One document matched: draft-ietf-mmusic-rtsp-03.txt
Differences from draft-ietf-mmusic-rtsp-02.txt
Real Time Streaming Protocol (RTSP)
STATUS OF THIS MEMO
This document is an Internet-Draft. Internet-Drafts are working
documents of the Internet Engineering Task Force (IETF), its areas,
and its working groups. Note that other groups may also distribute
working documents as Internet-Drafts.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as ``work in progress''.
To learn the current status of any Internet-Draft, please check the
``1id-abstracts.txt'' listing contained in the Internet-Drafts Shadow
Directories on ftp.is.co.za (Africa), nic.nordu.net (Europe),
munnari.oz.au (Pacific Rim), ds.internic.net (US East Coast), or
ftp.isi.edu (US West Coast).
Distribution of this document is unlimited.
Abstract:
The Real Time Streaming Protocol, or RTSP, is an application-level
protocol for control over 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 delivery mechanisms based upon RTP (RFC
1889).
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Contents
* 1 Introduction
+ 1.1 Purpose
+ 1.2 Requirements
+ 1.3 Terminology
+ 1.4 Protocol Properties
+ 1.5 Extending RTSP
+ 1.6 Overall Operation
+ 1.7 RTSP States
+ 1.8 Relationship with Other Protocols
* 2 Notational Conventions
* 3 Protocol Parameters
+ 3.1 RTSP Version
+ 3.2 RTSP URL
+ 3.3 Conference Identifiers
+ 3.4 Session Identifiers
+ 3.5 SMPTE Relative Timestamps
+ 3.6 Normal Play Time
+ 3.7 Absolute Time
* 4 RTSP Message
+ 4.1 Message Types
+ 4.2 Message Headers
+ 4.3 Message Body
+ 4.4 Message Length
* 5 General Header Fields
* 6 Request
+ 6.1 Request Line
+ 6.2 Request Header Fields
* 7 Response
+ 7.1 Status-Line
o 7.1.1 Status Code and Reason Phrase
o 7.1.2 Response Header Fields
* 8 Entity
+ 8.1 Entity Header Fields
+ 8.2 Entity Body
* 9 Connections
+ 9.1 Pipelining
+ 9.2 Reliability and Acknowledgements
* 10 Method Definitions
+ 10.1 OPTIONS
+ 10.2 DESCRIBE
+ 10.3 ANNOUNCE
+ 10.4 SETUP
+ 10.5 PLAY
+ 10.6 PAUSE
+ 10.7 TEARDOWN
+ 10.8 GET_PARAMETER
+ 10.9 SET_PARAMETER
+ 10.10 REDIRECT
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+ 10.11 RECORD
+ 10.12 Embedded (Interleaved) Binary Data
* 11 Status Code Definitions
+ 11.1 Redirection 3xx
+ 11.2 Client Error 4xx
o 11.2.1 405 Method Not Allowed
o 11.2.2 451 Parameter Not Understood
o 11.2.3 452 Conference Not Found
o 11.2.4 453 Not Enough Bandwidth
o 11.2.5 45x Session Not Found
o 11.2.6 45x Method Not Valid in This State
o 11.2.7 45x Header Field Not Valid for Resource
o 11.2.8 45x Invalid Range
o 11.2.9 45x Parameter Is Read-Only
o 11.2.10 45x Aggregate operation not allowed
o 11.2.11 45x Only aggregate operation allowed
* 12 Header Field Definitions
+ 12.1 Accept
+ 12.2 Accept-Encoding
+ 12.3 Accept-Language
+ 12.4 Allow
+ 12.5 Authorization
+ 12.6 Bandwidth
+ 12.7 Blocksize
+ 12.8 C-PEP
+ 12.9 C-PEP-Info
+ 12.10 Cache-Control
+ 12.11 Conference
+ 12.12 Connection
+ 12.13 Content-Encoding
+ 12.14 Content-Language
+ 12.15 Content-Length
+ 12.16 Content-Type
+ 12.17 Date
+ 12.18 Expires
+ 12.19 From
+ 12.20 Host
+ 12.21 If-Modified-Since
+ 12.22 Last-Modified
+ 12.23 Location
+ 12.24 PEP
+ 12.25 PEP-Info
+ 12.26 Proxy-Authenticate
+ 12.27 Public
+ 12.28 Range
+ 12.29 Referer
+ 12.30 Retry-After
+ 12.31 Scale
+ 12.32 Speed
+ 12.33 Server
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+ 12.34 Session
+ 12.35 Transport
+ 12.36 Transport-Info
+ 12.37 User-Agent
+ 12.38 Vary
+ 12.39 Via
+ 12.40 WWW-Authenticate
* 13 Caching
* 14 Examples
+ 14.1 Media on Demand (Unicast)
+ 14.2 Streaming of a Container file
+ 14.3 Live Media Presentation Using Multicast
+ 14.4 Playing media into an existing session
+ 14.5 Recording
* 15 Syntax
+ 15.1 Base Syntax
* 16 Security Considerations
* A RTSP Protocol State Machines
+ A.1 Client State Machine
+ A.2 Server State Machine
* B Open Issues
* C Changes
* D Author Addresses
* E Acknowledgements
* References
1 Introduction
1.1 Purpose
The Real-Time Streaming Protocol (RTSP) establishes and controls
either a single or several time-synchronized streams of continuous
media such as audio and video. It does not typically deliver the
continuous streams itself, although interleaving of the continuous
media stream with the control stream is possible (see Section 10.12).
In other words, RTSP acts as a ``network remote control'' for
multimedia servers.
The set of streams to be controlled is defined by a presentation
description. This memorandum does not define a format for a
presentation description.
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There is no notion of an RTSP connection; instead, a server maintains
a session labeled by an identifier. An RTSP session is in no way tied
to a transport-level connection such as a TCP connection. During an
RTSP session, an RTSP client may open and close many reliable
transport connections to the server to issue RTSP requests.
Alternatively, it may use a connectionless transport protocol such as
UDP.
The streams controlled by RTSP may use RTP [1], but the operation of
RTSP does not depend on the transport mechanism used to carry
continuous media.
The protocol is intentionally similar in syntax and operation to
HTTP/1.1, so that extension mechanisms to HTTP can in most cases also
be added to RTSP. However, RTSP differs in a number of important
aspects from HTTP:
* RTSP introduces a number of new methods and has a different
protocol identifier.
* An RTSP server needs to maintain state by default in almost all
cases, as opposed to the stateless nature of HTTP.
* Both an RTSP server and client can issue requests.
* Data is carried out-of-band, by a different protocol. (There is an
exception to this.)
* RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,
consistent with current HTML internationalization efforts [3].
* The Request-URI always contains the absolute URI. Because of
backward compatibility with a historical blunder, HTTP/1.1 carries
only the absolute path in the request and puts the host name in a
separate header field.
This makes ``virtual hosting'' easier, where a single host with one
IP address hosts several document trees.
The protocol supports the following operations:
Retrieval of media from media server:
The client can request a presentation description via HTTP or
some other method. If the presentation is being multicast, the
presentation description contains the multicast addresses and
ports to be used for the continuous media. If the presentation
is to be sent only to the client via unicast, the client
provides the destination for security reasons.
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Invitation of a media server to a conference:
A media server can be ``invited'' to join an existing
conference, either to play back media into the presentation or
to record all or a subset of the media in a presentation. This
mode is useful for distributed teaching applications. Several
parties in the conference may take turns ``pushing the remote
control buttons''.
Addition of media to an existing presentation:
Particularly for live presentations, it is useful if the server
can tell the client about additional media becoming available.
RTSP requests may be handled by proxies, tunnels and caches as in
HTTP/1.1.
1.2 Requirements
The key words ``MUST'', ``MUST NOT'', ``REQUIRED'', ``SHALL'', ``SHALL
NOT'', ``SHOULD'', ``SHOULD NOT'', ``RECOMMENDED'', ``MAY'', and
``OPTIONAL'' in this document are to be interpreted as described in
RFC 2119 [4].
1.3 Terminology
Some of the terminology has been adopted from HTTP/1.1 [5]. Terms not
listed here are defined as in HTTP/1.1.
Conference:
a multiparty, multimedia presentation, where ``multi'' implies
greater than or equal to one.
Client:
The client requests continuous media data from the media
server.
Connection:
A transport layer virtual circuit established between two
programs for the purpose of communication.
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Continuous media:
Data where there is a timing relationship between source and
sink, that is, the sink must reproduce the timing relationshop
that existed at the source. The most common examples of
continuous media are audio and motion video. Continuous media
can be realtime (interactive), where there is a ``tight''
timing relationship between source and sink, or streaming
(playback), where the relationship is less strict.
Participant:
Participants are members of conferences. A participant may be a
machine, e.g., a media record or playback server.
Media server:
The network entity providing playback or recording 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 or a different host as the
web server the presentation is invoked from.
Media parameter:
Parameter specific to a media type that may be changed while
the stream is being played or prior to it.
(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. This is
equivalent to the definition of a DSM-CC stream([18]).
Message:
The basic unit of RTSP communication, consisting of a
structured sequence of octets matching the syntax defined in
Section 15 and transmitted via a connection or a connectionless
protocol.
Presentation:
A set of one or more streams which the server allows the client
to manipulate together. A presentation has a single time axis
for all streams belonging to it. Presentations are defined by
presentation descriptions (see below). A presentation
description contains RTSP URIs that define which streams can be
controlled individually and an RTSP URI to control the whole
presentation. A movie or live concert consisting of one or more
audio and video streams is an example of a presentation.
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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 [6] use the term ``session''
for a live presentation. The presentation description may take
several different formats, including but not limited to the
session description format SDP.
Response:
An RTSP response. If an HTTP response is meant, that is
indicated explicitly.
Request:
An RTSP request. If an HTTP request is meant, that is indicated
explicitly.
RTSP session:
A complete RTSP ``transaction'', e.g., the viewing of a movie.
A session typically consists of a client setting up a transport
mechanism for the continuous media stream (SETUP), starting the
stream with PLAY or RECORD and closing the stream with
TEARDOWN.
1.4 Protocol Properties
RTSP has the following properties:
Extendable:
New methods and parameters can be easily added to RTSP.
Easy to parse:
RTSP can be parsed by standard HTTP or MIME parsers.
Secure:
RTSP re-uses web security mechanisms, either at the transport
level (TLS [7]) or within the protocol itself. All HTTP
authentication mechanisms such as basic [5, Section 11.1] and
digest authentication [8] are directly applicable.
Transport-independent:
RTSP may use either an unreliable datagram protocol (UDP) [9],
a reliable datagram protocol (RDP, not widely used [10]) or a
reliable stream protocol such as TCP [11] as it implements
application-level reliability.
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Multi-server capable:
Each media stream within a presentation can reside on a
different server. The client automatically establishes several
concurrent control sessions with the different media servers.
Media synchronization is performed at the transport level.
Control of recording devices:
The protocol can control both recording and playback devices,
as well as devices that can alternate between the two modes
(``VCR'').
Separation of stream control and conference initiation:
Stream control is divorced from inviting a media server to a
conference. The only requirement is that the conference
initiation protocol either provides or can be used to create a
unique conference identifier. In particular, SIP [12] or H.323
may be used to invite a server to a conference.
Suitable for professional applications:
RTSP supports frame-level accuracy through SMPTE time stamps to
allow remote digital editing.
Presentation description neutral:
The protocol does not impose a particular presentation
description or metafile format and can convey the type of
format to be used. However, the presentation description must
contain at least one RTSP URI.
Proxy and firewall friendly:
The protocol should be readily handled by both application and
transport-layer (SOCKS [13]) firewalls. A firewall may need to
understand the SETUP method to open a ``hole'' for the UDP
media stream.
HTTP-friendly:
Where sensible, RTSP re-uses HTTP concepts, so that the
existing infrastructure can be re-used. This infrastructure
includes PICS (Platform for Internet Content Selection [20])
for associating labels with content. However, RTSP does not
just add methods to HTTP, since the controlling continuous
media requires server state in most cases.
Appropriate server control:
If a client can start a stream, it must be able to stop a
stream. Servers should not start streaming to clients in such a
way that clients cannot stop the stream.
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Transport negotiation:
The client can negotiate the transport method prior to actually
needing to process a continuous media stream.
Capability negotiation:
If basic features are disabled, there must be some clean
mechanism for the client to determine which methods are not
going to be implemented. This allows clients to present the
appropriate user interface. For example, if seeking is not
allowed, the user interface must be able to disallow moving a
sliding position indicator.
An earlier requirement in RTSP was multi-client capability.
However, it was determined that a better approach was to make sure
that the protocol is easily extensible to the multi-client
scenario. Stream identifiers can be used by several control
streams, so that ``passing the remote'' would be possible. The
protocol would not address how several clients negotiate access;
this is left to either a ``social protocol'' or some other floor
control mechanism.
1.5 Extending RTSP
Since not all media servers have the same functionality, media servers
by necessity will support different sets of requests. For example:
* A server may only be capable of playback, not recording and thus
has no need to support the RECORD request.
* A server may not be capable of seeking (absolute positioning),
say, if it is to support live events only.
* Some servers may not support setting stream parameters and thus
not support GET_PARAMETER and SET_PARAMETER.
A server SHOULD implement all header fields described in Section 12.
It is up to the creators of presentation descriptions not to ask the
impossible of a server. This situation is similar in HTTP/1.1, where
the methods described in [H19.6] are not likely to be supported across
all servers.
RTSP can be extended in three ways, listed in order of the magnitude
of changes supported:
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* Existing methods can be extended with new parameters, as long as
these parameters can be safely ignored by the recipient. (This is
equivalent to adding new parameters to an HTML tag.)
* New methods can be added. If the recipient of the message does not
understand the request, it responds with error code 501 (Not
implemented) and the sender should not attempt to use this method
again. A client may also use the OPTIONS method to inquire about
methods supported by the server. The server SHOULD list the
methods it supports using the Public response header.
* A new version of the protocol can be defined, allowing almost all
aspects (except the position of the protocol version number) to
change.
1.6 Overall Operation
Each presentation and media stream may be identified by an RTSP URL.
The overall presentation and the properties of the media the
presentation is made up of are defined by a presentation description
file, the format of which is outside the scope of this specification.
The presentation description file may be obtained by the client using
HTTP or other means such as email and may not necessarily be stored on
the media server.
For the purposes of this specification, a presentation description is
assumed to describe one or more presentations, each of which maintains
a common time axis. For simplicity of exposition and without loss of
generality, it is assumed that the presentation description contains
exactly one such presentation. A presentation may contain several
media streams.
The presentation description file contains a description of the media
streams making up the presentation, including their encodings,
language, and other parameters that enable the client to choose the
most appropriate combination of media. In this presentation
description, each media stream that is individually controllable by
RTSP is identified by an RTSP URL, which points to the media server
handling that particular media stream and names the stream stored on
that server. Several media streams can be located on different
servers; for example, audio and video streams can be split across
servers for load sharing. The description also enumerates which
transport methods the server is capable of.
Besides the media parameters, the network destination address and port
need to be determined. Several modes of operation can be
distinguished:
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Unicast:
The media is transmitted to the source of the RTSP request,
with the port number chosen by the client. Alternatively, the
media is transmitted on the same reliable stream as RTSP.
Multicast, server chooses address:
The media server picks the multicast address and port. This is
the typical case for a live or near-media-on-demand
transmission.
Multicast, client chooses address:
If the server is to participate in an existing multicast
conference, the multicast address, port and encryption key are
given by the conference description, established by means
outside the scope of this specification.
1.7 RTSP States
RTSP controls a stream which may be sent via a separate protocol,
independent of the control channel. For example, RTSP control may
occur on a TCP connection while the data flows via UDP. Thus, data
delivery continues even if no RTSP requests are received by the media
server. Also, during its lifetime, a single media stream may be
controlled by RTSP requests issued sequentially on different TCP
connections. Therefore, the server needs to maintain ``session state''
to be able to correlate RTSP requests with a stream. The state
transitions are described in Section A.
Many methods in RTSP do not contribute to state. However, the
following play a central role in defining the allocation and usage of
stream resources on the server: SETUP, PLAY, RECORD, PAUSE, and
TEARDOWN.
SETUP:
Causes the server to allocate resources for a stream and start
an RTSP session.
PLAY and RECORD:
Starts data transmission on a stream allocated via SETUP.
PAUSE:
Temporarily halts a stream, without freeing server resources.
TEARDOWN:
Frees resources associated with the stream. The RTSP session
ceases to exist on the server.
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1.8 Relationship with Other Protocols
RTSP has some overlap in functionality with HTTP. It also may interact
with HTTP in that the initial contact with streaming content is often
to be made through a web page. The current protocol specification aims
to allow different hand-off points between a web server and the media
server implementing RTSP. For example, the presentation description
can be retrieved using HTTP or RTSP. Having the presentation
description be returned by the web server makes it possible to have
the web server take care of authentication and billing, by handing out
a presentation description whose media identifier includes an
encrypted version of the requestor's IP address and a timestamp, with
a shared secret between web and media server.
However, RTSP differs fundamentally from HTTP in that data delivery
takes place out-of-band, in a different protocol. HTTP is an
asymmetric protocol, where the client issues requests and the server
responds. In RTSP, both the media client and media server can issue
requests. RTSP requests are also not stateless, in that they may set
parameters and continue to control a media stream long after the
request has been acknowledged.
Re-using HTTP functionality has advantages in at least two areas,
namely security and proxies. The requirements are very similar, so
having the ability to adopt HTTP work on caches, proxies and
authentication is valuable.
While most real-time media will use RTP as a transport protocol, RTSP
is not tied to RTP.
RTSP assumes the existence of a presentation description format that
can express both static and temporal properties of a presentation
containing several media streams.
2 Notational Conventions
Since many of the definitions and syntax 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 (RFC 2068).
All the mechanisms specified in this document are described in both
prose and an augmented Backus-Naur form (BNF) similar to that used in
RFC 2068 [H2.1]. It is described in detail in [14].
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In this draft, we use indented and smaller-type paragraphs to provide
background and motivation. Some of these paragraphs are marked with
HS, AR and RL, designating opinions and comments by the individual
authors which may not be shared by the co-authors and require
resolution.
3 Protocol Parameters
3.1 RTSP Version
[H3.1] applies, with HTTP replaced by RTSP.
3.2 RTSP URL
The ``rtsp'', ``rtspu'' and ``rtsps'' schemes are used to refer to
network resources via the RTSP protocol. This section defines the
scheme-specific syntax and semantics for RTSP URLs.
rtsp_URL = ( "rtsp:" | "rtspu:" | "rtsps:" )
"//" host [ ":" port ] [abs_path]
host = <A legal Internet host domain name of IP address
(in dotted decimal form), as defined by Section 2.1
of RFC 1123>
port = *DIGIT
abs_path is defined in [H3.2.1].
Note that fragment and query identifiers do not have a well-defined
meaning at this time, with the interpretation left to the RTSP
server.
The scheme rtsp requires that commands are issued via a reliable
protocol (within the Internet, TCP), while the scheme rtspu identifies
an unreliable protocol (within the Internet, UDP). The scheme rtsps
indicates that a TCP connection secured by TLS [7] must be used.
If the port is empty or not given, port 554 is assumed. The semantics
are that the identified resource can be controlled be RTSP at the
server listening for TCP (scheme ``rtsp'') connections or UDP (scheme
``rtspu'') packets on that port of host, and the Request-URI for the
resource is rtsp_URL.
The use of IP addresses in URLs SHOULD be avoided whenever possible
(see RFC 1924 [15]).
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A presentation or a stream is identified by an textual media
identifier, using the character set and escape conventions [H3.2] of
URLs [16]. URLs may refer to a stream or an aggregate of streams ie. a
presentation. Accordingly, requests described in Section 10 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.
For example, the RTSP URL
rtsp://media.example.com:554/twister/audiotrack
identifies 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.
Also, the RTSP URL
rtsp://media.example.com:554/twister
identifies the presentation ``twister'', which may be composed of
audio and video streams.
This does not imply a standard way to reference streams in URLs.
The presentation description defines the hierarchical relationships
in the presentation and the URLs for the individual streams. A
presentation description may name a stream 'a.mov' and the whole
presentation 'b.mov'.
The path components of the RTSP URL are opaque to the client and do
not imply any particular file system structure for the server.
This decoupling also allows presentation descriptions to be used
with non-RTSP media control protocols, simply by replacing the
scheme in the URL.
3.3 Conference Identifiers
Conference identifiers are opaque to RTSP and are encoded using
standard URI encoding methods (i.e., LWS is escaped with %). They can
contain any octet value. The conference identifier MUST be globally
unique. For H.323, the conferenceID value is to be used.
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conference-id = 1*OCTET ; LWS must be URL-escaped
Conference identifiers are used to allow to allow RTSP sessions to
obtain parameters from multimedia conferences the media server is
participating in. These conferences are created by protocols
outside the scope of this specification, e.g., H.323 [17] or SIP
[12]. Instead of the RTSP client explicitly providing transport
information, for example, it asks the media server to use the
values in the conference description instead. If the conference
participant inviting the media server would only supply a
conference identifier which is unique for that inviting party, the
media server could add an internal identifier for that party, e.g.,
its Internet address. However, this would prevent that the
conference participant and the initiator of the RTSP commands are
two different entities.
3.4 Session Identifiers
Session identifiers are opaque strings of arbitrary length. Linear
white space must be URL-escaped. A session identifier SHOULD be chosen
randomly and SHOULD be at least eight octets long to make guessing it
more difficult. (See Section 16).
session-id = 1*OCTET ; LWS must be URL-escaped
3.5 SMPTE Relative Timestamps
A SMPTE relative time-stamp expresses time relative to the start of
the clip. Relative timestamps are expressed as SMPTE time codes for
frame-level access accuracy. The time code has the format
hours:minutes:seconds:frames.subframes, with the origin at the start
of the clip. For NTSC, the frame rate is 29.97 frames per second. This
is handled by dropping the first two frame indices (values 00 and 01)
of every minute, except every tenth minute. If the frame value is
zero, it may be omitted. Subframes are measured in one-hundredth of a
frame.
smpte-range = "smpte" "=" smpte-time "-" [ smpte-time ]
smpte-time = 2DIGIT ":" 2DIGIT ":" 2DIGIT [ ":" 2DIGIT ] [ "." 2DIGIT]
Examples:
smpte=10:12:33:20-
smpte=10:07:33-
smpte=10:07:00-10:07:33:05.01
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3.6 Normal Play Time
Normal play time (NPT) indicates the stream absolute position relative
to the beginning of the presentation, measured in seconds and
microseconds. The beginning of a presentation corresponds to 0 seconds
and 0 microseconds. Negative values are not defined. The microsecond
field is always less than 1,000,000. NPT is defined as in DSM-CC [18]:
``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 (high negative scale ratio) and is fixed in pause mode. NPT is
(logically) equivalent to SMPTE time codes.'' [18]
npt-range = "npt" "=" npt-time "-" [ npt-time ]
npt-time = 1*DIGIT [ ":" *DIGIT ]
Examples:
npt=123:45-125
3.7 Absolute Time
Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT).
Fractions of a second may be indicated.
utc-range = "clock" "=" utc-time "-" [ utc-time ]
utc-time = utc-date "T" utc-time "Z"
utc-date = 8DIGIT ; < YYYYMMDD >
utc-time = 6DIGIT [ "." fraction ] ; < HHMMSS.fraction >
Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
UTC:
19961108T143720.25Z
Example
4 RTSP Message
RTSP is a text-based protocol and uses the ISO 10646 character set
in UTF-8 encoding (RFC 2044). Lines are terminated by CRLF, but
receivers should be prepared to also interpret CR and LF by themselves
as line terminators.
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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.
The 10646 character set avoids tricky character set switching, but
is invisible to the application as long as US-ASCII is being used.
This is also the encoding used for RTCP. ISO 8859-1 translates
directly into Unicode, with a high-order octet of zero. ISO 8859-1
characters with the most-significant bit set are represented as
1100001x 10xxxxxx.
RTSP messages can be carried over any lower-layer transport protocol
that is 8-bit clean.
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.
4.1 Message Types
See [H4.1]
4.2 Message Headers
See [H4.2]
4.3 Message Body
See [H4.3]
4.4 Message Length
When a message-body is included with a message, the length of that
body is determined by one of the following (in order of precedence):
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1.
Any response message which MUST NOT include a message-body
(such as the 1xx, 204, and 304 responses) is always terminated
by the first empty line after the header fields, regardless of
the entity-header fields present in the message. (Note: An
empty line consists of only CRLF.)
2.
If a Content-Length header field (section 12.15) is present,
its value in bytes represents the length of the message-body.
If this header field is not present, a value of zero is
assumed.
3.
By the server closing the connection. (Closing the connection
cannot be used to indicate the end of a request body, since
that would leave no possibility for the server to send back a
response.)
Note that RTSP does not (at present) support the HTTP/1.1 ``chunked''
transfer coding(see [H3.6]) and requires the presence of the
Content-Length header field.
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. Even though Content-Length must be present if there is
any entity body, the rules ensure reasonable behavior even if the
length is not given explicitly.
5 General Header Fields
See [H4.5], except that Pragma, Transfer-Encoding and Upgrade
headers are not defined:
general-header = Cache-Control ; Section 12.10
| Connection ; Section 12.12
| Date ; Section 12.17
| Via ; Section 12.39
6 Request
A request message from a client to a server or vice versa includes,
within the first line of that message, the method to be applied to the
resource, the identifier of the resource, and the protocol version in
use.
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Request = Request-Line ; Section 6.1
*( general-header ; Section 5
| request-header ; Section 6.2
| entity-header ) ; Section 8.1
CRLF
[ message-body ] ; Section 4.3
6.1 Request Line
Request-Line = Method SP Request-URI SP RTSP-Version SP seq-no CRLF
Method = "DESCRIBE" ; Section 10.2
| "ANNOUNCE" ; Section 10.3
| "GET_PARAMETER" ; Section 10.8
| "OPTIONS" ; Section 10.1
| "PAUSE" ; Section 10.6
| "PLAY" ; Section 10.5
| "RECORD" ; Section 10.11
| "REDIRECT" ; Section 10.10
| "SETUP" ; Section 10.4
| "SET_PARAMETER" ; Section 10.9
| "TEARDOWN" ; Section 10.7
| extension-method
extension-method = token
Request-URI = "*" | absolute_URI
RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT
seq-no = 1*DIGIT
6.2 Request Header Fields
request-header = Accept ; Section 12.1
| Accept-Encoding ; Section 12.2
| Accept-Language ; Section 12.3
| Authorization ; Section 12.5
| From ; Section 12.19
| If-Modified-Since ; Section 12.21
| Range ; Section 12.28
| Referer ; Section 12.29
| User-Agent ; Section 12.37
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Note that in contrast to HTTP/1.1, RTSP requests always contain the
absolute URL (that is, including the scheme, host and port) rather
than just the absolute path.
HTTP/1.1 requires servers to understand the absolute URL, 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.
The asterisk "*" in the Request-URI means that the request does not
apply to a particular resource, but to the server itself, and is only
allowed when the method used does not necessarily apply to a resource.
One example would be
OPTIONS * RTSP/1.0
7 Response
[H6] applies except that HTTP-Version is replaced by RTSP-Version.
Also, RTSP defines additional status codes and does not define some
HTTP codes. The valid response codes and the methods they can be used
with are defined in the table 1.
After receiving and interpreting a request message, the recipient
responds with an RTSP response message.
Response = Status-Line ; Section 7.1
*( general-header ; Section 5
| response-header ; Section 7.1.2
| entity-header ) ; Section 8.1
CRLF
[ message-body ] ; Section 4.3
7.1 Status-Line
The first line of a Response message is the Status-Line, consisting
of the protocol version followed by a numeric status code, the
sequence number of the corresponding request 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.
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Status-Line = RTSP-Version SP Status-Code SP seq-no SP Reason-Phrase CRLF
7.1.1 Status Code and Reason Phrase
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 section11. 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.
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:
* 1xx: Informational - Request received, continuing process
* 2xx: Success - The action was successfully received, understood,
and accepted
* 3xx: Redirection - Further action must be taken in order to
complete the request
* 4xx: Client Error - The request contains bad syntax or cannot be
fulfilled
* 5xx: Server Error - The server failed to fulfill an apparently
valid request
The individual values of the numeric status codes defined for
RTSP/1.0, and an example set of corresponding Reason-Phrase's, are
presented below. 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 status codes and adds
RTSP-specific status codes in the starting at 450 to avoid conflicts
with newly defined HTTP status codes.
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Status-Code = "100" ; Continue
| "200" ; OK
| "201" ; Created
| "300" ; Multiple Choices
| "301" ; Moved Permanently
| "302" ; Moved Temporarily
| "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
| "409" ; Conflict
| "410" ; Gone
| "411" ; Length Required
| "412" ; Precondition Failed
| "413" ; Request Entity Too Large
| "414" ; Request-URI Too Large
| "415" ; Unsupported Media Type
| "451" ; Parameter Not Understood
| "452" ; Conference Not Found
| "453" ; Not Enough Bandwidth
| "45x" ; Session Not Found
| "45x" ; Method Not Valid in This State
| "45x" ; Header Field Not Valid for Resource
| "45x" ; Invalid Range
| "45x" ; Parameter Is Read-Only
| "45x" ; Aggregate operation not allowed
| "45x" ; Only aggregate operation allowed
| "500" ; Internal Server Error
| "501" ; Not Implemented
| "502" ; Bad Gateway
| "503" ; Service Unavailable
| "504" ; Gateway Time-out
| "505" ; RTSP Version not supported
| extension-code
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extension-code = 3DIGIT
Reason-Phrase = *<TEXT, excluding CR, LF>
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 the exception that an unrecognized
response 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 entity returned with the response, since that
entity is likely to include human-readable information which will
explain the unusual status.
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Code Reason
100 Continue all
200 OK all
201 Created RECORD
300 Multiple Choices all
301 Moved Permanently all
302 Moved Temporarily all
303 See Other all
305 Use Proxy all
400 Bad Request all
401 Unauthorized all
402 Payment Required all
403 Forbidden all
404 Not Found all
405 Method Not Allowed all
406 Not Acceptable all
407 Proxy Authentication Required all
408 Request Timeout all
409 Conflict RECORD
410 Gone all
411 Length Required SETUP
412 Precondition Failed DESCRIBE, SETUP
413 Request Entity Too Large SETUP
414 Request-URI Too Long all
415 Unsupported Media Type SETUP
45x Session not found all
45x Invalid parameter SETUP
45x Not Enough Bandwidth SETUP
45x Illegal Conference Identifier SETUP
45x Illegal Session Identifier PLAY, RECORD, TEARDOWN
45x Parameter Is Read-Only SET_PARAMETER
45x Header Field Not Valid all
45x Method Not Valid In This State all
45x Aggregate operation not allowed all
45x Only aggregate operation allowed all
500 Internal Server Error all
501 Not Implemented all
502 Bad Gateway all
503 Service Unavailable all
504 Gateway Timeout all
505 RTSP Version Not Supported all
!
Table 1: Status codes and their usage with RTSP methods
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7.1.2 Response Header Fields
The response-header fields allow the request recipient to pass
additional information about the response which cannot be placed in
the Status-Line. These header fields give information about the server
and about further access to the resource identified by the
Request-URI.
response-header = Location ; Section 12.23
| Proxy-Authenticate ; Section 12.26
| Public ; Section 12.27
| Retry-After ; Section 12.30
| Server ; Section 12.33
| Vary ; Section 12.38
| WWW-Authenticate ; Section 12.40
Response-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of
response-header fields if all parties in the communication recognize
them to be response-header fields. Unrecognized header fields are
treated as entity-header fields.
8 Entity
Request and Response messages MAY transfer an entity if not
otherwise restricted by the request method or response status code. An
entity consists of entity-header fields and an entity-body, although
some responses will only include the entity-headers.
In this section, both sender and recipient refer to either the client
or the server, depending on who sends and who receives the entity.
8.1 Entity Header Fields
Entity-header fields define optional metainformation about the
entity-body or, if no body is present, about the resource identified
by the request.
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entity-header = Allow ; Section 12.4
| Content-Encoding ; Section 12.13
| Content-Language ; Section 12.14
| Content-Length ; Section 12.15
| Content-Type ; Section 12.16
| Expires ; Section 12.18
| Last-Modified ; Section 12.22
| extension-header
extension-header = message-header
The extension-header mechanism allows additional entity-header fields
to be defined without changing the protocol, but these fields cannot
be assumed to be recognizable by the recipient. Unrecognized header
fields SHOULD be ignored by the recipient and forwarded by proxies.
8.2 Entity Body
See [H7.2]
9 Connections
RTSP requests can be transmitted in several different ways:
* persistent transport connections used for several request-response
transactions;
* one connection per request/response transaction;
* connectionless mode.
The type of transport connection is defined by the RTSP URI
(Section 3.2). For the scheme ``rtsp'', a persistent connection is
assumed, while the scheme ``rtspu'' calls for RTSP requests to be send
without setting up a connection.
Unlike HTTP, RTSP allows the media server to send requests to the
media client. However, this is only supported for persistent
connections, as the media server otherwise has no reliable way of
reaching the client. Also, this is the only way that requests from
media server to client are likely to traverse firewalls.
9.1 Pipelining
A client that supports persistent connections or connectionless mode
MAY ``pipeline'' its requests (i.e., send multiple requests without
waiting for each response). A server MUST send its responses to those
requests in the same order that the requests were received.
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9.2 Reliability and Acknowledgements
Requests are acknowledged by the receiver unless they are sent to a
multicast group. If there is no acknowledgement, the sender may resend
the same message after a timeout of one round-trip time (RTT). The
round-trip time is estimated as in TCP (RFC TBD), with an initial
round-trip value of 500 ms. An implementation MAY cache the last RTT
measurement as the initial value for future connections. If a reliable
transport protocol is used to carry RTSP, the timeout value MAY be set
to an arbitrarily large value.
This can greatly increase responsiveness for proxies operating in
local-area networks with small RTTs. The mechanism is defined such
that the client implementation does not have be aware of whether a
reliable or unreliable transport protocol is being used. It is
probably a bad idea to have two reliability mechanisms on top of
each other, although the RTSP RTT estimate is likely to be larger
than the TCP estimate.
Each request carries a sequence number, which is incremented by one
for each request transmitted. If a request is repeated because of lack
of acknowledgement, the sequence number is incremented.
This avoids ambiguities when computing round-trip time estimates.
[TBD: An initial sequence number negotiation needs to be added for
UDP; otherwise, a new stream connection may see a request be
acknowledged by a delayed response from an earlier ``connection''.
This handshake can be avoided with a sequence number containing a
timestamp of sufficiently high resolution.]
The reliability mechanism described here does not protect against
reordering. This may cause problems in some instances. For example, a
TEARDOWN followed by a PLAY has quite a different effect than the
reverse. Similarly, if a PLAY request arrives before all parameters
are set due to reordering, the media server would have to issue an
error indication. Since sequence numbers for retransmissions are
incremented (to allow easy RTT estimation), the receiver cannot just
ignore out-of-order packets. [TBD: This problem could be fixed by
including both a sequence number that stays the same for
retransmissions and a timestamp for RTT estimation.]
Systems implementing RTSP MUST support carrying RTSP over TCP and MAY
support UDP. The default port for the RTSP server is 554 for both UDP
and TCP.
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A number of RTSP packets destined for the same control end point may
be packed into a single lower-layer PDU or encapsulated into a TCP
stream. RTSP data MAY be interleaved with RTP and RTCP packets. Unlike
HTTP, an RTSP message MUST contain a Content-Length header whenever
that message contains a payload. Otherwise, an RTSP packet is
terminated with an empty line immediately following the last message
header.
10 Method Definitions
The method token indicates the method to be performed on the
resource identified by the Request-URI. The method is case-sensitive.
New methods may be defined in the future. Method names may not start
with a $ character (decimal 24) and must be a token. Methods are
summarized in Table 2.
method direction object requirement
DESCRIBE C->S P,S recommended
ANNOUNCE C->S, S->C P,S optional
GET_PARAMETER C->S, S->C P,S optional
OPTIONS C->S P,S required
PAUSE C->S P,S recommended
PLAY C->S P,S required
RECORD C->S P,S optional
REDIRECT S->C P,S optional
SETUP C->S S required
SET_PARAMETER C->S, S->C P,S optional
TEARDOWN C->S P,S required
!
Table 2: Overview of RTSP methods, their direction, and what objects (P:
presentation, S: stream) they operate on
Notes on Table 2: PAUSE is recommended, but not required in that a
fully functional server can be built that does not support this
method, for example, for live feeds. If a server does not support a
particular method, it MUST return "501 Not Implemented" and a client
SHOULD not try this method again for this server.
10.1 OPTIONS
The behavior is equivalent to that described in [H9.2]. An OPTIONS
request may be issued at any time, e.g., if the client is about to try
a non-standard request. It does not influence server state.
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Example :
C->S: OPTIONS * RTSP/1.0 1
PEP: {{map "http://www.iana.org/rtsp/implicit-play"}}
{{map "http://www.iana.org/rtsp/record-feature"}}
C-PEP: {{map "http://www.iana.org/rtsp/udp-control"}}
{{map "http://www.iana.org/rtsp/gzipped-messages"}}
S->C: RTSP/1.0 200 2 OK
PEP-Info: {{map "http://www.iana.org/rtsp/implicit-play"}
{for "/" *}}
{{map "http://www.iana.org/rtsp/record-feature"}
{for "/" *}}
C-PEP-Info: {{map "http://www.iana.org/rtsp/udp-control"}
{for "/" *}}
{{map "http://www.iana.org/rtsp/gzipped-messages"}
{for "/" *}}
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Note that these are fictional features (though we may want to make
them real one day).
DESCRIBE
The DESCRIBE method retrieves the description of a presentation or
media object identified by the request URL from a server. It may use
the Accept header to specify the description formats that the client
understands. The server responds with a description of the requested
resource.
Example:
C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0 312
Accept: application/sdp, application/rtsl, application/mheg
S->C: RTSP/1.0 200 312 OK
Date: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 376
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v=0
o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
e=mjh@isi.edu (Mark Handley)
c=IN IP4 224.2.17.12/127
t=2873397496 2873404696
a=recvonly
m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31
m=whiteboard 32416 UDP WB
a=orient:portrait
ANNOUNCE
The ANNOUNCE method serves two purposes:
When sent from client to server, ANNOUNCE posts the description of a
presentation or media object identified by the request URL to a
server.
When sent from server to client, ANNOUNCE updates the session
description in real-time.
If a new media stream is added to a presentation (e.g., during a live
presentation), the whole presentation description should be sent
again, rather than just the additional components, so that components
can be deleted.
Example:
C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0 312
Date: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 332
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v=0
o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
e=mjh@isi.edu (Mark Handley)
c=IN IP4 224.2.17.12/127
t=2873397496 2873404696
a=recvonly
m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31
S->C: RTSP/1.0 200 312 OK
SETUP
The SETUP request for a URI specifies the transport mechanism to be
used for the streamed media. A client can issue a SETUP request for a
stream that is already playing to change transport parameters, which a
server MAY allow(If it does not allow it, it must respond with error
``45x Method not valid in this state'' ). For the benefit of any
intervening firewalls, a client must indicate the transport parameters
even if it has no influence over these parameters, for example, where
the server advertises a fixed multicast address.
Segregating content desciption into a DESCRIBE message and
transport information in SETUP avoids having firewall to parse
numerous different presentation description formats for information
which is irrelevant to transport.
The Transport header specifies the transport parameters acceptable to
the client for data transmission; the response will contain the
transport parameters selected by the server.
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0 302
Transport: RTP/AVP;port=4588
S->C: RTSP/1.0 200 302 OK
Date: 23 Jan 1997 15:35:06 GMT
Transport: RTP/AVP;port=4588
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PLAY
The PLAY method tells the server to start sending data via the
mechanism specified in SETUP. A client MUST NOT issue a PLAY request
until any outstanding SETUP requests have been acknowledged as
successful.
The PLAY request positions the normal play time to the beginning of
the range specified and delivers stream data until the end of the
range is reached. PLAY requests may be pipelined (queued); a server
MUST queue PLAY requests to be executed in order. That is, a PLAY
request arriving while a previous PLAY request is still active is
delayed until the first has been completed.
This allows precise editing.
For example, regardless of how closely spaced the two PLAY commands in
the example below arrive, the server will play first second 10 through
15 and then, immediately following, seconds 20 to 25 and finally
seconds 30 through the end.
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 835
Range: npt=10-15
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 836
Range: npt=20-25
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 837
Range: npt=30-
See the description of the PAUSE request for further examples.
A PLAY request without a Range header is legal. It starts playing a
stream from the beginning unless the stream has been paused. If a
stream has been paused via PAUSE, stream delivery resumes at the pause
point. If a stream is playing, such a PLAY request causes no further
action and can be used by the client to test server liveness.
The Range header may also contain a time parameter. This parameter
specifies a time in UTC at which the playback should start. If the
message is received after the specified time, playback is started
immediately. The time parameter may be used to aid in synchronisation
of streams obtained from different sources.
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For a on-demand stream, the server replies back with the actual range
that will be played back. This may differ from the requested range if
alignment of the requested range to valid frame boundaries is required
for the media source. If no range is specified in the request, the
current position is returned in the reply. The unit of the range in
the reply is the same as that in the request.
After playing the desired range, the presentation is automatically
paused, as if a PAUSE request had been issued.
The following example plays the whole presentation starting at SMPTE
time code 0:10:20 until the end of the clip. The playback is to start
at 15:36 on 23 Jan 1997.
C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0 833
Range: smpte=0:10:20-;time=19970123T153600Z
S->C: RTSP/1.0 200 833 OK
Date: 23 Jan 1997 15:35:06 GMT
Range: smpte=0:10:22-;time=19970123T153600Z
For playing back a recording of a live presentation, it may be
desirable to use clock units:
C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0 835
Range: clock=19961108T142300Z-19961108T143520Z
S->C: RTSP/1.0 200 833 OK
Date: 23 Jan 1997 15:35:06 GMT
A media server only supporting playback MUST support the npt format
and MAY support the clock and smpte formats.
PAUSE
The PAUSE request causes the stream delivery to be interrupted
(halted) temporarily. If the request URL names a stream, only playback
and recording of that stream is halted. For example, for audio, this
is equivalent to muting. If the request URL names a presentation or
group of streams, delivery of all currently active streams within the
presentation or group is halted. After resuming playback or recording,
synchronization of the tracks MUST be maintained. Any server resources
are kept.
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The PAUSE request may contain a Range header specifying when the
stream or presentation is to be halted. The header must contain
exactly one value rather than a time range. The normal play time for
the stream is set to that value. The pause request becomes effective
the first time the server is encountering the time point specified. If
this header is missing, stream delivery is interrupted immediately on
receipt of the message.
For example, if the server has play requests for ranges 10 to 15 and
20 to 29 pending and then receives a pause request for NPT 21, it
would start playing the second range and stop at NPT 21. If the pause
request is for NPT 12 and the server is playing at NPT 13 serving the
first play request, it stops immediately. If the pause request is for
NPT 16, it stops after completing the first play request and discards
the second play request.
As another example, if a server has received requests to play ranges
10 to 15 and then 13 to 20, that is, overlapping ranges, the PAUSE
request for NPT=14 would take effect while playing the first range,
with the second PLAY request effectively being ignored, assuming the
PAUSE request arrives before the server has started playing the
second, overlapping range. Regardless of when the PAUSE request
arrives, it sets the NPT to 14.
If the server has already sent data beyond the time specified in the
Range header, a PLAY would still resume at that point in time, as it
is assumed that the client has discarded data after that point. This
ensures continuous pause/play cycling without gaps.
Example:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 834
Session: 1234
S->C: RTSP/1.0 200 834 OK
Date: 23 Jan 1997 15:35:06 GMT
TEARDOWN
Stop the stream delivery for the given URI, freeing the resources
associated with it. If the URI is the presentation URI for this
presentation, any RTSP session identifier associated with the session
is no longer valid. Unless all transport parameters are defined by the
session description, a SETUP request has to be issued before the
session can be played again.
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Example:
C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0 892
Session: 1234
S->C: RTSP/1.0 200 892 OK
GET_PARAMETER
The requests retrieves the value of a parameter of a presentation or
stream specified in the URI. Multiple parameters can be requested in
the message body using the content type text/rtsp-parameters. Note
that parameters include server and client statistics. IANA registers
parameter names for statistics and other purposes. GET_PARAMETER with
no entity body may be used to test client or server liveness
(``ping'').
Example:
S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0 431
Content-Type: text/rtsp-parameters
Session: 1234
Content-Length: 15
packets_received
jitter
C->S: RTSP/1.0 200 431 OK
Content-Length: 46
Content-Type: text/rtsp-parameters
packets_received: 10
jitter: 0.3838
SET_PARAMETER
This method requests to set the value of a parameter for a
presentation or stream specified by the URI.
A request SHOULD only contain a single parameter to allow the client
to determine why a particular request failed. A server MUST allow a
parameter to be set repeatedly to the same value, but it MAY disallow
changing parameter values.
Note: transport parameters for the media stream MUST only be set with
the SETUP command.
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Restricting setting transport parameters to SETUP is for the
benefit of firewalls.
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.
A SET_PARAMETER request without parameters can be used as a way to
detect client or server liveness.
Example:
C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0 421
Content-type: text/rtsp-parameters
barparam: barstuff
S->C: RTSP/1.0 450 421 Invalid Parameter
Content-Length: 6
barparam
REDIRECT
A redirect request informs the client that it must connect to
another server location. It contains the mandatory header Location,
which indicates that the client should issue requests for that URL. It
may contain the parameter Range, which indicates when the redirection
takes effect.
This example request redirects traffic for this URI to the new server
at the given play time:
S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0 732
Location: rtsp://bigserver.com:8001
Range: clock=19960213T143205Z-
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RECORD
This method initiates recording a range of media data according to
the presentation description. The timestamp reflects start and end
time (UTC). If no time range is given, use the start or end time
provided in the presentation description. If the session has already
started, commence recording immediately.
The server decides whether to store the recorded data under the
request-URI or another URI. If the server does not use the
request-URI, the response SHOULD be 201 (Created) and contain an
entity which describes the status of the request and refers to the new
resource, and a Location header.
A media server supporting recording of live presentations MUST support
the clock range format; the smpte format does not make sense.
In this example, the media server was previously invited to the
conference indicated.
C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0 954
Session: 1234
Conference: 128.16.64.19/32492374
10.12 Embedded (Interleaved) Binary Data
Certain firewall designs and other circumstances may force a server
to interleave RTSP methods and stream data. This interleaving should
generally be avoided unless necessary since it complicates client and
server operation and imposes additional overhead. Interleaved binary
data SHOULD only be used if RTSP is carried over TCP.
Stream data such as RTP packets is encapsulated by an ASCII dollar
sign (24 decimal), followed by a one-byte channel identifier, followed
by the length of the encapsulated binary data as a binary, two-byte
integer in network byte order. The stream data follows immediately
afterwards, without a CRLF, but including the upper-layer protocol
headers. Each $ block contains exactly one upper-layer protocol data
unit, e.g., one RTP packet.
The channel identifier is defined in the Transport header 12.35.
C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0 2
Transport: RTP/AVP/TCP;channel=0
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S->C: RTSP/1.0 200 2 OK
Date: 05 Jun 1997 18:57:18 GMT
Transport: RTP/AVP/TCP;channel=0
Session: 12345
C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0 3
Session: 12345
S->C: RTSP/1.0 200 3 OK
Session: 12345
Date: 05 Jun 1997 18:59:15 GMT
S->C: $\000{2 byte length}{"length" bytes data, w/RTP header}
S->C: $\000{2 byte length}{"length" bytes data, w/RTP header}
11 Status Code Definitions
Where applicable, HTTP status [H10] codes are re-used. Status codes
that have the same meaning are not repeated here. See Table 1 for a
listing of which status codes may be returned by which request.
11.1 Redirection 3xx
See [H10.3].
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.
11.2 Client Error 4xx
11.2.1 405 Method Not Allowed
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, e.g., if a RECORD request is issued
even though the mode parameter in the Transport header only specified
PLAY.
11.2.2 451 Parameter Not Understood
The recipient of the request does not support one or more parameters
contained in the request.
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11.2.3 452 Conference Not Found
The conference indicated by a Conference header field is unknown to
the media server.
11.2.4 453 Not Enough Bandwidth
The request was refused since there was insufficient bandwidth. This
may, for example, be the result of a resource reservation failure.
11.2.5 45x Session Not Found
The RTSP session identifier is invalid or has timed out.
11.2.6 45x Method Not Valid in This State
The client or server cannot process this request in its current state.
11.2.7 45x Header Field Not Valid for Resource
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.
11.2.8 45x Invalid Range
The Range value given is out of bounds, e.g., beyond the end of the
presentation.
11.2.9 45x Parameter Is Read-Only
The parameter to be set by SET_PARAMETER can only be read, but not
modified.
11.2.10 45x Aggregate operation not allowed
The requested method may not be applied on the URL in question since
it is an aggregate(presentation) URL. The method may be applied on a
stream URL.
11.2.11 45x Only aggregate operation allowed
The requested method may not be applied on the URL in question since
it is not an aggregate(presentation) URL. The method may be applied on
the presentation URL.
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12 Header Field Definitions
HTTP/1.1 or other, non-standard header fields not listed here
currently have no well-defined meaning and SHOULD be ignored by the
recipient.
Tables 3 summarizes the header fields used by RTSP. Type ``g''
designates general request headers, to be found in both requests and
responses, type ``R'' designates request headers, type ``r'' response
headers, type ``e'' entity header fields. Fields marked with ``req.''
in the column labeled ``support'' MUST be implemented by the recipient
for a particular method, while fields marked ``opt.'' are optional.
Note that not all fields marked 'r' will be send in every request of
this type; merely, that client (for response headers) and server (for
request headers) MUST implement them. The last column lists the method
for which this header field is meaningful; the designation ``entity''
refers to all methods that return a message body. Within this
specification, DESCRIBE and GET_PARAMETER fall into this class.
If the field content does not apply to the particular resource, the
server MUST return status 45x (Header Field Not Valid for Resource).
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Header type support methods
Accept R opt. entity
Accept-Encoding R opt. entity
Accept-Language R opt. all
Authorization R opt. all
Bandwidth R opt. all
Blocksize R opt. all but OPTIONS, TEARDOWN
Cache-Control g opt. SETUP
Conference R opt. SETUP
Connection g req. all
Content-Encoding e req. SET_PARAMETER
Content-Encoding e req. DESCRIBE, ANNOUNCE
Content-Language e req. DESCRIBE, ANNOUNCE
Content-Length e req. SET_PARAMETER, ANNOUNCE
Content-Length e req. entity
Content-Type e req. SET_PARAMETER, ANNOUNCE
Content-Type r req. entity
Date g opt. all
Expires e opt. DESCRIBE, ANNOUNCE
From R opt. all
If-Modified-Since R opt. DESCRIBE, SETUP
Last-Modified e opt. entity
Public r opt. all
Range R opt. PLAY, PAUSE, RECORD
Range r opt. PLAY, PAUSE, RECORD
Referer R opt. all
Retry-After r opt. all
Scale Rr opt. PLAY, RECORD
Session Rr req. all but SETUP, OPTIONS
Server r opt. all
Speed Rr opt. PLAY
Transport Rr req. SETUP
Transport-Info r req. PLAY
User-Agent R opt. all
Via g opt. all
WWW-Authenticate r opt. all
!
Table 3: Overview of RTSP header fields
12.1 Accept
The Accept request-header field can be used to specify certain
presentation description content types which are acceptable for the
response.
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The ``level'' parameter for presentation descriptions is properly
defined as part of the MIME type registration, not here.
See [H14.1] for syntax.
Example of use:
Accept: application/rtsl, application/sdp;level=2
12.2 Accept-Encoding
See [H14.3]
12.3 Accept-Language
See [H14.4]. Note that the language specified applies to the
presentation description and any reason phrases, not the media
content.
12.4 Allow
The Allow response header field lists the methods supported by the
resource identified by the request-URI. The purpose of this field is
to strictly inform the recipient of valid methods associated with the
resource. An Allow header field must be present in a 405 (Method not
allowed) response.
Example of use:
Allow: SETUP, PLAY, RECORD, SET_PARAMETER
12.5 Authorization
See [H14.8]
12.6 Bandwidth
The Bandwidth request header field describes the estimated bandwidth
available to the client, expressed as a positive integer and measured
in bits per second.
The bandwidth available to the client may change during an RTSP
session, e.g., due to modem retraining.
Bandwidth = "Bandwidth" ":" 1*DIGIT
Example:
Bandwidth: 4000
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12.7 Blocksize
This 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 is a strictly
positive decimal number and measured in octets. The server only
returns an error (416) if the value is syntactically invalid.
12.8 C-PEP
This corresponds to the C-PEP: header in the ``Protocol Extension
Protocol'' defined in RFC XXXX [21]. This field differs from the PEP
field (Section 12.24) only in that it is hop-by-hop rather than
end-to-end as PEP is. Servers and proxies MUST parse this field and
MUST return "420 Bad Extension" when there is a PEP extension of
strength "must". See RFC XXXX for more details on this.
12.9 C-PEP-Info
This corresponds to the C-PEP-Info: header in the ``Protocol
Extension Protocol'' defined in RFC XXXX [21].
12.10 Cache-Control
The Cache-Control general header field is used to specify directives
that MUST be obeyed by all caching mechanisms along the
request/response chain.
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.
Cache-Control should only be specified in a SETUP request and its
response. Note: Cache-Control does not govern the caching of responses
as for HTTP, but rather of the stream identified by the SETUP request.
Responses to RTSP requests are not cacheable, except for responses to
DESCRIBE.
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Cache-Control = "Cache-Control" ":" 1#cache-directive
cache-directive = cache-request-directive
| cache-response-directive
cache-request-directive =
"no-cache"
| "max-stale"
| "min-fresh"
| "only-if-cached"
| cache-extension
cache-response-directive =
"public"
| "private"
| "no-cache"
| "no-transform"
| "must-revalidate"
| "proxy-revalidate"
| "max-age" "=" delta-seconds
| cache-extension
cache-extension = token [ "=" ( token | quoted-string ) ]
no-cache:
Indicates that the media stream 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.
public:
Indicates that the media stream is cachable by any cache.
private:
Indicates that the media stream is intended for a single user
and MUST NOT be cached by a shared cache. A private
(non-shared) cache may cache the media stream.
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no-transform:
An intermediate cache (proxy) may find it useful to convert the
media type of 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 entity-body. Therefore, if a
response includes the no-transform directive, an intermediate
cache or proxy MUST NOT change the encoding of the stream.
Unlike HTTP, RTSP does not provide for partial transformation
at this point, e.g., allowing translation into a different
language.
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 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 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.
max-stale:
Indicates that the client is willing to accept a media stream
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.
min-fresh:
Indicates that the client is willing to accept a media stream
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.
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must-revalidate:
When the must-revalidate directive is present in a SETUP
response received by a cache, that cache MUST NOT use the entry
after it becomes stale to respond to a subsequent request
without first revalidating it with the origin server. (I.e.,
the cache must do an end-to-end revalidation every time, if,
based solely on the origin server's Expires, the cached
response is stale.)
12.11 Conference
This request header field establishes a logical connection between a
conference, established using non-RTSP means, and an RTSP stream. The
conference-id must not be changed for the same RTSP session.
Conference = "Conference" ":" conference-id
Example:
Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
12.12 Connection
See [H14.10].
12.13 Content-Encoding
See [H14.12]
12.14 Content-Language
See [H14.13]
12.15 Content-Length
This field contains the length of the content of the method (i.e.
after the double CRLF following the last header). Unlike HTTP, it MUST
be included in all messages that carry content beyond the header
portion of the message. It is interpreted according to [H14.14].
12.16 Content-Type
See [H14.18]. Note that the content types suitable for RTSP are
likely to be restricted in practice to presentation descriptions and
parameter-value types.
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12.17 Date
See [H14.19].
12.18 Expires
The Expires entity-header field gives the date/time after which the
media-stream should be considered stale. 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 entity). See section
13.2 for further discussion of the expiration model.
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.
The format is an absolute date and time as defined by HTTP-date in
[H3.3]; it MUST be in RFC1123-date format:
Expires = "Expires" ":" HTTP-date
An example of its use is
Expires: Thu, 01 Dec 1994 16:00:00 GMT
RTSP/1.0 clients and caches MUST treat other invalid date formats,
especially including the value "0", as in the past (i.e., "already
expired").
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/1.0 servers should not send Expires dates more than one
year in the future.
The presence of an Expires header field with a date value of some time
in the future on a media stream that otherwise would by default be
non-cacheable indicates that the media stream is cachable, unless
indicated otherwise by a Cache-Control header field (Section 12.10).
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12.19 From
See [H14.22].
12.20 Host
This HTTP request header field is not needed for RTSP. It should be
silently ignored if sent.
12.21 If-Modified-Since
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 setup (SETUP); instead, a 304 (not modified)
response will be returned without any message-body.
If-Modified-Since = "If-Modified-Since" ":" HTTP-date
An example of the field is:
If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT
12.22 Last-Modified
The Last-Modified entity-header field indicates the date and time at
which the origin server believes the variant was last modified. See
[H14.29]. If the request URI refers to an aggregate, the field
indicates the last modification time across all leave nodes of that
aggregate.
12.23 Location
See [H14.30].
12.24 PEP
This corresponds to the PEP: header in the ``Protocol Extension
Protocol'' defined in RFC XXXX. Servers MUST parse this field and MUST
return ``420 Bad Extension'' when there is a PEP extension of strength
``must'' (see RFC XXXX).
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12.25 PEP-Info
This corresponds to the PEP-Info: header in the ``Protocol Extension
Protocol'' defined in RFC XXXX.
12.26 Proxy-Authenticate
See [H14.33].
12.27 Public
See [H14.35].
12.28 Range
This request header field specifies a range of time. The range can
be specified in a number of units. This specification defines the
smpte (see Section 3.5) and clock (see Section 3.7) range units.
Within RTSP, byte ranges [H14.36.1] are not meaningful and MUST NOT be
used. The header may also contain a time parameter in UTC, specifying
the time at which the operation is to be made effective. Servers
supporting the Range header MUST understand the NPT and SMPTE range
formats.
Range = "Range" ":" 1#ranges-specifier [ ";" "time" "=" utc-time ]
ranges-specifier = npt-range | utc-range | smpte-range
Example:
Range: clock=19960213T143205Z-;time=19970123T143720Z
The notation is similar to that used for the HTTP/1.1 header. It
allows to select a clip from the media object, to play from a given
point to the end and from the current location to a given point.
The start of playback can be scheduled for at any time in the
future, although a server may refuse to keep server resources for
extended idle periods.
12.29 Referer
See [H14.37]. The URL refers to that of the presentation
description, typically retrieved via HTTP.
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12.30 Retry-After
See [H14.38].
12.31 Scale
A scale value of 1 indicates normal play or record 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 normal play time increase at twice the wallclock rate. For every
second of elapsed (wallclock) time, 2 seconds of content will be
delivered. A negative value indicates reverse direction.
Unless requested otherwise by the Speed parameter, the data rate
SHOULD not be changed. Implementation of scale changes depends on the
server and media type. For video, a server may, for example, deliver
only key frames or selected key frames. For audio, it may time-scale
the audio while preserving pitch or, less desirably, deliver fragments
of audio.
The server should try to approximate the viewing rate, but may
restrict the range of scale values that it supports. The response MUST
contain the actual scale value chosen by the server.
If the request contains a Range parameter, the new scale value will
take effect at that time.
Scale = "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ]
Example of playing in reverse at 3.5 times normal rate:
Scale: -3.5
12.32 Speed
This request header fields parameter requests the server to deliver
data to the client at a particular speed, contingent on the server's
ability and desire to serve the media stream at the given speed.
Implementation by the server is OPTIONAL. The default is the bit rate
of the stream.
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The parameter value is expressed as a decimal ratio, e.g., a value of
2.0 indicates that data is to be delivered twice as fast as normal. A
speed of zero is invalid. If the request contains a Range parameter,
the new speed value will take effect at that time.
Speed = "Speed" ":" 1*DIGIT [ "." *DIGIT ]
Example:
Speed: 2.5
Use of this field changes the bandwidth used for data delivery. It is
meant for use in specific circumstances where preview of the
presentation at a higher or lower rate is necessary. 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. When data is delivered over UDP, it is highly
recommended that means such as RTCP be used to track packet loss
rates.
12.33 Server
See [H14.39]
12.34 Session
This request and response header field identifies an RTSP session,
started by the media server in a SETUP response and concluded by
TEARDOWN on the presentation URL. The session identifier is chosen by
the media server (see Section 3.4). Once a client receives a Session
identifier, it MUST return it for any request related to that session.
Session = "Session" ":" session-id
Note that a session identifier identifies a RTSP session across
transport sessions or connections. Control messages for more than one
RTSP URL may be sent within a single RTSP session. Hence, it is
possible that clients use the same session for controlling many
streams comprising a presentation, as long as all the streams come
from the same server. (See example in Section 14). However, multiple
``user'' sessions for the same URL from the same client MUST use
different session identifiers.
The session identifier is needed to distinguish several delivery
requests for the same URL coming from the same client.
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12.35 Transport
This request 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.
Transports are comma separated, listed in order of preference.
Parameters may be added to each tranpsort, separated by a semicolon.
The Transport header MAY also be used to change certain transport
parameters. A server MAY refuse to change parameters of an existing
stream.
The server MAY return a Transport response header in the response to
indicate the values actually chosen.
A Transport request header field may contain a list of transport
options acceptable to the client. In that case, the server MUST return
a single option which was actually chosen.
The syntax for the transport specifier is
transport/profile/lower-transport. Defaults for "lower-transport" are
specific to the profile. For RTP/AVP, the default is UDP.
Below are the configuration parameters associated with transport:
General parameters:
destination:
The address to which a stream will be sent. The client may
specify the multicast address with the destination parameter. A
server SHOULD authenticate the client and SHOULD log such
attempts before allowing the client to direct a media stream to
an address not chosen by the server to avoid becoming the
unwitting perpetrator of a remote-controlled denial-of-service
attack. This is particularly important if RTSP commands are
issued via UDP, but TCP cannot be relied upon as reliable means
of client identification by itself. A server SHOULD not allow a
client to direct media streams to an address that differs from
the address commands are coming from.
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mode:
The mode parameter indicates the methods to be supported for
this session. Valid values are PLAY and RECORD. If not
provided, the default is PLAY. For RECORD, the append flag
indicates that the media data should be appended to the
existing resource rather than overwriting it. If appending is
requested and the server does not support this, it MUST refuse
the request rather than overwrite the resouce identified by the
URI. The append parameter is ignored if the mode parameter does
not contain RECORD.
interleaved:
The interleaved parameter implies mixing the media stream with
the control stream, in whatever protocol is being used by the
control stream. Currently, the next-layer protocols RTP is
defined. The `channel' parameter defines the channel number to
be used in the $ statement (see section 10.12).
Multicast specific:
ttl:
multicast time-to-live
RTP Specific:
compressed:
Boolean parameter indicating compressed RTP according to RFC
XXXX.
port:
RTP/RTCP destination ports on client. The client receives RTCP
reports on the value of port plus one, as is standard RTP
convention.
cport:
the control port that the data server wishes the client to send
its RTCP reports to.
ssrc:
Indicates the RTP SSRC [19, Sec. 3] value that should be
(request) or will be (response) used by the media server. This
parameter is only valid for unicast transmission. It identifies
the synchronization source to be associated with the media
stream.
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Transport = "Transport" ":"
1#transport-protocol/profile[/lower-transport] *parameter
transport-protocol = "RTP"
profile = "AVP"
lower-transport = "TCP" | "UDP"
parameter = ";" "destination" [ "=" address ]
| ";" "compressed"
| ";" "channel" "=" channel
| ";" "append"
| ";" "ttl" "=" ttl
| ";" "port" "=" port
| ";" "cport" "=" port
| ";" "ssrc" "=" ssrc
| ";" "mode" = <"> 1#mode <">
ttl = 1*3(DIGIT)
port = 1*5(DIGIT)
ssrc = 8*8(HEX)
channel = 1*3(DIGIT)
address = host
mode = "PLAY" | "RECORD" *parameter
Example:
Transport: RTP/AVP;compressed;ttl=127;port=3456;
mode="PLAY,RECORD;append"
The Transport header is restricted to describing a single RTP
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.
12.36 Transport-Info
This field is used to set Transport specific parameters in the PLAY
response.
seq:
Indicates the sequence number of the first packet of the
stream. 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.
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Transport-Info = "Transport-Info" ":"
1#transport-protocol/profile[/lower-transport] ";"
streamid
*parameter
transport-protocol = "RTP"
profile = "AVP"
lower-transport = "TCP" | "UDP"
stream-id = "streamid" "=" streamid
parameter = ";" "seq" "=" sequence number
sequence-number = 1*16(DIGIT)
Example:
Transport-Info: RTP/AVP;streamid=0;seq=43754027,
RTP/AVP;streamid=1;seq=34834738
12.37 User-Agent
See [H14.42]
12.38 Vary
See [H14.43]
12.39 Via
See [H14.44].
12.40 WWW-Authenticate
See [H14.46].
13 Caching
In HTTP, response-request pairs are cached. RTSP differs
significantly in that respect. Responses are not cachable, with the
exception of the stream 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.
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On receiving a SETUP or PLAY request, the proxy would ascertain as to
whether it has an up-to-date copy of the continuous media content. If
not, it would modify the SETUP transport parameters as appropriate and
forward the request to the origin server. Subsequent control commands
such as PLAY or PAUSE would pass the proxy unmodified. The proxy would
then pass the continuous media data to the client, while possibly
making a local copy for later re-use. The exact behavior allowed to
the cache is given by the cache-response directives described in
Section 12.10. A cache MUST answer any DESCRIBE requests if it is
currently serving the stream to the requestor, as it is possible that
low-level details of the stream description may have changed on the
origin-server.
Note that an RTSP cache, unlike the HTTP 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.
To the client, an RTSP proxy cache would appear like a regular media
server, to the media origin server like a client. Just like an HTTP
cache has to store the content type, content language, etc. for the
objects it caches, a media cache has to store the presentation
description. Typically, a cache would eliminate all
transport-references (that is, 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.
14 Examples
The following examples reference stream description formats that are
not finalized, such as RTSL and SDP. Please do not use these examples
as a reference for those formats.
14.1 Media on Demand (Unicast)
Client C requests a movie from 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 time line of the movie.
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In our example, the client is only interested in the last part of the
movie. The server requires authentication for this movie.
C->W: GET /twister.sdp HTTP/1.1
Host: www.example.com
Accept: application/sdp
W->C: HTTP/1.0 200 OK
Content-Type: application/sdp
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
m=audio 0 RTP/AVP 0
a=murl:rtsp://audio.example.com/twister/audio.en
m=video 0 RTP/AVP 31
a=murl:rtsp://audio.example.com/twister/video
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0 1
Transport: rtp/udp;port=3056
A->C: RTSP/1.0 200 1 OK
Session: 1234
C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0 1
Transport: rtp/udp;port=3058
V->C: RTSP/1.0 200 1 OK
Session: 1235
C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0 2
Session: 1235
Range: smpte=0:10:00-
V->C: RTSP/1.0 200 2 OK
C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0 2
Session: 1234
Range: smpte=0:10:00-
A->C: RTSP/1.0 200 2 OK
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0 3
Session: 1234
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A->C: RTSP/1.0 200 3 OK
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0 3
Session: 1235
V->C: RTSP/1.0 200 3 OK
Even though the audio and video track are on two different servers,
and may start at slightly different times and may drift with respect
to each other, the client can synchronize the two using standard RTP
methods, in particular the time scale contained in the RTCP sender
reports.
14.2 Streaming of 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 a
RTSP presentation, with each of its components being RTSP streams.
Container files are a widely used means to store such presentations.
While the components are essentially transported as independant
streams, it is desirable to maintain a common context for those
streams at the server end.
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.
It is also possible that the presentation author may wish to prevent
selective retreival of the streams by 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 URL.
The following is an example of using a single RTSP session to control
multiple streams. It also illustrates the use of aggregate URLs.
Client C requests a presentation from media server M . The movie is
stored in a container file. The client has obtained a RTSP URL to the
container file.
C->M: DESCRIBE rtsp://foo/twister RTSP/1.0 1
M->C: RTSP/1.0 200 1 OK
Content-Type: application/sdp
Content-Length: 64
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s= sample rtsp presentation
r = rtsp://foo/twister /* aggregate URL*/
m= audio 0 RTP/AVP 0
r = rtsp://foo/twister/audio
m=video 0 RTP/AVP 26
r = rtsp://foo/twister/video
C->M: SETUP rtsp://foo/twister/audio RTSP/1.0 2
Transport: RTP/AVP;port=8000
M->C: RTSP/1.0 200 2 OK
Session: 1234
C->M: SETUP rtsp://foo/twister/video RTSP/1.0 3
Transport: RTP/AVP;port=8002
Session: 1234
M->C: RTSP/1.0 200 3 OK
Session: 1234
C->M: PLAY rtsp://foo/twister RTSP/1.0 4
Range: npt=0-
Session: 1234
M->C: RTSP/1.0 200 4 OK
Session: 1234
C->M: PAUSE rtsp://foo/twister/video RTSP/1.0 5
Session: 1234
M->C: RTSP/1.0 4xx 5 Only aggregate operation allowed
C->M: PAUSE rtsp://foo/twister RTSP/1.0 6
Session: 1234
M->C: RTSP/1.0 200 6 OK
Session: 1234
C->M: SETUP rtsp://foo/twister RTSP/1.0 7
Transport: RTP/AVP;port=10000
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M->C: RTSP/1.0 4xx 7 Aggregate operation not allowed
In the first instance of failure, the client tries to pause one
stream(in this case video) of the presentation which is disallowed for
that presentation by the server. In the second instance, the aggregate
URL may not be used for SETUP and one control message is required per
stream to setup transport parameters.
This keeps the syntax of the Transport header simple, and allows
easy parsing of transport information by firewalls.
14.3 Live Media Presentation Using Multicast
The media server M chooses the multicast address and port. Here, we
assume that the web server only contains a pointer to the full
description, while the media server M maintains the full description.
C->W: GET /concert.sdp HTTP/1.1
Host: www.example.com
W->C: HTTP/1.1 200 OK
Content-Type: application/rtsl
<session>
<track src="rtsp://live.example.com/concert/audio">
</session>
C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0 1
M->C: RTSP/1.0 200 1 OK
Content-Type: application/sdp
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
m=audio 3456 RTP/AVP 0
c=IN IP4 224.2.0.1/16
C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0 2
Transport: multicast=224.2.0.1; port=3456; ttl=16
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C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0 3
M->C: RTSP/1.0 200 3 OK
The attempt to position the stream fails since this is a live
presentation.
14.4 Playing media into an existing session
A conference participant C wants to have the media server M play back
a demo tape into an existing conference. When retrieving the
presentation description, C indicates to the media server that the
network addresses and encryption keys are already given by the
conference, so they should not be chosen by the server. The example
omits the simple ACK responses.
C->M: DESCRIBE rtsp://server.example.com/demo/548/sound RTSP/1.0 1
Accept: application/sdp
M->C: RTSP/1.0 200 1 OK
Content-type: application/rtsl
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
m=audio 0 RTP/AVP 0
C->M: SETUP rtsp://server.example.com/demo/548/sound RTSP/1.0 2
Conference: 218kadjk
14.5 Recording
The conference participant C asks the media server M to record a
meeting. If the presentation description contains any alternatives,
the server records them all.
C->M: DESCRIBE rtsp://server.example.com/meeting RTSP/1.0 90
Content-Type: application/sdp
v=0
s=Mbone Audio
i=Discussion of Mbone Engineering Issues
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M->C: RTSP/1.0 200 90 OK
C->S: SETUP rtsp://server.example.com/meeting RTSP/1.0 91
Transport: RTP/AVP;mode=record
S->C: RTSP/1.0 200 91 OK
Transport: RTP/AVP;port=3244;mode=record
Session: 508876
C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0 92
Session: 508876
Range: clock 19961110T1925-19961110T2015
15 Syntax
The RTSP syntax is described in an augmented Backus-Naur form (BNF)
as used in RFC 2068 (HTTP/1.1).
15.1 Base Syntax
OCTET = <any 8-bit sequence of data>
CHAR = <any US-ASCII character (octets 0 - 127)>
UPALPHA = <any US-ASCII uppercase letter "A".."Z">
LOALPHA = <any US-ASCII lowercase letter "a".."z">
ALPHA = UPALPHA | LOALPHA
DIGIT = <any US-ASCII digit "0".."9">
CTL = <any US-ASCII control character
(octets 0 - 31) and DEL (127)>
CR = <US-ASCII CR, carriage return (13)>
LF = <US-ASCII LF, linefeed (10)>
SP = <US-ASCII SP, space (32)>
HT = <US-ASCII HT, horizontal-tab (9)>
<"> = <US-ASCII double-quote mark (34)>
CRLF = CR LF
LWS = [CRLF] 1*( SP | HT )
TEXT = <any OCTET except CTLs>
tspecials = "(" | ")" | "<" | ">" | "@"
| "," | ";" | ":" | "\" | <">
| "/" | "[" | "]" | "?" | "="
| "{" | "}" | SP | HT
token = 1*<any CHAR except CTLs or tspecials>
quoted-string = ( <"> *(qdtext) <"> )
qdtext = <any TEXT except <">>
quoted-pair = "\" CHAR
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message-header = field-name ":" [ field-value ] CRLF
field-name = token
field-value = *( field-content | LWS )
field-content = <the OCTETs making up the field-value and consisting
of either *TEXT or combinations of token, tspecials,
and quoted-string>
16 Security Considerations
The protocol offers the opportunity for a remote-controlled
denial-of-service attack.
The attacker, using a forged source IP address, can ask for a stream
to be played back to that forged IP address. Thus, an RTSP server
SHOULD only allow client-specified destinations for RTSP-initiated
traffic flows if the server has verified the client's identity, e.g.,
using the RTSP authentication mechanisms.
Since there is no 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 would affect unsuspecting
clients. This does not require spoofing of network packet addresses.
The server SHOULD use a large random session identifier to make this
attack more difficult.
Both problems can be be prevented by appropriate authentication.
Servers SHOULD implement both basic and digest [8] authentication.
In addition, the security considerations outlined in [H15] apply.
A RTSP Protocol State Machines
The RTSP client and server state machines describe the behavior of
the protocol from RTSP session initialization through RTSP session
termination.
State is defined on a per object basis. An object is uniquely
identified by the stream URL and the RTSP session identifier. Any
request/reply using aggregate URLs denoting RTSP presentations
comprised of multiple streams will have an effect on the individual
states of all the streams. For example, if the presentation /movie
contains two streams /movie/audio and /movie/video, then the following
command:
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PLAY rtsp://foo.com/movie RTSP/1.0 559
Session: 12345
will have an effect on the states of movie/audio and movie/video.
This example does not imply a standard way to represent streams in
URLs or a relation to the filesystem. See Section 3.2.
The requests OPTIONS, DESCRIBE, GET_PARAMETER, SET_PARAMETER do not
have any effect on client or server state and are therefore not listed
in the state tables.
A.1 Client State Machine
The client can assume the following states:
Init:
SETUP has been sent, waiting for reply.
Ready:
SETUP reply received OR after playing, PAUSE reply received.
Playing:
PLAY reply received
Recording:
RECORD reply received
In general, the client changes state on receipt of replies to
requests. Note that some requests are effective at a future time or
position(such as a PAUSE), and state also changes accordingly. If no
explicit SETUP is required for the object (for example, it is
available via a multicast group), state begins at READY. In this case,
there are only two states, READY and PLAYING.
The client also changes state from Playing/Recording to Ready when the
end of the requested range is reached.
The ``next state'' column indicates the state assumed after receiving
a success response (2xx). If a request yields a status code of 3xx,
the state becomes Init, and a status code of 4xx yields no change in
state. Messages not listed for each state MUST NOT be issued by the
client in that state, with the exception of messages not affecting
state, as listed above. Receiving a REDIRECT from the server is
equivalent to receiving a 3xx redirect status from the server.
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state message next state
Init SETUP Ready
TEARDOWN Init
Ready PLAY Playing
RECORD Recording
TEARDOWN Init
Playing PAUSE Ready
TEARDOWN Init
PLAY Playing
SETUP Playing (changed transport)
Recording PAUSE Ready
TEARDOWN Init
RECORD Recording
SETUP Recording (changed transport)
A.2 Server State Machine
The server can assume the following states:
Init:
The initial state, no valid SETUP received.
Ready:
Last SETUP received was successful, reply sent or after
playing, last PAUSE received was successful, reply sent.
Playing:
Last PLAY received was successful, reply sent. Data is being
sent.
Recording:
The server is recording media data.
In general,the server changes state on receiving requests. If the
server is in state Playing or Recording and in unicast mode, it MAY
revert to Init and tear down the RTSP session if it has not received
``wellness'' information, such as RTCP reports, from the client for a
defined interval, with a default of one minute. If the server is in
state Ready, it MAY revert to Init if it does not receive an RTSP
request for an interval of more than one minute. Note that some
requests(such as PAUSE) may be effective at a future time or position,
and server state transitions at the appropriate time. The server
reverts from state Playing or Recording to state Ready at the end of
the range requested by the client.
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The REDIRECT message, when sent, is effective immediately unless it
has a Range: header specifying when the redirect is effective. In such
a case, server state will also change at the appropriate time.
If no explicit SETUP is required for the object, state starts at
READY, there are only two states READY and PLAYING.
The ``next state'' column indicates the state assumed after sending a
success response (2xx). If a request results in a status code of 3xx,
the state becomes Init. A status code of 4xx results in no change.
state message next state
Init SETUP Ready
TEARDOWN Init
Ready PLAY Playing
SETUP Ready
TEARDOWN Init
RECORD Recording
Playing PLAY Playing
PAUSE Ready
TEARDOWN Init
SETUP Playing
Recording RECORD Recording
PAUSE Ready
TEARDOWN Init
SETUP Recording
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B Open Issues
1.
Define text/rtsp-parameter MIME type.
2.
Reverse: Scale: -1, with reversed start times, or both?
3.
HS believes that RTSP should only control individual media
objects rather than aggregates. This avoids disconnects between
presentation descriptions and streams and avoids having to deal
separately with single-host and multi-host case. Cost: several
PLAY/PAUSE/RECORD in one packet, one for each stream.
4.
Allow changing of transport for a stream that's playing? May
not be a great idea since the same can be accomplished by tear
down and re-setup. Exception: near-video-on-demand, where the
server changes the address in a PLAY response. Servers may not
be able to reliably send TEARDOWN to clients and the client
wouldn't know why this happened in any event.
5.
How does the server get back to the client unless a persistent
connection is used? Probably cannot, in general.
6.
Server issues TEARDOWN and other 'event' notifications to
client? This raises the problem discussed in the previous open
issue, but is useful for the client if the data stream contains
no end indication.
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C Changes
Since the March 1997 version, the following changes were made:
* Allowing the Transport header to direct media streams to unicast
and multicast addresses, with an appropriate warning about
denial-of-service attacks.
* Add mode parameter to Transport header to allow RECORD or PLAY.
* The Embedded binary data section was modified to clearly indicate
the stream the data corresponds to, and a reference to the
Transport header was added.
* The Transport header format has been changed to use a more general
means to specify data channel and application level protocol. It
also conveys the port to be used at the server for RTCP messages,
and the start sequence number that will be used in the RTP
packets.
* The use of the Session: header has been enhanced. Requests for
multiple URLs may be sent in a single session.
* There is a distinction between aggregate(presentation) URLs and
stream URLs. Error codes have been added to reflect the fact that
some methods may be allowed only on a particular type of URL.
* Example showing the use of aggregate/presentation control using a
single RTSP session has been added.
* Support for the PEP(Protocol Extension Protocol) headers has been
added.
* Server-Client DESCRIBE messages have been renamed to ANNOUNCE for
better clarity and differentiation.
Note that this list does not reflect minor changes in wording or
correction of typographical errors.
D Author Addresses
Henning Schulzrinne
Dept. of Computer Science
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
electronic mail: schulzrinne@cs.columbia.edu
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Anup Rao
Netscape Communications Corp.
501 E. Middlefield Road
Mountain View, CA 94043
USA
electronic mail: anup@netscape.com
Robert Lanphier
Progressive Networks
1111 Third Avenue Suite 2900
Seattle, WA 98101
USA
electronic mail: robla@prognet.com
E Acknowledgements
This draft is based on the functionality of the original RTSP draft
submitted in October 96. It also borrows format and descriptions from
HTTP/1.1.
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:
Rahul Agarwal Eduardo F. Llach
Bruce Butterfield Rob McCool
Steve Casner David Oran
Martin Dunsmuir Sujal Patel
Eric Fleischman
Mark Handley Igor Plotnikov
Peter Haight Pinaki Shah
Brad Hefta-Gaub Jeff Smith
John K. Ho Alexander Sokolsky
Ruth Lang Dale Stammen
Stephanie Leif John Francis Stracke
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References
1
H. Schulzrinne, ``RTP profile for audio and video conferences
with minimal control,'' RFC 1890, Internet Engineering Task
Force, Jan. 1996.
2
D. Kristol and L. Montulli, ``HTTP state management
mechanism,'' RFC 2109, Internet Engineering Task Force, Feb.
1997.
3
F. Yergeau, G. Nicol, G. Adams, and M. Duerst,
``Internationalization of the hypertext markup language,'' RFC
2070, Internet Engineering Task Force, Jan. 1997.
4
S. Bradner, ``Key words for use in RFCs to indicate requirement
levels,'' RFC 2119, Internet Engineering Task Force, Mar. 1997.
5
R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T.
Berners-Lee, ``Hypertext transfer protocol - HTTP/1.1,'' RFC
2068, Internet Engineering Task Force, Jan. 1997.
6
M. Handley, ``SDP: Session description protocol,'' Internet
Draft, Internet Engineering Task Force, Nov. 1996.
Work in progress.
7
A. Freier, P. Karlton, and P. Kocher, ``The TLS protocol,''
Internet Draft, Internet Engineering Task Force, Dec. 1996.
Work in progress.
8
J. Franks, P. Hallam-Baker, J. Hostetler, P. A. Luotonen, and
E. L. Stewart, ``An extension to HTTP: digest access
authentication,'' RFC 2069, Internet Engineering Task Force,
Jan. 1997.
9
J. Postel, ``User datagram protocol,'' STD 6, RFC 768, Internet
Engineering Task Force, Aug. 1980.
10
R. Hinden and C. Partridge, ``Version 2 of the reliable data
protocol (RDP),'' RFC 1151, Internet Engineering Task Force,
Apr. 1990.
11
J. Postel, ``Transmission control protocol,'' STD 7, RFC 793,
Internet Engineering Task Force, Sept. 1981.
12
M. Handley, H. Schulzrinne, and E. Schooler, ``SIP: Session
initiation protocol,'' Internet Draft, Internet Engineering
Task Force, Dec. 1996.
Work in progress.
13
P. McMahon, ``GSS-API authentication method for SOCKS version
5,'' RFC 1961, Internet Engineering Task Force, June 1996.
14
D. Crocker, ``Augmented BNF for syntax specifications: ABNF,''
Internet Draft, Internet Engineering Task Force, Oct. 1996.
Work in progress.
15
R. Elz, ``A compact representation of IPv6 addresses,'' RFC
1924, Internet Engineering Task Force, Apr. 1996.
16
T. Berners-Lee, L. Masinter, and M. McCahill, ``Uniform
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17
International Telecommunication Union, ``Visual telephone
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non-guaranteed quality of service,'' Recommendation H.323,
Telecommunication Standardization Sector of ITU, Geneva,
Switzerland, May 1996.
18
ISO/IEC, ``Information technology - generic coding of moving
pictures and associated audio informaiton - part 6: extension
for digital storage media and control,'' Draft International
Standard ISO 13818-6, International Organization for
Standardization ISO/IEC JTC1/SC29/WG11, Geneva, Switzerland,
Nov. 1995.
19
H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson,
``RTP: a transport protocol for real-time applications,'' RFC
1889, Internet Engineering Task Force, Jan. 1996.
20
J. Miller, P. Resnick, and D. Singer, ``Rating Services and
Rating Systems(and Their Machine Readable Descriptions), ''
REC-PICS-services-961031, Worldwide Web Consortium, Oct. 1996.
21
D. Connolly, R. Khare, H.F. Nielsen, ``PEP - an Extension
Mechanism for HTTP", Internet draft, work-in-progress. W3C
Draft WD-http-pep-970714
http://www.w3.org/TR/WD-http-pep-970714, July, 1996.
H. Schulzrinne, A. Rao, R. Lanphier Page 71
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