URI.</t>
<t><list style="empty">
<t>Note, RFC 2326 was very unclear on the processing of relative
URI and several RTSP 1.0 implementations at the point of
publishing this document did not perform RFC 3986 processing to
determine the resulting URI, instead simple concatenation is
common. To avoid this issue completely it is recommended to use
absolute URI in the SDP.</t>
</list>The URI handling for SDPs from container files need special
consideration. For example let's assume that a container file has
the URI: "rtsp://example.com/container.mp4". Let's further assume
this URI is the base URI, and that there is an absolute media level
URI: "rtsp://example.com/container.mp4/trackID=2". A relative media
level URI that resolves in accordance with RFC 3986 <xref
target="RFC3986"/> to the above given media URI is:
"container.mp4/trackID=2". It is usually not desirable to need to
include in or modify the SDP stored within the container file with
the server local name of the container file. To avoid this, one can
modify the base URI used to include a trailing slash, e.g.,
"rtsp://example.com/container.mp4/". In this case the relative URI
for the media will only need to be: "trackID=2". However, this will
also mean that using "*" in the SDP will result in control URI
including the trailing slash, i.e.,
"rtsp://example.com/container.mp4/".</t>
<t><list style="hanging">
<t>Note: The usage of TrackID in the above is not a standardized
form, but one example out of several similar strings such as
TrackID, Track_ID, StreamID that is used by different server
vendors to indicate a particular piece of media inside a
container file.</t>
</list></t>
</section>
<section title="Media Streams">
<t>The "m=" field is used to enumerate the streams. It is expected
that all the specified streams will be rendered with appropriate
synchronization. If the session is over multicast, the port number
indicated SHOULD be used for reception. The client MAY try to
override the destination port, through the Transport header. The
servers MAY allow this, the response will indicate if allowed or
not. If the session is unicast, the port numbers are the ones
RECOMMENDED by the server to the client, about which receiver ports
to use; the client MUST still include its receiver ports in its
SETUP request. The client MAY ignore this recommendation. If the
server has no preference, it SHOULD set the port number value to
zero.</t>
<t>The "m=" lines contain information about which transport
protocol, profile, and possibly lower-layer is to be used for the
media stream. The combination of transport, profile and lower layer,
like RTP/AVP/UDP needs to be defined for how to be used with RTSP.
The currently defined combinations are defined in <xref
target="sec_mediatran"/>, further combinations MAY be specified.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[ m=audio 0 RTP/AVP 31
]]></artwork>
</figure>
</section>
<section title="Payload Type(s)">
<t>The payload type(s) are specified in the "m=" line. In case the
payload type is a static payload type from RFC 3551 <xref
target="RFC3551"/>, no other information may be required. In case it
is a dynamic payload type, the media attribute "rtpmap" is used to
specify what the media is. The "encoding name" within the "rtpmap"
attribute may be one of those specified in <xref target="RFC4856"/>,
or a media type registered with IANA according to <xref
target="RFC4855"/>, or an experimental encoding as specified in
<xref target="RFC4566">SDP</xref>). Codec-specific parameters are
not specified in this field, but rather in the "fmtp" attribute
described below.</t>
<t>The selection of the RTP payload type numbers used may be
required to consider <xref target="RFC5761">RTP and RTCP
Multiplexing</xref> if that is to be supported by the server.</t>
</section>
<section title="Format-Specific Parameters">
<t>Format-specific parameters are conveyed using the "fmtp" media
attribute. The syntax of the "fmtp" attribute is specific to the
encoding(s) that the attribute refers to. Note that some of the
format specific parameters may be specified outside of the fmtp
parameters, like for example the "ptime" attribute for most audio
encodings.</t>
</section>
<section anchor="sec_sdp-direction"
title="Directionality of media stream">
<t>The SDP attributes "a=sendrecv", "a=recvonly" and "a=sendonly"
provide instructions about the direction the media streams flow
within a session. When using RTSP the SDP can be delivered to a
client using either RTSP DESCRIBE or a number of RTSP external
methods, like HTTP, FTP, and email. Based on this the SDP applies to
how the RTSP client will see the complete session. Thus media
streams delivered from the RTSP server to the client, would be given
the "a=recvonly" attribute.</t>
<t>"a=recvonly" in a SDP provided to the RTSP client indicates that
media delivery will only occur in the direction from the RTSP server
to the client. SDP provided to the RTSP client that lacks any of the
directionality attributes (a=recvonly, a=sendonly, a=sendrecv) would
be interpreted as having a=sendrecv. At the time of writing there
exist no RTSP mode suitable for media traffic in the direction from
the RTSP client to the server. Thus all RTSP SDP SHOULD have
a=recvonly attribute when using the PLAY mode defined in this
document. If future modes are defined for media in client to server
direction, then usage of a=sendonly, or a=sendrecv may become
suitable to indicate intended media directions.</t>
</section>
<section anchor="sec_sdp-range" title="Range of Presentation">
<t>The "a=range" attribute defines the total time range of the
stored session or an individual media. Non-seekable live sessions
can be indicated as specified below, while the length of live
sessions can be deduced from the "t=" and "r=" SDP parameters.</t>
<t>The attribute is both a session and a media level attribute. For
presentations that contain media streams of the same duration, the
range attribute SHOULD only be used at session-level. In case of
different lengths the range attribute MUST be given at media level
for all media, and SHOULD NOT be given at session level. If the
attribute is present at both media level and session level the media
level values MUST be used.</t>
<t>Note: Usually one will specify the same length for all media,
even if there isn't media available for the full duration on all
media. However, that requires that the server accepts PLAY requests
within that range.</t>
<t>Servers MUST take care to provide RTSP Range (see <xref
target="sec_Range"/>) values that are consistent with what is
presented in the SDP for the content. There is no reason for non
dynamic content, like media clips provided on demand to have
inconsistent values. Inconsistent values between the SDP and the
actual values for the content handled by the server is likely to
generate some failure, like 457 "Invalid Range", in case the client
uses PLAY requests with a Range header. In case the content is
dynamic in length and it is infeasible to provide a correct value in
the SDP the server is recommended to describe this as non-seekable
content (see below). The server MAY override that property in the
response to a PLAY request using the correct values in the Range
header.</t>
<t>The unit is specified first, followed by the value range. The
units and their values are as defined in <xref target="sec_smpte"/>,
<xref target="sec_npt"/> and <xref target="sec_clock"/> and MAY be
extended with further formats. Any open ended range (start-), i.e.,
without stop range, is of unspecified duration and MUST be
considered as non-seekable content unless this property is
overridden. Multiple instances carrying different clock formats MAY
be included at either session or media level.</t>
<t>ABNF for the attribute is defined in <xref
target="sec_sdp-syntax"/>.</t>
<t>Examples:</t>
<figure>
<artwork><![CDATA[ a=range:npt=0-34.4368
a=range:clock=19971113T211503Z-19971113T220300Z
Non seekable stream of unknown duration:
a=range:npt=0-
]]></artwork>
</figure>
</section>
<section title="Time of Availability">
<t>The "t=" field defines when the SDP is valid. For on-demand
content the server SHOULD indicate a stop time value for which it
guarantees the description to be valid, and a start time that is
equal to or before the time at which the DESCRIBE request was
received. It MAY also indicate start and stop times of 0, meaning
that the session is always available.</t>
<t>For sessions that are of live type, i.e., specific start time,
unknown stop time, likely unseekable, the "t=" and "r=" field SHOULD
be used to indicate the start time of the event. The stop time
SHOULD be given so that the live event will have ended at that time,
while still not be unnecessary long into the future.</t>
</section>
<section title="Connection Information">
<t>In SDP used with RTSP, the "c=" field contains the destination
address for the media stream. If a multicast address is specified
the client SHOULD use this address in any SETUP request as
destination address, including any additional parameters, such as
TTL. For on-demand unicast streams and some multicast streams, the
destination address MAY be specified by the client via the SETUP
request, thus overriding any specified address. To identify streams
without a fixed destination address, where the client is required to
specify a destination address, the "c=" field SHOULD be set to a
null value. For addresses of type "IP4", this value MUST be
"0.0.0.0", and for type "IP6", this value MUST be "0:0:0:0:0:0:0:0"
(can also be written as "::"), i.e., the unspecified address
according to RFC 4291 <xref target="RFC4291"/>.</t>
</section>
<section anchor="sec_sdp-mtag" title="Message Body Tag">
<t>The optional "a=mtag" attribute identifies a version of the
session description. It is opaque to the client. SETUP requests may
include this identifier in the If-Match field (see <xref
target="sec_If-Match"/>) to only allow session establishment if this
attribute value still corresponds to that of the current
description. The attribute value is opaque and may contain any
character allowed within SDP attribute values.</t>
<t>ABNF for the attribute is defined in <xref
target="sec_sdp-syntax"/>.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[ a=mtag:"158bb3e7c7fd62ce67f12b533f06b83a"]]></artwork>
</figure>
<t><list style="hanging">
<t>One could argue that the "o=" field provides identical
functionality. However, it does so in a manner that would put
constraints on servers that need to support multiple session
description types other than SDP for the same piece of media
content.</t>
</list></t>
</section>
</section>
<section anchor="sdp_no_aggr_control"
title="Aggregate Control Not Available">
<t>If a presentation does not support aggregate control no session
level "a=control:" attribute is specified. For a SDP with multiple
media sections specified, each section will have its own control URI
specified via the "a=control:" attribute.</t>
<t>Example:</t>
<figure>
<artwork><![CDATA[v=0
o=- 2890844256 2890842807 IN IP4 192.0.2.56
s=I came from a web page
e=adm@example.com
c=IN IP4 0.0.0.0
t=0 0
m=video 8002 RTP/AVP 31
a=control:rtsp://audio.example.com/movie.aud
m=audio 8004 RTP/AVP 3
a=control:rtsp://video.example.com/movie.vid]]></artwork>
</figure>
<t>Note that the position of the control URI in the description
implies that the client establishes separate RTSP control sessions to
the servers audio.example.com and video.example.com.</t>
<t>It is recommended that an SDP file contains the complete media
initialization information even if it is delivered to the media client
through non-RTSP means. This is necessary as there is no mechanism to
indicate that the client should request more detailed media stream
information via DESCRIBE.</t>
</section>
<section anchor="sdp_aggr_control" title="Aggregate Control Available">
<t>In this scenario, the server has multiple streams that can be
controlled as a whole. In this case, there are both a media-level
"a=control:" attributes, which are used to specify the stream URIs,
and a session-level "a=control:" attribute which is used as the
Request-URI for aggregate control. If the media-level URI is relative,
it is resolved to absolute URIs according to <xref
target="sec_sdp-control-url"/> above.</t>
<t>Example: <figure>
<artwork><![CDATA[C->M: DESCRIBE rtsp://example.com/movie RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Date: Thu, 23 Jan 1997 15:35:06 GMT
Expires: Thu, 23 Jan 1997 16:35:06 GMT
Content-Type: application/sdp
Content-Base: rtsp://example.com/movie/
Content-Length: 227
v=0
o=- 2890844256 2890842807 IN IP4 192.0.2.211
s=I contain
i=<more info>
e=adm@example.com
c=IN IP4 0.0.0.0
a=control:*
t=0 0
m=video 8002 RTP/AVP 31
a=control:trackID=1
m=audio 8004 RTP/AVP 3
a=control:trackID=2
]]></artwork>
</figure></t>
<t>In this example, the client is recommended to establish a single
RTSP session to the server, and uses the URIs
rtsp://example.com/movie/trackID=1 and
rtsp://example.com/movie/trackID=2 to set up the video and audio
streams, respectively. The URI rtsp://example.com/movie/, which is
resolved from the "*", controls the whole presentation (movie).</t>
<t>A client is not required to issue SETUP requests for all streams
within an aggregate object. Servers should allow the client to ask for
only a subset of the streams.</t>
</section>
<section anchor="sec_sdp_m_grouping"
title="Grouping of Media Lines in SDP">
<t>For some types of media it is desirable to express a relationship
between various media components, for instance, for lip
synchronization or Scalable Video Codec (SVC) <xref
target="RFC5583"/>. This relationship is expressed on the SDP level by
grouping of media lines, as described in <xref target="RFC5888"/> and
can be exposed to RTSP.</t>
<t>For RTSP it is mainly important to know how to handle grouped
medias received by means of SDP, i.e., if the media are under
aggregate control (see <xref target="sdp_aggr_control"/>) or if
aggregate control is not available (see <xref
target="sdp_no_aggr_control"/>).</t>
<t>It is RECOMMENDED that grouped medias are handled by aggregate
control, to give the client the ability to control either the whole
presentation or single medias.</t>
</section>
<section title="RTSP external SDP delivery">
<t>There are some considerations that need to be made when the session
description is delivered to the client outside of RTSP, for example
via HTTP or email.</t>
<t>First of all, the SDP needs to contain absolute URIs, since
relative will in most cases not work as the delivery will not
correctly forward the base URI.</t>
<t>The writing of the SDP session availability information, i.e., "t="
and "r=", needs to be carefully considered. When the SDP is fetched by
the DESCRIBE method, the probability that it is valid is very high.
However, the same is much less certain for SDPs distributed using
other methods. Therefore the publisher of the SDP should take care to
follow the recommendations about availability in the SDP specification
<xref target="RFC4566"/> in Section 4.2.</t>
</section>
</section>
<section anchor="sec_usecases" title="RTSP Use Cases">
<t>This Appendix describes the most important and considered use cases
for RTSP. They are listed in descending order of importance in regards
to ensuring that all necessary functionality is present. This
specification only fully supports usage of the two first. Also in these
first two cases, there are special cases or exceptions that are not
supported without extensions, e.g., the redirection of media delivery to
another address than the controlling agent's (client's).</t>
<section anchor="sec_usecases-on-demand"
title="On-demand Playback of Stored Content">
<t>An RTSP capable server stores content suitable for being streamed
to a client. A client desiring playback of any of the stored content
uses RTSP to set up the media transport required to deliver the
desired content. RTSP is then used to initiate, halt and manipulate
the actual transmission (playout) of the content. RTSP is also
required to provide necessary description and synchronization
information for the content.</t>
<t>The above high level description can be broken down into a number
of functions that RTSP needs to be capable of. <list hangIndent="6"
style="hanging">
<t hangText="Presentation Description:">Provide initialization
information about the presentation (content); for example, which
media codecs are needed for the content. Other information that is
important includes the number of media streams the presentation
contains, the transport protocols used for the media streams, and
identifiers for these media streams. This information is required
before setup of the content is possible and to determine if the
client is even capable of using the content. <vspace
blankLines="1"/> This information need not be sent using RTSP;
other external protocols can be used to transmit the transport
presentation descriptions. Two good examples are the use of HTTP
<xref target="RFC2616"/> or email to fetch or receive presentation
descriptions like SDP <xref target="RFC4566"/></t>
<t hangText="Setup:">Set up some or all of the media streams in a
presentation. The setup itself consists of selecting the protocol
for media transport and the necessary parameters for the protocol,
like addresses and ports.</t>
<t hangText="Control of Transmission:">After the necessary media
streams have been established the client can request the server to
start transmitting the content. The client must be allowed to
start or stop the transmission of the content at arbitrary times.
The client must also be able to start the transmission at any
point in the timeline of the presentation.</t>
<t hangText="Synchronization:">For media transport protocols like
RTP <xref target="RFC3550"/> it might be beneficial to carry
synchronization information within RTSP. This may be due to either
the lack of inter-media synchronization within the protocol
itself, or the potential delay before the synchronization is
established (which is the case for RTP when using RTCP).</t>
<t hangText="Termination:">Terminate the established contexts.</t>
</list> For this use case there are a number of assumptions about
how it works. These are: <list hangIndent="6" style="hanging">
<t hangText="On-Demand content:">The content is stored at the
server and can be accessed at any time during a time period when
it is intended to be available.</t>
<t hangText="Independent sessions:">A server is capable of serving
a number of clients simultaneously, including from the same piece
of content at different points in that presentations
time-line.</t>
<t hangText="Unicast Transport:">Content for each individual
client is transmitted to them using unicast traffic.</t>
</list> It is also possible to redirect the media traffic to a
different destination than that of the agent controlling the traffic.
However, allowing this without appropriate mechanisms for checking
that the destination approves of this allows for distributed denial of
service attacks (DDoS).</t>
</section>
<section title="Unicast Distribution of Live Content">
<t>This use case is similar to the above on-demand content case (see
<xref target="sec_usecases-on-demand"/>) the difference is the nature
of the content itself. Live content is continuously distributed as it
becomes available from a source; i.e., the main difference from
on-demand is that one starts distributing content before the end of it
has become available to the server.</t>
<t>In many cases the consumer of live content is only interested in
consuming what actually happens "now"; i.e., very similar to broadcast
TV. However, in this case it is assumed that there exists no broadcast
or multicast channel to the users, and instead the server functions as
a distribution node, sending the same content to multiple receivers,
using unicast traffic between server and client. This unicast traffic
and the transport parameters are individually negotiated for each
receiving client.</t>
<t>Another aspect of live content is that it often has a very limited
time of availability, as it is only available for the duration of the
event the content covers. An example of such a live content could be a
music concert which lasts 2 hour and starts at a predetermined time.
Thus there is a need to announce when and for how long the live
content is available.</t>
<t>In some cases, the server providing live content may be saving some
or all of the content to allow clients to pause the stream and resume
it from the paused point, or to "rewind" and play continuously from a
point earlier than the live point. Hence, this use case does not
necessarily exclude playing from other than the live point of the
stream, playing with scales other than 1.0, etc.</t>
</section>
<section title="On-demand Playback using Multicast">
<t>It is possible to use RTSP to request that media be delivered to a
multicast group. The entity setting up the session (the controller)
will then control when and what media is delivered to the group. This
use case has some potential for denial of service attacks by flooding
a multicast group. Therefore, a mechanism is needed to indicate that
the group actually accepts the traffic from the RTSP server.</t>
<t>An open issue in this use case is how one ensures that all
receivers listening to the multicast or broadcast receives the session
presentation configuring the receivers. This specification has to rely
on an external solution to solve this issue.</t>
</section>
<section title="Inviting an RTSP server into a conference">
<t>If one has an established conference or group session, it is
possible to have an RTSP server distribute media to the whole group.
Transmission to the group is simplest when controlled by a single
participant or leader of the conference. Shared control might be
possible, but would require further investigation and possibly
extensions.</t>
<t>This use case assumes that there exists either multicast or a
conference focus that redistribute media to all participants.</t>
<t>This use case is intended to be able to handle the following
scenario: A conference leader or participant (hereafter called the
controller) has some pre-stored content on an RTSP server that he
wants to share with the group. The controller sets up an RTSP session
at the streaming server for this content and retrieves the session
description for the content. The destination for the media content is
set to the shared multicast group or conference focus. When desired by
the controller, he/she can start and stop the transmission of the
media to the conference group.</t>
<t>There are several issues with this use case that are not solved by
this core specification for RTSP: <list hangIndent="6" style="hanging">
<t hangText="Denial of service:">To avoid an RTSP server from
being an unknowing participant in a denial of service attack the
server needs to be able to verify the destination's acceptance of
the media. Such a mechanism to verify the approval of received
media does not yet exist; instead, only policies can be used,
which can be made to work in controlled environments.</t>
<t
hangText="Distributing the presentation description to all participants in the group:">To
enable a media receiver to correctly decode the content the media
configuration information needs to be distributed reliably to all
participants. This will most likely require support from an
external protocol.</t>
<t hangText="Passing control of the session:">If it is desired to
pass control of the RTSP session between the participants, some
support will be required by an external protocol to exchange state
information and possibly floor control of who is controlling the
RTSP session.</t>
</list></t>
</section>
<section title="Live Content using Multicast">
<t>This use case in its simplest form does not require any use of RTSP
at all; this is what multicast conferences being announced with <xref
target="RFC2974">SAP</xref> and SDP are intended to handle. However,
in use cases where more advanced features like access control to the
multicast session are desired, RTSP could be used for session
establishment.</t>
<t>A client desiring to join a live multicasted media session with
cryptographic (encryption) access control could use RTSP in the
following way. The source of the session announces the session and
gives all interested an RTSP URI. The client connects to the server
and requests the presentation description, allowing configuration for
reception of the media. In this step it is possible for the client to
use secured transport and any desired level of authentication; for
example, for billing or access control. An RTSP link also allows for
load balancing between multiple servers.</t>
<t>If these were the only goals, they could be achieved by simply
using HTTP. However, for cases where the sender likes to keep track of
each individual receiver of a session, and possibly use the session as
a side channel for distributing key-updates or other information on a
per-receiver basis, and the full set of receivers is not known prior
to the session start, the state establishment that RTSP provides can
be beneficial. In this case a client would establish an RTSP session
for this multicast group with the RTSP server. The RTSP server will
not transmit any media, but instead will point to the multicast group.
The client and server will be able to keep the session alive for as
long as the receiver participates in the session thus enabling, for
example, the server to push updates to the client.</t>
<t>This use case will most likely not be able to be implemented
without some extensions to the server-to-client push mechanism. Here
the PLAY_NOTIFY method (see <xref target="sec_PLAY_NOTIFY"/>) with a
suitable extension could provide clear benefits.</t>
</section>
</section>
<!-- title="Use of SDP for RTSP Session Descriptions" -->
<section anchor="sec_text-parameters" title="Text format for Parameters">
<t>A resource of type "text/parameters" consists of either 1) a list of
parameters (for a query) or 2) a list of parameters and associated
values (for an response or setting of the parameter). Each entry of the
list is a single line of text. Parameters are separated from values by a
colon. The parameter name MUST only use US-ASCII visible characters
while the values are UTF-8 text strings. The media type registration
form is in <xref target="sec_iana_textpar"/>.</t>
<t>There is a potential interoperability issue for this format. It was
named in RFC 2326 but never defined, even if used in examples that hint
at the syntax. This format matches the purpose and its syntax supports
the examples provided. However, it goes further by allowing UTF-8 in the
value part, thus usage of UTF-8 strings may not be supported. However,
as individual parameters are not defined, the using application anyway
needs to have out-of-band agreement or using feature-tag to determine if
the end-point supports the parameters.</t>
<t>The <xref target="RFC5234">ABNF</xref> grammar for "text/parameters"
content is:</t>
<t><figure>
<artwork><![CDATA[file = *((parameter / parameter-value) CRLF)
parameter = 1*visible-except-colon
parameter-value = parameter *WSP ":" value
visible-except-colon = %x21-39 / %x3B-7E ; VCHAR - ":"
value = *(TEXT-UTF8char / WSP)
TEXT-UTF8char = %x21-7E / UTF8-NONASCII
UTF8-NONASCII = %xC0-DF 1UTF8-CONT
/ %xE0-EF 2UTF8-CONT
/ %xF0-F7 3UTF8-CONT
/ %xF8-FB 4UTF8-CONT
/ %xFC-FD 5UTF8-CONT
UTF8-CONT = %x80-BF
WSP = <See RFC 5234> ; Space or HTAB
VCHAR = <See RFC 5234>
CRLF = <See RFC 5234>]]></artwork>
</figure></t>
<t/>
</section>
<section anchor="sec_unreliable"
title="Requirements for Unreliable Transport of RTSP">
<t>This appendix provides guidance for those who want to implement RTSP
messages over unreliable transports as has been defined in <xref
target="RFC2326">RTSP 1.0</xref>. RFC 2326 defined the "rtspu" URI
scheme and provided some basic information for the transport of RTSP
messages over UDP. The information is being provided here as there has
been at at least one commercial implementation and compatibility with
that should be maintained.</t>
<t>The following points should be considered for an interoperable
implementation:</t>
<t><list hangIndent="3" style="symbols">
<t>Requests shall be acknowledged by the receiver. If there is no
acknowledgement, the sender may resend the same message after a
timeout of one round-trip time (RTT). Any retransmissions due to
lack of acknowledgement must carry the same sequence number as the
original request.</t>
<t>The round-trip time can be estimated as in TCP (RFC 6298) <xref
target="RFC6298"/>, with an initial round-trip value of 500 ms. An
implementation may cache the last RTT measurement as the initial
value for future connections.</t>
<t>The Timestamp header (<xref target="sec_Timestamp"/>) is used to
avoid <xref target="Stevens98">the retransmission ambiguity
problem</xref>.</t>
<t>The registered default port for RTSP over UDP for the server is
554.</t>
<t>RTSP messages can be carried over any lower-layer transport
protocol that is 8-bit clean.</t>
<t>RTSP messages are vulnerable to bit errors and should not be
subjected to them.</t>
<t>Source authentication, or at least validation that RTSP messages
comes from the same entity becomes extremely important, as session
hijacking may be substantially easier for RTSP message transport
using an unreliable protocol like UDP than for TCP.</t>
</list></t>
<t>There are two RTSP headers that are primarily intended for being used
by the unreliable handling of RTSP messages and which will be
maintained: <list hangIndent="3" style="symbols">
<t>CSeq: See <xref target="sec_CSeq"/>. It should be noted that the
CSeq header is also required to match requests and responses
independent whether a reliable or unreliable transport is used.</t>
<t>Timestamp: See <xref target="sec_Timestamp"/></t>
</list></t>
</section>
<!-- title="Requirements for Unreliable Transport of RTSP"> -->
<section anchor="sec_backwards"
title="Backwards Compatibility Considerations">
<t>This section contains notes on issues about backwards compatibility
with clients or servers being implemented according to RFC 2326 <xref
target="RFC2326"/>. Note that there exists no requirement to implement
RTSP 1.0; in fact we recommend against it as it is difficult to do in an
interoperable way.</t>
<t>A server implementing RTSP/2.0 MUST include an RTSP-Version of
RTSP/2.0 in all responses to requests containing RTSP-Version RTSP/2.0.
If a server receives an RTSP/1.0 request, it MAY respond with an
RTSP/1.0 response if it chooses to support RFC 2326. If the server
chooses not to support RFC 2326, it MUST respond with a 505 (RTSP
Version not supported) status code. A server MUST NOT respond to an
RTSP-Version RTSP/1.0 request with an RTSP-Version RTSP/2.0
response.</t>
<t>Clients implementing RTSP/2.0 MAY use an OPTIONS request with a
RTSP-Version of 2.0 to determine whether a server supports RTSP/2.0. If
the server responds with either an RTSP-Version of 1.0 or a status code
of 505 (RTSP Version not supported), the client will have to use
RTSP/1.0 requests if it chooses to support RFC 2326.</t>
<section anchor="sec_back-play" title="Play Request in Play State">
<t>The behavior in the server when a Play is received in Play state
has changed (<xref target="sec_PLAY"/>). In RFC 2326, the new PLAY
request would be queued until the current Play completed. Any new PLAY
request now takes effect immediately replacing the previous
request.</t>
</section>
<section anchor="sec_back-persistent-connection"
title="Using Persistent Connections">
<t>Some server implementations of RFC 2326 maintain a one-to-one
relationship between a connection and an RTSP session. Such
implementations require clients to use a persistent connection to
communicate with the server and when a client closes its connection,
the server may remove the RTSP session. This is worth noting if a RTSP
2.0 client also supporting 1.0 connects to a 1.0 server.</t>
</section>
</section>
<!-- title="Backwards Compatibility Considerations" -->
<!--
<section anchor="sec_open" title="Open Issues">
<t>Open issues are filed and tracked in the bug and feature trackers at
http://rtspspec.sourceforge.net. Open issues are discussed on MMUSIC
list (mmusic@ietf.org).</t>
<t>Note to RFC-editor: Please remove this section before publication of
this document as an RFC.</t>
</section>
-->
<!-- title="Open Issues" -->
<section anchor="sec_changes" title="Changes">
<t>This appendix briefly lists the differences between <xref
target="RFC2326">RTSP 1.0</xref> and RTSP 2.0 for an informational
purpose. For implementers of RTSP 2.0 it is recommended to read
carefully through this memo and not to rely on the list of changes below
to adapt from RTSP 1.0 to RTSP 2.0, as RTSP 2.0 is not intended to be
backwards compatible with <xref target="RFC2326">RTSP 1.0</xref> other
than the version negotiation mechanism.</t>
<section title="Brief Overview">
<t>The following protocol elements were removed in RTSP 2.0 compared
to RTSP 1.0:<list style="symbols">
<t>there is no section on minimal implementation anymore, but more
the definition of RTSP 2.0 core;</t>
<t>the RECORD and ANNOUNCE methods and all related functionality
(including 201 (Created) and 250 (Low On Storage Space) status
codes);</t>
<t>the use of UDP for RTSP message transport was removed due to
missing interest and to broken specification;</t>
<t>the use of PLAY method for keep-alive in Play state.</t>
</list></t>
<t>The following protocol elements were added or changed in RTSP 2.0
compared to RTSP 1.0:<list style="symbols">
<t>RTSP session TEARDOWN from the server to the client;</t>
<t>IPv6 support;</t>
<t>extended IANA registries (e.g., transport headers parameters,
transport-protocol, profile, lower-transport, and mode);</t>
<t>request pipelining for quick session start-up;</t>
<t>fully reworked state-machine;</t>
<t>RTSP messages now use URIs rather then URLs;</t>
<t>incorporated much of related HTTP text (<xref
target="RFC2616"/>) in this memo, compared to just referencing the
sections in HTTP, to avoid ambiguities;</t>
<t>the REDIRECT method was expanded and diversified for different
situations;</t>
<t>Includes a new section about how to setup different media
transport alternatives and their profiles, and lower layer
protocols. This caused the appendix on RTP interaction to be moved
there instead of being in the part which describes RTP. The
section also includes guidelines what to consider when writing
usage guidelines for new protocols and profiles;</t>
<t>Added an asynchronous notification method PLAY_NOTIFY. This
method is used by the RTSP server to asynchronously notify clients
about session changes while in Play state. To a limited extent
this is comparable with some implementations of ANNOUNCE in RTSP
1.0 not intended for Recording.</t>
</list></t>
</section>
<section title="Detailed List of Changes">
<t>Compared to RTSP 1.0 (RFC 2326), the below changes has been made
when defining RTSP 2.0. Note that this list does not reflect minor
changes in wording or correction of typographical errors. <list
hangIndent="3" style="symbols">
<t>The section on minimal implementation was deleted without
substitution.</t>
<t>The Transport header has been changed in the following way:
<list style="symbols">
<t>The ABNF has been changed to define that extensions are
possible, and that unknown parameters result in that servers
ignore the transport specification.</t>
<t>To prevent backwards compatibility issues, any extension or
new parameter requires the usage of a feature-tag combined
with the Require header.</t>
<t>Syntax unclarities with the Mode parameter have been
resolved.</t>
<t>Syntax error with ";" for multicast and unicast has been
resolved.</t>
<t>Two new addressing parameters have been defined, src_addr
and dest_addr. These replace the parameters "port",
"client_port", "server_port", "destination", "source".</t>
<t>Support for IPv6 explicit addresses in all address fields
has been included.</t>
<t>To handle URI definitions that contain ";" or "," a quoted
URI format has been introduced and is required.</t>
<t>Defined IANA registries for the transport headers
parameters, transport-protocol, profile, lower-transport, and
mode.</t>
<t>The transport headers interleaved parameter's text was made
more strict and uses formal requirements levels. It was also
clarified that the interleaved channels are symmetric and that
it is the server that sets the channel numbers.</t>
<t>It has been clarified that the client can't request of the
server to use a certain RTP SSRC, using a request with the
transport parameter SSRC.</t>
<t>Syntax definition for SSRC has been clarified to require
8HEX. It has also been extended to allow multiple values for
clients supporting this version.</t>
<t>Clarified the text on the transport headers "dest_addr"
parameters regarding what security precautions the server is
required to perform.</t>
</list></t>
<t>The Range formats has been changed in the following way: <list
style="symbols">
<t>The NPT format has been given an initial NPT identifier
that must now be used.</t>
<t>All formats now support initial open ended formats of type
"npt=-10" and also format only "Range: smpte" ranges for usage
with GET_PARAMETER requests.</t>
</list></t>
<t>RTSP message handling has been changed in the following way:
<list style="symbols">
<t>RTSP messages now use URIs rather then URLs.</t>
<t>It has been clarified that a 4xx message due to missing
CSeq header shall be returned without a CSeq header.</t>
<t>The 300 (Multiple Choices) response code has been
removed.</t>
<t>Rules for how to handle timing out RTSP messages has been
added.</t>
<t>Extended Pipelining rules allowing for quick session
startup.</t>
<t>Sequence numbering and proxy handling of sequence number
defined, including case when response arrive out of order.</t>
</list></t>
<t>The HTTP references have been updated to RFC 2616 and RFC 2617.
Most of the text has been copied and then altered to fit RTSP into
this specification. Public, and the Content-Base header has also
been imported from RFC 2068 so that they are defined in the RTSP
specification. Known effects on RTSP due to HTTP clarifications:
<list style="symbols">
<t>Content-Encoding header can include encoding of type
"identity".</t>
</list></t>
<t>The state machine section has been completely rewritten. It now
includes more details and is also more clear about the model
used.</t>
<t>An IANA section has been included which contains a number of
registries and their rules. This will allow us to use IANA to keep
track of RTSP extensions.</t>
<t>The transport of RTSP messages has seen the following changes:
<list style="symbols">
<t>The use of UDP for RTSP message transport has been
deprecated due to missing interest and to broken
specification.</t>
<t>The rules for how TCP connections are to be handled has
been clarified. Now it is made clear that servers should not
close the TCP connection unless they have been unused for
significant time.</t>
<t>Strong recommendations why server and clients should use
persistent connections have also been added.</t>
<t>There is now a requirement on the servers to handle
non-persistent connections as this provides fault
tolerance.</t>
<t>Added wording on the usage of Connection:Close for
RTSP.</t>
<t>Specified usage of TLS for RTSP messages, including a
scheme to approve a proxy's TLS connection to the next
hop.</t>
</list></t>
<t>The following header related changes have been made: <list
style="symbols">
<t>Accept-Ranges response-header is added. This header
clarifies which range formats that can be used for a
resource.</t>
<t>Fixed the missing definitions for the Cache-Control header.
Also added to the syntax definition the missing delta-seconds
for max-stale and min-fresh parameters.</t>
<t>Put requirement on CSeq header that the value is increased
by one for each new RTSP request. A Recommendation to start at
0 has also been added.</t>
<t>Added requirement that the Date header must be used for all
messages with message body and the Server should always
include it.</t>
<t>Removed possibility of using Range header with Scale header
to indicate when it is to be activated, since it can't work as
defined. Also added rule that lack of Scale header in response
indicates lack of support for the header. Feature-tags for
scaled playback has been defined.</t>
<t>The Speed header must now be responded to indicate support
and the actual speed going to be used. A feature-tag is
defined. Notes on congestion control were also added.</t>
<t>The Supported header was borrowed from <xref
target="RFC3261">SIP</xref> to help with the feature
negotiation in RTSP.</t>
<t>Clarified that the Timestamp header can be used to resolve
retransmission ambiguities.</t>
<t>The Session header text has been expanded with an
explanation on keep-alive and which methods to use.
SET_PARAMETER is now recommended to use if only keep-alive
within RTSP is desired.</t>
<t>It has been clarified how the Range header formats are used
to indicate pause points in the PAUSE response.</t>
<t>Clarified that RTP-Info URIs that are relative, use the
Request-URI as base URI. Also clarified that the used URI must
be the one that was used in the SETUP request. The URIs are
now also required to be quoted. The header also expresses the
SSRC for the provided RTP timestamp and sequence number
values.</t>
<t>Added text that requires the Range to always be present in
PLAY responses. Clarified what should be sent in case of live
streams.</t>
<t>The headers table has been updated using a structure
borrowed from SIP. Those tables convey much more information
and should provide a good overview of the available
headers.</t>
<t>It has been clarified that any message with a message body
is required to have a Content-Length header. This was the case
in RFC 2326, but could be misinterpreted.</t>
<t>ETag has changed name to MTag.</t>
<t>To resolve functionality around MTag. The MTag and
If-None-Match header have been added from HTTP with necessary
clarification in regards to RTSP operation.</t>
<t>Imported the Public header from HTTP RFC 2068 <xref
target="RFC2068"/> since it has been removed from HTTP due to
lack of use. Public is used quite frequently in RTSP.</t>
<t>Clarified rules for populating the Public header so that it
is an intersection of the capabilities of all the RTSP agents
in a chain.</t>
<t>Added the Media-Range header for listing the current
availability of the media range.</t>
<t>Added the Notify-Reason header for giving the reason when
sending PLAY_NOTIFY requests.</t>
<t>A new header Seek-Style has been defined to direct and
inform how any seek operation should/have been performed.</t>
</list></t>
<t>The Protocol Syntax has been changed in the following way:
<list style="symbols">
<t>All ABNF definitions are updated according to the rules
defined in RFC 5234 <xref target="RFC5234"/> and have been
gathered in a separate <xref target="sec_syntax"/>.</t>
<t>The ABNF for the User-Agent and Server headers have been
corrected.</t>
<t>Some definitions in the introduction regarding the RTSP
session have been changed.</t>
<t>The protocol has been made fully IPv6 capable.</t>
<!--
<t>Added a fragment part to the RTSP URI. This seemed to be
indicated by the note below the definition, however, it was
not part of the ABNF.</t>
-->
<t>The CHAR rule has been changed to exclude NULL.</t>
</list></t>
<t>The Status codes have been changed in the following way: <list
style="symbols">
<t>The use of status code 303 "See Other" has been deprecated
as it does not make sense to use in RTSP.</t>
<t>When sending response 451 and 458 the response body should
contain the offending parameters.</t>
<t>Clarification on when a 3rr redirect status code can be
received has been added. This includes receiving 3rr as a
result of a request within a established session. This
provides clarification to a previous unspecified behavior.</t>
<t>Removed the 201 (Created) and 250 (Low On Storage Space)
status codes as they are only relevant to recording, which is
deprecated.</t>
<t>Several new Status codes have been defined: 464 "Data
Transport Not Ready Yet", 465 "Notification Reason Unknown",
470 "Connection Authorization Required", 471 "Connection
Credentials not accepted", 472 "Failure to establish secure
connection".</t>
</list></t>
<t>The following functionality has been deprecated from the
protocol: <list style="symbols">
<t>The use of Queued Play.</t>
<t>The use of PLAY method for keep-alive in Play state.</t>
<t>The RECORD and ANNOUNCE methods and all related
functionality. Some of the syntax has been removed.</t>
<t>The possibility to use timed execution of methods with the
time parameter in the Range header.</t>
<t>The description on how rtspu works is not part of the core
specification and will require external description. Only that
it exists is defined here and some requirements for the
transport is provided.</t>
</list></t>
<t>The following changes have been made in relation to methods:
<list style="symbols">
<t>The OPTIONS method has been clarified with regards to the
use of the Public and Allow headers.</t>
<t>Added text clarifying the usage of SET_PARAMETER for
keep-alive and usage without any body.</t>
<t>PLAY method is now allowed to be pipelined with the
pipelining of one or more SETUP requests following the initial
that generates the session for aggregated control.</t>
<t>REDIRECT has been expanded and diversified for different
situations.</t>
<t>Added a new method PLAY_NOTIFY. This method is used by the
RTSP server to asynchronously notify clients about session
changes.</t>
</list></t>
<t>Wrote a new section about how to setup different media
transport alternatives and their profiles, and lower layer
protocols. This caused the appendix on RTP interaction to be moved
there instead of being in the part which describes RTP. The
section also includes guidelines what to consider when writing
usage guidelines for new protocols and profiles.</t>
<t>Setup and usage of independent TCP connections for transport of
RTP has been specified.</t>
<t>Added a new section describing the available mechanisms to
determine if functionality is supported, called "Capability
Handling". Renamed option-tags to feature-tags.</t>
<t>Added a contributors section with people who have contributed
actual text to the specification.</t>
<t>Added a section Use Cases that describes the major use cases
for RTSP.</t>
<t>Clarified the usage of a=range and how to indicate live content
that are not seekable with this header.</t>
<t>Text specifying the special behavior of PLAY for live
content.</t>
<t>Security features of RTSP have been clarified:<list
style="symbols">
<t>HTTP based authorization has been clarified requring both
Basic and DIGEST support</t>
<t>TLS support mandated</t>
<t>IF one implements RTP then SRTP and defined MIKEY based
key-exchange must be supported</t>
<t>Various minor mitigations discussed or resulted in protocol
changes.</t>
</list></t>
</list></t>
</section>
</section>
<!-- title="Changes" -->
<section anchor="Acknowledgements" title="Acknowledgements">
<t>This memorandum defines RTSP version 2.0 which is a revision of the
Proposed Standard RTSP version 1.0 which is defined in <xref
target="RFC2326"/>. The authors of RFC 2326 are Henning Schulzrinne,
Anup Rao, and Robert Lanphier.</t>
<t>Both RTSP version 1.0 and RTSP version 2.0 borrow format and
descriptions from HTTP/1.1.</t>
<t>Robert Sparks and especially Elwyn Davies provided very valuable and
detailed reviews in the IETF last call that greately improved the
document and resolved many issues, especially regarding consistency.</t>
<t>This document has benefited greatly from the comments of all those
participating in the MMUSIC-WG. In addition to those already mentioned,
the following individuals have contributed to this specification:</t>
<t>Rahul Agarwal, Claudio Allocchio, Jeff Ayars, Milko Boic, Torsten
Braun, Brent Browning, Bruce Butterfield, Steve Casner, Maureen Chesire,
Jinhang Choi, Francisco Cortes, Elwyn Davies, Kelly Djahandari, Martin
Dunsmuir, Stephen Farrell, Ross Finlayson, Eric Fleischman, Jay Geagan,
Andy Grignon, Christian Groves, V. Guruprasad, Peter Haight, Mark
Handley, Brad Hefta-Gaub, Volker Hilt, John K. Ho, Patrick Hoffman, Go
Hori, Philipp Hoschka, Anne Jones, Ingemar Johansson, Jae-Hwan Kim,
Anders Klemets, Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F.
Llach, Chris Lonvick, Xavier Marjou, Thomas Marshall, Rob McCool, Martti
Mela, David Oran, Joerg Ott, Joe Pallas, Maria Papadopouli, Sujal Patel,
Ema Patki, Alagu Periyannan, Colin Perkins, Pekka Pessi, Igor Plotnikov,
Peter Saint-Andre, Holger Schmidt, Jonathan Sergent, Pinaki Shah, David
Singer, Lior Sion, Jeff Smith, Alexander Sokolsky, Dale Stammen, John
Francis Stracke, Geetha Srikantan, Scott Taylor, David Walker, Stephan
Wenger, Dale R. Worley, and Byungjo Yoon , and especially to Flemming
Andreasen.</t>
<section anchor="sec_contributors" title="Contributors">
<t>The following people have made written contributions that were
included in the specification: <list hangIndent="3" style="symbols">
<t>Tom Marshall contributed text on the usage of 3rr status
codes.</t>
<t>Thomas Zheng contributed text on the usage of the Range in PLAY
responses and proposed an earlier version of the PLAY_NOTIFY
method.</t>
<t>Sean Sheedy contributed text on the timeout behavior of RTSP
messages and connections, the 463 status code, and proposed an
earlier version of the PLAY_NOTIFY method.</t>
<t>Greg Sherwood proposed an earlier version of the PLAY_NOTIFY
method.</t>
<t>Fredrik Lindholm contributed text about the RTSP security
framework.</t>
<t>John Lazzaro contributed the text for RTP over Independent
TCP.</t>
<t>Aravind Narasimhan contributed by rewriting <xref
target="sec_mediatran">Media Transport Alternatives</xref> and
editorial improvements on a number of places in the
specification.</t>
<t>Torbjorn Einarsson has done some editorial improvements of the
text.</t>
</list></t>
</section>
<!-- title="Contributors" -->
</section>
<!-- title="Acknowledgements" -->
<section anchor="RFCEditorConsideration" title="RFC Editor Consideration">
<t>Please replace RFC XXXX with the RFC number this specification
receives.</t>
</section>
<!-- title="RFC Editor Consideration" -->
</back>
</rfc>
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