One document matched: draft-cbran-rtcweb-protocols-00.txt
Network Working Group C. Bran
Internet-Draft C. Jennings
Intended status: Standards Track Cisco
Expires: December 7, 2011 June 5, 2011
RTC-Web Communications Protocols
draft-cbran-rtcweb-protocols-00
Abstract
The real time communications web (RTC-Web) will enable applications
such as web browsers to natively support real time interactive voice
and video. This document outlines the communication protocols for
realizing RTC-Web functionality within applications such as web
browsers. In addition to communications protocols, this document
proposes a set of application programming interface requirements for
controlling the protocol stack.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. This document may not be modified,
and derivative works of it may not be created, and it may not be
published except as an Internet-Draft.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
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."
This Internet-Draft will expire on December 7, 2011.
Copyright Notice
Copyright (c) 2011 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
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to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Protocol Requirements . . . . . . . . . . . . . . . . . . . . 4
3.1. Connection Management Requirements . . . . . . . . . . . . 4
3.2. Signaling Protocol Requirements . . . . . . . . . . . . . 5
3.2.1. Client Application SIP Requirements . . . . . . . . . 5
3.2.2. Client Application Optional SIP Support . . . . . . . 5
3.2.3. Required SIP Methods . . . . . . . . . . . . . . . . . 6
3.2.4. Multipart SIP Message Requirements . . . . . . . . . . 6
3.2.5. SIP Identity Requirements . . . . . . . . . . . . . . 6
3.2.6. SIP Network Address Traversal . . . . . . . . . . . . 7
3.3. Codec Requirements . . . . . . . . . . . . . . . . . . . . 7
3.3.1. Audio Codec Requirements . . . . . . . . . . . . . . . 7
3.3.2. Video Codec Requirements . . . . . . . . . . . . . . . 8
3.4. Real-Time Media Transport Requirements . . . . . . . . . . 8
3.4.1. RTP Profile . . . . . . . . . . . . . . . . . . . . . 9
3.4.2. RTP Optimizations . . . . . . . . . . . . . . . . . . 9
3.4.3. RTP Extensions . . . . . . . . . . . . . . . . . . . . 11
3.4.4. RTP Transport Robustness . . . . . . . . . . . . . . . 12
3.4.5. RTP Rate Control . . . . . . . . . . . . . . . . . . . 13
3.5. Non-Media Data Transport Requirements . . . . . . . . . . 13
3.5.1. Non-Media Data In RTP . . . . . . . . . . . . . . . . 14
3.5.2. Transporting Non-Media Data . . . . . . . . . . . . . 14
4. API Requirements . . . . . . . . . . . . . . . . . . . . . . . 16
5. Legacy VoIP Interoperability . . . . . . . . . . . . . . . . . 16
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 17
7. Security Considerations . . . . . . . . . . . . . . . . . . . 17
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 17
9. Normative References . . . . . . . . . . . . . . . . . . . . . 17
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 20
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1. Introduction
The Internet was, from very early in its lifetime, considered a
possible vehicle for the deployment of real-time, interactive
applications - with the most easily imaginable being audio
conversations (aka "Internet telephony") and videoconferencing.
The first attempts to build this were dependent on special networks,
special hardware and custom-built software, often at very high prices
or at low quality, placing great demands on the infrastructure.
As the available bandwidth has increased, and as processors and other
hardware has become ever faster, the barriers to participation have
decreased, and it is possible to deliver a satisfactory experience on
commonly available computing hardware.
Still, there are a number of barriers to the ability to communicate
universally - one of these is that there are, as of yet, no single
set of communication protocols that all agree should be made
available for communication; another is the sheer lack of universal
identification systems (such as is served by telephone numbers or
email addresses in other communications systems).
Development of "The Universal Solution" has proved hard, however, for
all the usual reasons. This memo aims to take a more building-block-
oriented approach, and try to find consensus on a set of substrate
components that we think will be useful in any real-time
communications systems.
The last few years have also seen a new platform rise for deployment
of services: The browser-embedded application, or "Web application".
It turns out that as long as the browser platform has the necessary
interfaces, it is possible to deliver almost any kind of service on
it.
Traditionally, these interfaces have been delivered by plugins, which
had to be downloaded and installed separately from the browser; in
the development of HTML5, much promise is seen by the possibility of
making those interfaces available in a standardized way within the
browser.
Other efforts, for instance the W3C Web Applications and Device API
working groups, focus on making standardized APIs and interfaces
available, within or alongside the HTML5 effort, for those functions;
this memo concentrates on specifying the protocols and subprotocols
that are needed to specify the interactions that happen across the
network.
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2. Terminology
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 [RFC2119].
3. Protocol Requirements
The section defines the set of protocols and selected subset profiles
of these protocols that RTC-WEB client applications will need to
implement. This set of protocols forms the requirements for the
controlling APIs in [Section 4]. At a high level this section is
split into five subsections that address requirements for RTC-WEB
client application: connection management, signaling protocols, codec
requirements, transports for real time media such as audio and video
and transports for non media data .
3.1. Connection Management Requirements
It is quite probable that many RTC-WEB client applications, such as
web browsers will be deployed behind a NAT. To set up secure data
plane sessions, all RTC-WEB client application implementations will
use ICE [RFC5245] or ICE-Lite Section 2.7 of [RFC5245]. ICE is
leveraged here to address the security concerns discussed in
[section] Section 7.
There are two deployment scenarios for RTC-WEB client applications.
The first scenario is when applications are deployed behind NAT and
have to worry about NAT traversal. The second scenario is when the
application is not behind a NAT, such as an RTC-WEB application that
is always connected to the public Internet. As stated in section 2.7
of [RFC5245], ICE requires that both endpoints to support it in order
for ICE to be used on a call.
With regards to RTC-WEB client applications, all applications that
are deployed behind a NAT or do not have a public IP address are
REQUIRED to support ICE [RFC5245], applications that are not behind a
NAT and have a public IP address are REQUIRED to support ICE-Lite and
MAY fully support ICE. RTC-WEB client applications that fully
support ICE are REQUIRED to support AGGRESSIVE NOMINATION, and MAY
support REGULAR NOMINATION.
Implicit to supporting ICE, all RTC-WEB client applications are
REQUIRED to implement Simple Traversal of User Datagram Protocol
(UDP) Through Network Address Translators (NATs) (STUN) [RFC3489] and
Traversal Using Relays around NAT (TURN) [RFC5766].
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[Open Issue: there is a strong interest to define a TURN-like
protocol that looks like HTTP to intermediaries, so that media can be
tunneled over HTTP. Should this be done?]
3.2. Signaling Protocol Requirements
This section covers the signaling protocol to be used by RTC-WEB
applications. To ensure interoperability not just between RTC-WEB
applications, but with legacy IPPBX phone systems as well, a small
subset of SIP will be REQUIRED for all RTC-WEB client application
implementations. In addition to the subset of SIP specification
[RFC3261], RTC-WEB client application implementations will be
REQUIRED to support DNS resolutions as specified in [RFC3263] and the
offer/answer model with SDP as specified in [RFC3264].
3.2.1. Client Application SIP Requirements
This section focuses on the subset of SIP functionality that will
exist within all RTC-WEB client applications. The following User
Agent Client (UAC) subset of the SIP specification [RFC3261] is
REQUIRED.
o General User Agent Behavior - [Section 8]
o Registration - [Section 10]
o Client Transaction - [Section 17.1]
o Canceling a Request - [Section 9.1]
o Terminating a Session - [Section 15.1], [Section 15.1.1]
o 3.XX Redirect Responses - [Section 8.1.3.4]
o TLS - [Section 26.3.1]
o Outbound Proxy
3.2.2. Client Application Optional SIP Support
In the SIP specification [RFC3261], the SIP features listed below are
required for all UAC implementations. RTC-WEB client applications
are not a fully featured SIP UAC and will only be implementing a
subset of the SIP specification. Thusly, unlike SIP UACs, the
following list of SIP features is to be considered OPTIONAL for RTC-
WEB client application implementations.
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o INVITEs without an offer
o re-INVITEs - [Section 14.1]
o forking - [Section 19.3]
o S/MIME - [Section 23]
o SIPS URI Scheme - [Section 26.2.2]
3.2.3. Required SIP Methods
This section outlines the REQUIRED SIP methods for all RTC-WEB client
applications.
o INVITE - [RFC3261] [Section 13]
o REGISTER - [RFC3261] [Section 10]
o ACK - [RFC3261] [Section 13.2]
o CANCEL - [RFC3261] [Section 13.2]
o BYE - [RFC3261] [Section 13.2]
o UPDATE - [RFC3311]
3.2.4. Multipart SIP Message Requirements
For handling SIP messages RTC-WEB client applications are required to
implement the multipart MIME handling scheme as specified in
[RFC5621].
3.2.5. SIP Identity Requirements
Identity, for the purposes of this section, is defined as a SIP URI.
There are two areas concerning SIP identity this specification will
address.
The first area covers validation of the message originator. To
securely validate a the identity of a SIP message originator, all
RTC-WEB client applications are REQUIRED to implement the mechanism
specified in [RFC4474].
To support cases were the identify of a caller/callee may change,
such as when a call is parked and transferred from the original
callee to another party, all RTC-WEB client applications are REQUIRED
to implement the identity mechanism specified in [RFC4916].
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[RFC3261]implicitly REQUIRES the implementation of the UPDATE method
as specified in [RFC3311]
3.2.6. SIP Network Address Traversal
RTC-WEB client applications MUST support Network Address Translator
(NAT) traversal. This section will address SIP-related areas to
support NAT traversal.
As called for in [3.1] RTC-WEB client applications will implement
STUN. To support client-managed connections, STUN-based keep-alives
as specified in [RFC5626] are REQUIRED.
When SIP is used with UDP, responses to requests are returned to the
source address the request came from, and to the port written into
the topmost Via header field value of the request. This behavior is
not desirable when the RTC-WEB client application is behind a Network
Address Translator (NAT). To address UDP traversal problem the
"rport" extension as specified in [RFC3581] is REQUIRED.
3.3. Codec Requirements
This section covers the audio and video codec requirements for RTC-
WEB client applications. To ensure a baseline level of
interoperability between RTC-Web applications, a minimum set of
required codes is specified below. While this section specifies the
codecs that will be supported by all RTC-Web application
implementations, it leaves the question of supporting additional
codecs to the will of the implementer.
3.3.1. Audio Codec Requirements
RTC-WEB applications are REQUIRED to implement the following audio
codecs.
o PCMA/PCMU - see section 4.5.14 of [RFC3551]
o Telephone-event - [RFC4734]
o Opus [draft-ietf-codec-opus]
Implementations of the PCMU and PMCA codecs are REQUIRED to support 1
channel with a rate of 8000 and a ptime of 20.
The following codecs are OPTIONAL for RTC-WEB application
implementations.
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o G729
o G722
o G722.1
o G723
o AMR
o AMR-WB
o iLBC
o L16
[Open Issue: minimum profile and identifying any additional mandatory
to implement audio codecs.]
3.3.2. Video Codec Requirements
RTC-WEB applications are REQUIRED to implement the following video
codecs.
o VP8 [I-D.westin-payload-vp8]
The following codecs are OPTIONAL for RTC-WEB application
implementations.
o H.263
o H.264-AVC
o H.264-SVC
[Open Issue: For the mandatory to implement video codec(s) what is
the minimum profile?]
3.4. Real-Time Media Transport Requirements
This section defines the real-time media transport requirements for
RTC-Web client application implementation. This section breaks down
the RTC-WEB RTP requirements into several sections. The sections
cover the RTP requirements for: profile, optimizations, extensions,
transport robustness and rate control.
[OPEN ISSUE: identify missing requirements]
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3.4.1. RTP Profile
RTC-Web applications to will need to provide a secure, interoperable,
bandwidth friendly, media transport profile. The Secure Audio-visual
Profile Feedback (SAVPF) as defined in [RFC5124] will meet the needs
of RTC-Web applications by providing media encryption,
interoperability and a flexible, bandwidth conscious RTCP packet
transmission model. All RTC-Web applications are REQUIRED to
implement SAVPF. Requiring the implementation of SAVPF also means
that RTC-Web applications MUST implicitly support Audio-visual
Profile Feedback (AVPF) [RFC4585], Audio-visual Profile (AVP)
[RFC3551] and Secure Audio-visual Profile (SAVP) [RFC3711].
3.4.1.1. Profile Encryption Mechanism
SAVPF supports SRTP by providing media encryption, integrity
protection, replay protection and a limited form of source
authentication. Though the SAVPF profile does support secure media
transport, it does not specify an encryption keying mechanism. To
support keying for SRTP, WEB-RTC application implementors are
REQUIRED to implement DTLS-SRPT [RFC5764].
3.4.2. RTP Optimizations
This section describes the optimization requirements for RTP within
RTC-Web applications.
3.4.2.1. RTP and RTCP Multiplexing
Historically, RTP and RTCP have been run on separate UDP ports. With
the increased use of Network Address Port Translation (NAPT) so have
the problems increased for maintaining multiple, costly NAT bindings
for each UDP port. This dual UDP port paradigm also complicates
firewall administration, since multiple ports must be opened to allow
for RTP traffic. To reduce these costs and session setup times,
support for multiplexing RTP data packets and RTCP control packets on
a single port [RFC5761] is REQUIRED.
Note that the use of RTP and RTCP multiplexed on a single port
ensures that there is occasional traffic sent on that port, even if
there is no active media traffic. This may be useful to keep-alive
NAT bindings.
3.4.2.2. Reduced-Size RTCP
RTCP packets are usually sent as compound RTCP packets and [RFC3550]
demands that the RTCP compound packets always start with a Sender
Report (SR) or Receiver Report (RR) packet. The SR and RR packets
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provide reception quality statistics and increase the mean RTCP
packet size. Because the mean compound RTCP packet size is larger,
the frequency at which RTCP packets can be sent within the RTCP
bandwidth share decreases. The decreased transmission frequency
creates a performance bottleneck that is especially noticeable when
using frequent feedback messages.
As mentioned in section [Add ref] RTC-Web applications will be
required to implement SAVPF, which implicitly requires feedback.
[RFC5506] specifies how to reduce the mean RTCP message and allow for
more frequent feedback. Frequent feedback, in turn, is essential to
make real-time application quickly aware of changing network
conditions and allow them to adapt their transmission and encoding
behavior. Support for [RFC5506] is REQUIRED
3.4.2.3. Symmetric RTP/RTCP
RTP entities choose the RTP and RTCP transport addresses (IP
addresses and port numbers), to bind to and receive packets on.
However when sending RTP and RTCP packets, senders may use an IP
address or port number that is different than the one specified for
receiving packets. Using different transport addresses is
problematic with regards to NAT traversal. The NAT traversal problem
can be alleviated using symmetric RTP/RTCP [RFC4961]. Symmetric RTP/
RTCP requires that the transport addresses for sending and receiving
RTP/RTCP packets are identical. All RTC-WEB client applications are
REQUIRED to implement Symmetric RTP/RTCP [RFC4961].
3.4.2.4. CNAME Generation
The RTCP Canonical Name (CNAME) provides a persistent transport-level
identifier for an RTP endpoint. While the Synchronization Source
(SSRC) identifier for an RTP endpoint may change if a collision is
detected, or when the RTP application is restarted, it's RTCP CNAME
is meant to stay unchanged, so that RTP endpoints can be uniquely
identified and associated with their RTP media streams. For proper
functionality, RTCP CNAMEs should be unique within the participants
of an RTP session.
The RTP specification [RFC3550] includes guidelines for choosing a
unique RTP CNAME. These guidelines are not sufficient in the
presence of NAT devices or with regards to addressing privacy
concerns resulting from the long-term, persistent identifiers.
To address the shortcomings of CNAME selection in[RFC3550], it is
RECOMMENDED that RTP CNAME generation follows the approach specified
in section 5 of [RFC6222].
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For RTC-WEB client applications, such as a web browser, it may not be
possible to retrieve the EUI-64 identifier or the host system's MAC
address which is needed to fulfill the CNAME generation procedure
outlined in section 5 of [RFC6222]. As an alternative to the EUI-64/
MAC address, RTC-WEB client applications MAY generate and use a
random number for the unique CNAME generation procedure.
3.4.3. RTP Extensions
.This section describes the RTP extensions that could be very useful
within the RTC-WEB context.
3.4.3.1. Conferencing Extensions
RTC-Web applications will support conferencing capabilities. While
this document remains silent regarding what conferencing topology
should be supported for RTC-Web applications, the following section
will provide guidance around RTP extensions to support centralized
conferencing.
For more information on RTP conferencing topologies please refer to
[RFC5117]
3.4.3.1.1. FIR RTCP Feedback Message
The Full Intra Request (FIR) command and message are defined in
sections 3.5.1 and 4.3.1 of [RFC5104]. FIR messages will request
that the currently distributed session participants send new intra
coded pictures to the mixer. FIR is used when switching between
sources to ensure that the receivers can decode the video or other
predicted media encoding with long prediction chains. It is
RECOMMENDED that the FIR message is supported.
3.4.3.1.2. PLI RTCP Feedback Message
The Picture Loss Indicator (PLI) is defined in Section 6.3.1 of
[RFC4585]. PLI messages tell the encoder that a receiver has lost
the decoder context and would like it repaired. It is RECOMMENDED
that the PLI message is supported.
3.4.3.1.3. TMMBR RTCP Feedback Message
The Temporary Maximum Media Stream Bit Rate Request (TMMBR, "timber")
message is defined in sections 3.5.4 and 4.2.1 of [RFC5104]. A
receiver, translator, or mixer uses the TMMBR to request a sender to
limit the maximum bit rate for a media stream to, or below, the
provided value. An example of using TMMBR would be for an RTP mixer
to constrain the media sender's bit rate to fit within the lower bit
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rate range of other session participants. It is RECOMMENDED that the
TMMBR message be supported.
3.4.3.2. Header Extensions
This section describes the requirements for RTC-WEB RTP header
extensions. For all RTC-WEB RTP header extensions it is REQUIRED
that they are formatted and signaled according to the general
mechanism defined in [RFC5285].
[Open Issue: should any of the following headers be added to the
list:
o Transmission time offsets[RFC5450]
o Associating time-codes with RTP streams [RFC5484] [Remove?]
Open Issue: There is also ongoing work to define RTP header
extensions for providing audio levels:
o From the media sender to the mixer [I-D.ietf-avtext-client-to-
mixer-audio-level]
o From a mixer to a receiver [I-D.ietf-avtext-mixer-to-client-audio-
level]
Which, if any of the above should be required? optional?
]
3.4.3.2.1. Rapid Synchronization
Basic RTP session synchronization as described in [RFC3550] can be
slow. To improve synchronization performance and maintain relative
backwards compatibility it is RECOMMENDED that the rapid RTP
synchronization extensions described in [RFC6051] be implemented.
3.4.4. RTP Transport Robustness
This section identifies tools that can be used to add robustness to
the RTP flows. Adding robustness to the RTP flow can reduce packet
loss and thus have a positive impact upon media quality.
3.4.4.1. RTP Retransmission
The retransmission scheme in RTP allows for flexibility of
retransmissions. From the receiving side, only selected missing
packets can be requested. From the sending side, packets can be
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prioritized based upon the senders knowledge of the receiver's
missing packets. Support for RTP retransmission as defined by
[RFC4588] is RECOMMENDED.
[Open Issue: is [RFC4588] the way we want to tackle this issue?]
3.4.4.2. Forward Error Correction
[Open issue - should there be a FEC scheme recommendation?]
3.4.4.3. Multicast
RTC-WEB client applications support for multicast RTP is NOT
REQUIRED.
3.4.5. RTP Rate Control
[OPEN ISSUE - There are currently no available, standardized RTP rate
control mechanism that uses media adaptation. Having a mechanism in
place will be REQUIRED for RTC-WEB applications and which means there
is a need for the IETF to produce this specification.
A potential starting point for defining a solution is "RTP with TCP
Friendly Rate Control" [rtp-tfrc].]
3.5. Non-Media Data Transport Requirements
The RTC-WEB will enable for rich voice and video communications from
client applications, such as a web browser. One of the natural
extensions of the RTC-WEB and the work emerging from the HTML5
community is video games. Video games have a similar stringent real-
time requirement for exchanging non-media data types such as a
player's screen position.
The question of how best to handle non-media data types has been
raised. There have been proposals to address this problem. Common
to all proposals is how the data transport session is set up, using
ICE [RFC5245] in a similar manner to that of RTP [RFC3550]. The
proposals vary from once the session is set up; one proposal is just
to use a thin shim on top of UDP or DTLS to de-multiplex the packets
from other packets such as RTP on the same connection. Another
proposal is DTLS over DCCP over UDP with some appropriate congestion
control scheme chosen for DCCP. Lastly there has been a proposal to
define a data codec to carry the data in RTP.
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3.5.1. Non-Media Data In RTP
This section will answer the question regarding the addition of non-
media data types into an RTC-WEB client application initiated RTP
session.
RTP by design adheres to the application level framing architectural
principle. This principle requires that RTP payload formats be
specified. By requiring specific payload formats RTP provides a
mechanism to optimize the transmission of encoded media. Other than
this optimization there is no congestion control mechanisms for RTP.
This principle also implies that if a payload format cannot be
specified, as the case is with generic data, it breaks one of the
fundamental architectural principles of RTP and makes optimization
impossible. Given that the ability to optimize the transmission of
non-media data types is lost and there are no capabilities for
congestion control within RTP, it follows that there is no benefit to
using RTP instead of a more general data transport such as UDP.
Until non-media data payload formats are created, the use of RTP as a
non-media data transport SHALL NOT be used in conjunction with any
RTC-WEB client application implementation.
[Open issue: There has been mention of actually creating new payload
formats for non-media data types. If new payload formats are
actually created for specific types of non-media data, the
requirement above would still stand as the application level framing
principle would be preserved and the new formats would have to adhere
to the principle. Any new formats would be specified outside of this
document but referred to]
3.5.2. Transporting Non-Media Data
[OPEN issue: need further discussion around this area]
There have been some ideas proposed but nothing has emerged as the
dominant paradigm. The current thinking is that, for RTC-WEB client
applications, RTP is not an option for non-media data types that do
not have a payload format specification. Without a payload format
specification a workable solution would resemble something that
allows datagrams to be transmitted via a secure, congestion-
controlled, unreliable transport mechanism.
3.5.2.1. Proposed Solutions for Non-Media Data Transport
One of the current proposed solutions could meet the requirements for
a non-media data type transport for RTC-WEB client application is to
use a DCCP via the following specifications:
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o DCCP [RFC4340] using TCP [RFC4341] or TFRC [RFC4342] for
congestion control
o DTLS for DCCP [RFC5238]
o DCCP over UDP [draft-ietf-dccp-udpencap]
The maturity of available implementations of DCCP is of concern along
with the partiality of this proposed solution. Another way of
tackling the problem of non-media data transport is to push the
requirements into the RTC-WEB client application implementation.
The following is a proposed set of REQUIRED RTC-WEB client
application non-media data transport requirements.
o Support for a broad congestion control constraints are REQUIRED,
it is RECOMMENDED that implementors support either TFRC [RFC5348]
or TFRC-SP [RFC4828].
o Support for full congestion control is RECOMMENDED, the specific
implementation is left to the RTC-WEB client application
implementor.
o Support for DTLS over UDP is REQUIRED
As an example of how these proposed requirements could be implemented
within an RTC-WEB client application, lets explore a web browser-
based implementation. In this specific implementation, the web
browser would provide DTLS over UDP and implement a broad congestion
control solution such as TFRC or TFRC-SP. This implementation will
yield a coarse-grained congestion controlled non-media data transport
solution that is accessible via JavaScript API calls. These non-
media data transport capabilities would provide a flexible solution
for web developers to build a full congestion control solution into
their WEB-RTC client application.
[Open Issue: Given that there is no consensus with regards to a
transport solution, this topic needs further discussion.
Open Issue: Areas for further discussion:
o Unreliable datagram transport - should this be UDP, SCTP or DCCP?
o Congestion control mechanisms - DCCP and SCTP something else?
o Protection of data confidentiality and integrity - DTLS
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o Receiver consent mechanisms for data transmission
]
4. API Requirements
NOT Ready - need to decide on protocols first, API comes after that
RTP
The API needs to allow the DSCP REF for each RTP or media stream to
be set.
The API needs to allow the browser app to observer and control the
SSRC values in the RTP.
Codec
The API needs to support the following OPTIONAL codecs: H263-2000,
H264, H264-SVC, raw and VP8.
The API needs to support the following OPTIONAL codecs: G729, G722,
G7221, G723, AMR, AMR-WB, iLBC, L16 and opus.
5. Legacy VoIP Interoperability
There is no way to meet all the security requirements and maintain
comparability with all legacy VoIP equipment. This draft tries to
minimize the impedance mismatch. The requirements here would allow
interoperability with legacy VoIP equipment as long as that equipment
either directly supported, or was fronted by an SBC that supported,
the following: a CORS [W3C.WD-cors-20090317] extension for SIP, ICE
or ICE-Lite, the mandatory to implement codecs in [SECTION],
supported SIP invites containing an offer, and supported DTMF over
RTP with telephone events.
Of the items listed above, support for ICE-Lite has historically been
lacking in VoIP equipment, this is changing and ICE-Lite becoming
increasingly prevalent, particularly on devices designed to sit on
the edge of a domain and connect to remote user agents that may be
behind NATs. Given the increasing adoption of ICE-Lite, it could be
conjectured that a substantial fraction of VoIP equipment meets the
RTC-WEB interoperability list except for the CORS extensions.
For an edge device that was willing to receive SIP call from others,
implementing the CORS is pretty trivial. When the UAS receives a SIP
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options request with an Origin header, it checks whether the header
field value is on the white list, and if it is then the UAS copies
the value to the Access-Control-Allow-Origin header field value in
the response. For many situations the white list would be
everything, while for others it would be just the list of websites
that are expected to originate calls to this SIP device.
6. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
7. Security Considerations
Because there are a number of security issues, considerations and
requirements for RTC-WEB client applications there is a draft that
specifically addresses the RTC-WEB application security
considerations. This draft defers it's security considerations and
requirements to the security considerations for RTC-Web draft
[I-D.ekr-security-considerations-for-rtc-web].
8. Acknowledgements
Many thanks to Harald Alvestrand, Magnus Westerlund, Colin Perkins,
Joerg Ott for a signifcant amount of text and contributed ideas on
this topic.
9. Normative References
[I-D.ekr-security-considerations-for-rtc-web]
Rescorla, E., "Security Considerations for RTC-Web",
May 2011.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3263] Rosenberg, J. and H. Schulzrinne, "Session Initiation
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Protocol (SIP): Locating SIP Servers", RFC 3263,
June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3311] Rosenberg, J., "The Session Initiation Protocol (SIP)
UPDATE Method", RFC 3311, October 2002.
[RFC3489] Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy,
"STUN - Simple Traversal of User Datagram Protocol (UDP)
Through Network Address Translators (NATs)", RFC 3489,
March 2003.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3581] Rosenberg, J. and H. Schulzrinne, "An Extension to the
Session Initiation Protocol (SIP) for Symmetric Response
Routing", RFC 3581, August 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram
Congestion Control Protocol (DCCP)", RFC 4340, March 2006.
[RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion
Control Protocol (DCCP) Congestion Control ID 2: TCP-like
Congestion Control", RFC 4341, March 2006.
[RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for
Datagram Congestion Control Protocol (DCCP) Congestion
Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
March 2006.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474, August 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
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"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[RFC4734] Schulzrinne, H. and T. Taylor, "Definition of Events for
Modem, Fax, and Text Telephony Signals", RFC 4734,
December 2006.
[RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control
(TFRC): The Small-Packet (SP) Variant", RFC 4828,
April 2007.
[RFC4916] Elwell, J., "Connected Identity in the Session Initiation
Protocol (SIP)", RFC 4916, June 2007.
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
BCP 131, RFC 4961, July 2007.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
January 2008.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
[RFC5238] Phelan, T., "Datagram Transport Layer Security (DTLS) over
the Datagram Congestion Control Protocol (DCCP)",
RFC 5238, May 2008.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
April 2010.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
RFC 5348, September 2008.
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[RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in
RTP Streams", RFC 5450, March 2009.
[RFC5484] Singer, D., "Associating Time-Codes with RTP Streams",
RFC 5484, March 2009.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC5621] Camarillo, G., "Message Body Handling in the Session
Initiation Protocol (SIP)", RFC 5621, September 2009.
[RFC5626] Jennings, C., Mahy, R., and F. Audet, "Managing Client-
Initiated Connections in the Session Initiation Protocol
(SIP)", RFC 5626, October 2009.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, November 2010.
[RFC6222] Begen, A., Perkins, C., and D. Wing, "Guidelines for
Choosing RTP Control Protocol (RTCP) Canonical Names
(CNAMEs)", RFC 6222, April 2011.
Authors' Addresses
Cary Bran
Cisco
170 West Tasman Drive
San Jose, CA 95134
USA
Phone: +1 206 256-3502
Email: cbran@cisco.com
Bran & Jennings Expires December 7, 2011 [Page 20]
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Cullen Jennings
Cisco
170 West Tasman Drive
San Jose, CA 95134
USA
Phone: +1 408 421-9990
Email: fluffy@cisco.com
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