Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
draft-ietf-rtcweb-rtp-usage-02
The information below is for an old version of the document.
| Document | Type |
This is an older version of an Internet-Draft that was ultimately published as RFC 8834.
|
|
|---|---|---|---|
| Authors | Colin Perkins , Magnus Westerlund , Joerg Ott | ||
| Last updated | 2012-03-12 | ||
| Replaces | draft-perkins-rtcweb-rtp-usage | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
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| Additional resources | Mailing list discussion | ||
| Stream | WG state | WG Document | |
| Document shepherd | (None) | ||
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| Consensus boilerplate | Unknown | ||
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| Send notices to | (None) |
draft-ietf-rtcweb-rtp-usage-02
Network Working Group C. Perkins
Internet-Draft University of Glasgow
Intended status: Standards Track M. Westerlund
Expires: September 13, 2012 Ericsson
J. Ott
Aalto University
March 12, 2012
Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
draft-ietf-rtcweb-rtp-usage-02
Abstract
The Web Real-Time Communication (WebRTC) framework provides support
for direct interactive rich communication using audio, video, text,
collaboration, games, etc. between two peers' web-browsers. This
memo describes the media transport aspects of the WebRTC framework.
It specifies how the Real-time Transport Protocol (RTP) is used in
the WebRTC context, and gives requirements for which RTP features,
profiles, and extensions need to be supported.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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 September 13, 2012.
Copyright Notice
Copyright (c) 2012 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
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carefully, as they describe your rights and restrictions with respect
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.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Discussion and Rationale . . . . . . . . . . . . . . . . . . . 4
3.1. Supported RTP Topologies . . . . . . . . . . . . . . . . . 4
3.2. Signalling Requirements . . . . . . . . . . . . . . . . . 7
4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 8
4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . . 8
4.2. Choice of RTP Profile . . . . . . . . . . . . . . . . . . 8
4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 9
4.4. RTP Session Multiplexing . . . . . . . . . . . . . . . . . 10
4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 10
4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10
4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 11
4.8. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11
5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 12
5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 12
5.1.1. Full Intra Request . . . . . . . . . . . . . . . . . . 13
5.1.2. Picture Loss Indication . . . . . . . . . . . . . . . 13
5.1.3. Slice Loss Indication . . . . . . . . . . . . . . . . 13
5.1.4. Reference Picture Selection Indication . . . . . . . . 13
5.1.5. Temporary Maximum Media Stream Bit Rate Request . . . 13
5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 14
5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 14
5.2.2. Client to Mixer Audio Level . . . . . . . . . . . . . 14
5.2.3. Mixer to Client Audio Level . . . . . . . . . . . . . 15
6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 15
6.1. Retransmission . . . . . . . . . . . . . . . . . . . . . . 15
6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . . 16
6.2.1. Basic Redundancy . . . . . . . . . . . . . . . . . . . 17
6.2.2. Block Based FEC . . . . . . . . . . . . . . . . . . . 18
6.2.3. Recommendations for FEC . . . . . . . . . . . . . . . 19
7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . . 19
7.1. Congestion Control Requirements . . . . . . . . . . . . . 20
7.2. RTCP Limiations . . . . . . . . . . . . . . . . . . . . . 20
7.3. Legacy Interop Limitations . . . . . . . . . . . . . . . . 21
8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 22
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 22
10. Security Considerations . . . . . . . . . . . . . . . . . . . 22
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 23
12.1. Normative References . . . . . . . . . . . . . . . . . . . 23
12.2. Informative References . . . . . . . . . . . . . . . . . . 26
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 27
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1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] provides a framework
for delivery of audio and video teleconferencing data and other real-
time media applications. Previous work has defined the RTP protocol,
plus numerous profiles, payload formats, and other extensions. When
combined with appropriate signalling, these form the basis for many
teleconferencing systems.
The Web Real-Time communication (WebRTC) framework is a new activity
that provides support for direct, interactive, and rich communication
using audio, video, collaboration, games, etc., between two peers'
web-browsers. This memo describes how the RTP framework is used in
the WebRTC context. It proposes a baseline set of RTP features that
should be implemented by all WebRTC-aware browsers, along with some
suggested extensions for enhanced functionality.
The WebRTC overview [I-D.ietf-rtcweb-overview] outlines the complete
WebRTC framework, of which this memo is a part.
2. Terminology
This memo is structured into different topics. For each topic, one
or several recommendations from the authors are given. When it comes
to the importance of extensions, or the need for implementation
support, we use three requirement levels to indicate the importance
of the feature to the WebRTC specification:
REQUIRED: Functionality that is absolutely needed to make the WebRTC
solution work well, or functionality of low complexity that
provides high value.
RECOMMENDED: Should be included as its brings significant benefit,
but the solution can potentially work without it.
OPTIONAL: Something that is useful in some cases, but not always a
benefit.
3. Discussion and Rationale
3.1. Supported RTP Topologies
RTP supports both unicast and group communication, with participants
being connected using wide range of transport-layer topologies. Some
of these topologies involve only the end-points, while others use RTP
translators and mixers to provide in-network processing. Properties
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of some RTP topologies are discussed in [RFC5117], and we further
describe those expected to be useful for WebRTC in the following.
+---+ +---+
| A |<------->| B |
+---+ +---+
Figure 1: Point to Point
The point to point topology (Figure 1) is going to be very common in
any single user to single user applications.
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 2: Multi-unicast
For small multiparty sessions it is practical enough to create RTP
sessions by letting every participant send individual unicast RTP/UDP
flows to each of the other participants. This is called multi-
unicast (Figure 2), and is unfortunately not discussed in the RTP
Topologies [RFC5117]. This topology has the benefit of not requiring
central nodes. The downside is that it increases the used bandwidth
at each sender by requiring one copy of the media streams for each
participant that are part of the same session beyond the sender
itself. Thus this is limited to scenarios with few end-points unless
the media is very low bandwidth.
This topology may be implemented as a single RTP session, spanning
multiple peer to peer transport layer connections, or as several
pairwise RTP sessions, one between each pair of peers. The later
approach simplifies rate adaptation, but reduces the effectiveness of
RTCP for debugging purposes, by limiting the ability to send third-
party RTCP reports.
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+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 3: RTP Mixer with Only Unicast Paths
An RTP mixer (Figure 3) is a centralised point that selects or mixes
content in a conference to optimise the RTP session so that each end-
point only needs connect to one entity, the mixer. The mixer also
reduces the bit-rate needs as the media sent from the mixer to the
end-point can be optimised in different ways. These optimisations
include methods like only choosing media from the currently most
active speaker or mixing together audio so that only one audio stream
is required in stead of 3 in the depicted scenario. The downside of
the mixer is that someone is required to provide the actual mixer.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 4: RTP Translator (Relay) with Only Unicast Paths
If one wants a less complex central node it is possible to use an
relay (called an Transport Translator) (Figure 4) that takes on the
role of forwarding the media to the other end-points but doesn't
perform any media processing. It simply forwards the media from all
other to all the other. Thus one endpoint A will only need to send a
media once to the relay, but it will still receive 3 RTP streams with
the media if B, C and D all currently transmits.
+------------+
| |
+---+ | | +---+
| A |<---->| Translator |<---->| B |
+---+ | | +---+
| |
+------------+
Figure 5: Translator towards Legacy end-point
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To support legacy end-point (B) that don't fulfil the requirements of
WebRTC it is possible to insert a Translator (Figure 5) that takes on
the role to ensure that from A's perspective B looks like a fully
compliant end-point. Thus it is the combination of the Translator
and B that looks like the end-point B. The intention is that the
presence of the translator is transparent to A, however it is not
certain that is possible. Thus this case is include so that it can
be discussed if any mechanism specified to be used for WebRTC results
in such issues and how to handle them.
3.2. Signalling Requirements
RTP is built with the assumption of an external signalling channel
that can be used to configure the RTP sessions and their features.
The basic configuration of an RTP session consists of the following
parameters:
RTP Profile: The name of the RTP profile to be used in session. The
RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate
on basic level, as can their secure variants RTP/SAVP [RFC3711]
and RTP/SAVPF [RFC5124]. The secure variants of the profiles do
not directly interoperate with the non-secure variants, due to the
presence of additional header fields in addition to any
cryptographic transformation of the packet content.
Transport Information: Source and destination address(s) and ports
for RTP and RTCP MUST be signalled for each RTP session. If RTP
and RTCP multiplexing [RFC5761] is to be used, such that a single
port is used for RTP and RTCP flows, this MUST be signalled (see
Section 4.5). If several RTP sessions are to be multiplexed onto
a single transport layer flow, this MUST also be signalled (see
Section 4.4).
RTP Payload Types, media formats, and media format
parameters: The mapping between media type names (and hence the RTP
payload formats to be used) and the RTP payload type numbers must
be signalled. Each media type may also have a number of media
type parameters that must also be signalled to configure the codec
and RTP payload format (the "a=fmtp:" line from SDP).
RTP Extensions: The RTP extensions one intends to use need to be
agreed upon, including any parameters for each respective
extension. At the very least, this will help avoiding using
bandwidth for features that the other end-point will ignore. But
for certain mechanisms there is requirement for this to happen as
interoperability failure otherwise happens.
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RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the
end-points will be necessary, as described in "Session Description
Protocol (SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP)
Bandwidth" [RFC3556], or something semantically equivalent. This
also ensures that the end-points have a common view of the RTCP
bandwidth, this is important as too different view of the
bandwidths may lead to failure to interoperate.
These parameters are often expressed in SDP messages conveyed within
an offer/answer exchange. RTP does not depend on SDP or on the
offer/answer model, but does require all the necessary parameters to
be agreed somehow, and provided to the RTP implementation. We note
that in RTCWEB context it will depend on the signalling model and API
how these parameters need to be configured but they will be need to
either set in the API or explicitly signalled between the peers.
4. WebRTC Use of RTP: Core Protocols
RTP and RTCP are flexible and extensible protocols that allow, on the
one hand, choosing from a variety of building blocks and combining
those to meet application needs, but on the other hand, offer the
ability to create extensions where existing mechanisms are not
sufficient. This memo requires a number of RTP and RTCP extensions
that have been shown to be provide important functionality in the
WebRTC context be implemented. It is possible that future extensions
will be needed: several documents provide guidelines for the use and
extension of RTP and RTCP, including Guidelines for Writers of RTP
Payload Format Specifications [RFC2736] and Guidelines for Extending
the RTP Control Protocol [RFC5968], and should be consulted before
extending this memo.
The following describe core features of RTP, and core extensions,
that must be supported in all WebRTC implementations.
4.1. RTP and RTCP
The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be
implemented as the media transport protocol for WebRTC. RTP itself
comprises two parts: the RTP data transfer protocol, and the RTP
control protocol (RTCP). RTCP is a fundamental and integral part of
the RTP protocol, and is REQUIRED to be implemented.
4.2. Choice of RTP Profile
The complete specification of RTP for a particular application domain
requires the choice of an RTP Profile. For WebRTC use, the "Extended
Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-
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Based Feedback (RTP/SAVPF)" [RFC5124] is REQUIRED to be implemented.
This builds on the basic RTP/AVP profile [RFC3551], the RTP/AVPF
feedback profile [RFC4585], and the secure RTP/SAVP profile
[RFC3711].
The RTP/AVPF part of RTP/SAVPF is required to get the improved RTCP
timer model, that allows more flexible transmission of RTCP packets
in response to events, rather than strictly according to bandwidth.
This also saves RTCP bandwidth and will commonly only use the full
amount when there is a lot of events on which to send feedback. This
functionality is needed to make use of the RTP conferencing
extensions discussed in Section 5.1. The improved RTCP timer model
defined by RTP/AVPF is backwards compatible with legacy systems that
implement only the RTP/AVP profile given some constraints on
parameter configuration such as RTCP banwidth and "trr-int".
The RTP/SAVP part of RTP/SAVPF is for support for Secure RTP (SRTP)
[RFC3711]. This provides media encryption, integrity protection,
replay protection and a limited form of source authentication.
(tbd: There is ongoing discussion on what keying mechanism is to be
required, what are the mandated cryptographic transforms, and whether
fallback to plain RTP should be supported. This section needs to be
updated based on the results of that discussion.)
4.3. Choice of RTP Payload Formats
(tbd: say something about the choice of RTP Payload Format for
WebRTC. If there is a mandatory to implement set of codecs, this
should reference them. In any case, it should reference a discussion
of signalling for the choice of codec, once that discussion reaches
closure.)
Endpoints may signal support for multiple media formats, or multiple
configurations of a single format, provided each uses a different RTP
payload type number. An endpoint that has signalled it's support for
multiple formats is REQUIRED to accept data in any of those formats
at any time, unless it has previously signalled limitations on it's
decoding capability (this is modified if several RTP sessions are to
be multiplexed onto one transport layer connection, such that an
endpoint must be prepared for a source to switch between formats of
the same media type at any time; see Section 4.4). To support rapid
rate adaptation, RTP does not require advance signalling for changes
between payload formats that were signalled during session setup.
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4.4. RTP Session Multiplexing
An association amongst a set of participants communicating with RTP
is known as an RTP session. A participant may be involved in
multiple RTP sessions at the same time. In a multimedia session,
each medium has typically been carried in a separate RTP session with
its own RTCP packets (i.e., one RTP session for the audio, with a
separate RTP session running on a different transport connection for
the video; if SDP is used, this corresponds to one RTP session for
each "m=" line in the SDP). WebRTC implementations of RTP are
REQUIRED to implement support of multimedia sessions in this way, for
compatibility with legacy systems.
In today's networks, however, with the prolific use of Network
Address Translators (NAT) and Firewalls (FW), there is a desire to
reduce the number of transport layer ports used by real-time media
applications using RTP by combining multimedia traffic in a single
RTP session. (Details of how this is to be done are tbd, but see
[I-D.lennox-rtcweb-rtp-media-type-mux],
[I-D.holmberg-mmusic-sdp-bundle-negotiation] and
[I-D.westerlund-avtcore-multiplex-architecture].) WebRTC
implementations of RTP are REQUIRED to support multiplexing of
multimedia sessions onto a single RTP session according to (tbd). If
such RTP session multiplexing is to be used, this MUST be negotiated
during the signalling phase.
4.5. 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) this has
become problematic, since maintaining multiple NAT bindings can be
costly. It also complicates firewall administration, since multiple
ports must be opened to allow 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 and is recommend method for application level keep-
alives of RTP sessions [RFC6263].
4.6. Reduced Size RTCP
RTCP packets are usually sent as compound RTCP packets, and RFC 3550
requires that those compound packets start with an SR or RR packet.
When using frequent feedback messages, these general statistics are
not needed in every packet and unnecessarily increase the mean RTCP
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packet size. This can limit the frequency at which RTCP packets can
be sent within the RTCP bandwidth share.
To avoid this problem, [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 behaviour. Support for RFC5506 is
REQUIRED.
4.7. Symmetric RTP/RTCP
RTP entities choose the RTP and RTCP transport addresses, i.e., IP
addresses and port numbers, to receive packets on and bind their
respective sockets to those. When sending RTP packets, however, they
may use a different IP address or port number for RTP, RTCP, or both;
e.g., when using a different socket instance for sending and for
receiving. Symmetric RTP/RTCP requires that the IP address and port
number for sending and receiving RTP/RTCP packets are identical.
The reasons for using symmetric RTP is primarily to avoid issues with
NAT and Firewalls by ensuring that the flow is actually bi-
directional and thus kept alive and registered as flow the intended
recipient actually wants. In addition it saves resources in the form
of ports at the end-points, but also in the network as NAT mappings
or firewall state is not unnecessary bloated. Also the number of QoS
state are reduced.
Using Symmetric RTP and RTCP [RFC4961] is REQUIRED.
4.8. Generation of the RTCP Canonical Name (CNAME)
The RTCP Canonical Name (CNAME) provides a persistent transport-level
identifier for an RTP endpoint. While the Synchronisation 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 among the participants of
an RTP session.
The RTP specification [RFC3550] includes guidelines for choosing a
unique RTP CNAME, but these are not sufficient in the presence of NAT
devices. In addition, some may find long-term persistent identifiers
problematic from a privacy viewpoint. Accordingly, support for
generating a short-term persistent RTCP CNAMEs following method (b)
as specified in Section 4.2 of "Guidelines for Choosing RTP Control
Protocol (RTCP) Canonical Names (CNAMEs)" [RFC6222] is REQUIRED,
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since this addresses both concerns.
5. WebRTC Use of RTP: Extensions
There are a number of RTP extensions that could be very useful in the
WebRTC context. One set is related to conferencing, others are more
generic in nature. None of these are mandatory to implement, but
they are expected to be generally useful.
5.1. Conferencing Extensions
RTP is inherently a group communication protocol. Groups can be
implemented using a centralised server, multi-unicast, or IP
multicast. While IP multicast was popular in early deployments, in
today's practice, overlay-based conferencing dominates, typically
using one or more central servers to connect endpoints in a star or
flat tree topology. These central servers can be implemented in a
number of ways [RFC5117], of which the following are the most common:
1. RTP Translator (Relay) with Only Unicast Paths ([RFC5117],
section 3.3)
2. RTP Mixer with Only Unicast Paths ([RFC5117], section 3.4)
3. Point to Multipoint Using a Video Switching MCU ([RFC5117],
section 3.5)
4. Point to Multipoint Using Content Modifying MCUs ([RFC5117],
section 3.6)
As discussed in [RFC5117] section 3.5, the use of a video switching
MCU makes the use of RTCP for congestion control, or any type of
quality reports, very problematic. Also, as discussed in [RFC5117]
section 3.6, the use of a content modifying MCU with RTCP termination
breaks RTP loop detection and removes the ability for receivers to
identify active senders. According only the first two options are
recommended.
RTP protocol extensions to be used with conferencing are included
because they are important in the context of centralised
conferencing, where one RTP Mixer (Conference Focus) receives a
participants media streams and distribute them to the other
participants. These messages are defined in the Extended RTP Profile
for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-
Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully
usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].
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5.1.1. Full Intra Request
The Full Intra Request is defined in Sections 3.5.1 and 4.3.1 of CCM
[RFC5104]. It is used to have the mixer request from a session
participants a new Intra picture. This 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 this feedback message is supported.
5.1.2. Picture Loss Indication
The Picture Loss Indication is defined in Section 6.3.1 of the RTP/
AVPF profile [RFC4585]. It is used by a receiver to tell the encoder
that it lost the decoder context and would like to have it repaired
somehow. This is semantically different from the Full Intra Request
above. It is RECOMMENDED that this feedback message is supported as
a loss tolerance mechanism.
5.1.3. Slice Loss Indication
The Slice Loss Indicator is defined in Section 6.3.2 of the RTP/AVPF
profile [RFC4585]. It is used by a receiver to tell the encoder that
it has detected the loss or corruption of one or more consecutive
macroblocks, and would like to have these repaired somehow. The use
of this feedback message is OPTIONAL as a loss tolerance mechanism.
5.1.4. Reference Picture Selection Indication
Reference Picture Selection Indication (RPSI) is defined in Section
6.3.3 of the RTP/AVPF profile [RFC4585]. Some video coding standards
allow the use of older reference pictures than the most recent one
for predictive coding. If such a codec is in used, and if the
encoder has learned about a loss of encoder-decoder synchronicity, a
known-as-correct reference picture can be used for future coding.
The RPSI message allows this to be signalled. The use of this RTCP
feedback message is OPTIONAL as a loss tolerance mechanism.
5.1.5. Temporary Maximum Media Stream Bit Rate Request
This feedback message is defined in Section 3.5.4 and 4.2.1 in CCM
[RFC5104]. This message and its notification message is used by a
media receiver, to inform the sending party that there is a current
limitation on the amount of bandwidth available to this receiver.
This can be for various reasons, and can for example be used by an
RTP mixer to limit the media sender being forwarded by the mixer
(without doing media transcoding) to fit the bottlenecks existing
towards the other session participants. It is RECOMMENDED that this
feedback message is supported.
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5.2. Header Extensions
The RTP specification [RFC3550] provides a capability to extend the
RTP header with in-band data, but the format and semantics of the
extensions are poorly specified. The use of header extensions is
OPTIONAL, but if are used, it is REQUIRED that they be formatted and
signalled according to the general mechanism of RTP header extensions
defined in [RFC5285].
As noted in [RFC5285], the requirement from the RTP specification
that header extensions are "designed so that the header extension may
be ignored" [RFC3550] stands. To be specific, header extensions must
only be used for data that can safely be ignored by the recipient
without affecting interoperability, and must not be used when the
presence of the extension has changed the form or nature of the rest
of the packet in a way that is not compatible with the way the stream
is signalled (e.g., as defined by the payload type). Valid examples
might include metadata that is additional to the usual RTP
information.
The RTP rapid synchronisation header extension [RFC6051] is
recommended, as discussed in Section 5.2.1 we also recommend the
client to mixer audio level [RFC6464], and consider the mixer to
client audio level [RFC6465] as optional feature.
If the client-to-mixer or mixer-to-client audio level indications are
in use in SRTP encrypted sessions, it is REQUIRED that the extensions
are encrypted according to
[I-D.ietf-avtcore-srtp-encrypted-header-ext] since the information
contained in these header extensions may be considered sensitive.
5.2.1. Rapid Synchronisation
Many RTP sessions require synchronisation between audio, video, and
other content. This synchronisation is performed by receivers, using
information contained in RTCP SR packets, as described in the RTP
specification [RFC3550]. This basic mechanism can be slow, however,
so it is RECOMMENDED that the rapid RTP synchronisation extensions
described in [RFC6051] be implemented. The rapid synchronisation
extensions use the general RTP header extension mechanism [RFC5285],
which requires signalling, but are otherwise backwards compatible.
5.2.2. Client to Mixer Audio Level
The Client to Mixer Audio Level [RFC6464] is an RTP header extension
used by a client to inform a mixer about the level of audio activity
in the packet the header is attached to. This enables a central node
to make mixing or selection decisions without decoding or detailed
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inspection of the payload. Thus reducing the needed complexity in
some types of central RTP nodes.
The Client to Mixer Audio Level [RFC6464] is RECOMMENDED to be
implemented.
5.2.3. Mixer to Client Audio Level
The Mixer to Client Audio Level header extension [RFC6465] provides
the client with the audio level of the different sources mixed into a
common mix from the RTP mixer. Thus enabling a user interface to
indicate the relative activity level of a session participant, rather
than just being included or not based on the CSRC field. This is a
pure optimisations of non critical functions and thus optional
functionality.
The Mixer to Client Audio Level [RFC6465] is OPTIONAL to implement.
6. WebRTC Use of RTP: Improving Transport Robustness
There are some tools that can make RTP flows robust against Packet
loss and reduce the impact on media quality. However they all add
extra bits compared to a non-robust stream. These extra bits needs
to be considered and the aggregate bit-rate needs to be rate
controlled. Thus improving robustness might require a lower base
encoding quality but has the potential to give that quality with
fewer errors in it.
6.1. Retransmission
Support for RTP retransmission as defined by "RTP Retransmission
Payload Format" [RFC4588] is RECOMMENDED.
The retransmission scheme in RTP allows flexible application of
retransmissions. Only selected missing packets can be requested by
the receiver. It also allows for the sender to prioritise between
missing packets based on senders knowledge about their content.
Compared to TCP, RTP retransmission also allows one to give up on a
packet that despite retransmission(s) still has not been received
within a time window.
"WebRTC Media Transport Requirements" [I-D.cbran-rtcweb-data] raises
two issues that they think makes RTP Retransmission unsuitable for
RTCWEB. We here consider these issues and explain why they are in
fact not a reason to exclude RTP retransmission from the tool box
available to RTCWEB media sessions.
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The additional latency added by [RFC4588] will exceed the latency
threshold for interactive voice and video: RTP Retransmission will
require at least one round trip time for a retransmission request
and repair packet to arrive. Thus the general suitability of
using retransmissions will depend on the actual network path
latency between the end-points. In many of the actual usages the
latency between two end-points will be low enough for RTP
retransmission to be effective. Interactive communication with
end-to-end delays of 400 ms still provide a fair quality. Even
removing half of that in end-point delays allows functional
retransmission between end-points on the same continent. In
addition, some applications may accept temporary delay spikes to
allow for retransmission of crucial codec information such an
parameter sets, intra picture etc, rather than getting no media at
all.
The undesirable increase in packet transmission at the point when
congestion occurs: Congestion loss will impact the rate controls
view of available bit-rate for transmission. When using
retransmission one will have to prioritise between performing
retransmissions and the quality one can achieve with ones
adaptable codecs. In many use cases one prefer error free or low
rates of error with reduced base quality over high degrees of
error at a higher base quality.
The RTCWEB end-point implementations will need to both select when to
enable RTP retransmissions based on API settings and measurements of
the actual round trip time. In addition for each NACK request that a
media sender receives it will need to make a prioritisation based on
the importance of the requested media, the probability that the
packet will reach the receiver in time for being usable, the
consumption of available bit-rate and the impact of the media quality
for new encodings.
To conclude, the issues raised are implementation concerns that an
implementation needs to take into consideration, they are not
arguments against including a highly versatile and efficient packet
loss repair mechanism.
6.2. Forward Error Correction (FEC)
Support of some type of FEC to combat the effects of packet loss is
beneficial, but is heavily application dependent. However, some FEC
mechanisms are encumbered.
The main benefit from FEC is the relatively low additional delay
needed to protect against packet losses. The transmission of any
repair packets should preferably be done with a time delay that is
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just larger than any loss events normally encountered. That way the
repair packet isn't also lost in the same event as the source data.
The amount of repair packets needed varies depending on the amount
and pattern of packet loss to be recovered, and on the mechanism used
to derive repair data. The later choice also effects the the
additional delay required to both encode the repair packets and in
the receiver to be able to recover the lost packet(s).
6.2.1. Basic Redundancy
The method for providing basic redundancy is to simply retransmit a
some time earlier sent packet. This is relatively simple in theory,
i.e. one saves any outgoing source (original) packet in a buffer
marked with a timestamp of actual transmission, some X ms later one
transmit this packet again. Where X is selected to be longer than
the common loss events. Thus any loss events shorter than X can be
recovered assuming that one doesn't get an another loss event before
all the packets lost in the first event has been received.
The downside of basic redundancy is the overhead. To provide each
packet with once chance of recovery, then the transmission rate
increases with 100% as one needs to send each packet twice. It is
possible to only redundantly send really important packets thus
reducing the overhead below 100% for some other trade-off is
overhead.
In addition the basic retransmission of the same packet using the
same SSRC in the same RTP session is not possible in RTP context.
The reason is that one would then destroy the RTCP reporting if one
sends the same packet twice with the same sequence number. Thus one
needs more elaborate mechanisms.
RTP Payload Format Support: Some RTP payload format do support basic
redundancy within the RTP paylaod format itself. Examples are
AMR-WB [RFC4867] and G.719 [RFC5404].
RTP Payload for Redundant Audio Data: This audio and text redundancy
format defined in [RFC2198] allows for multiple levels of
redundancy with different delay in their transmissions, as long as
the source plus payload parts to be redundantly transmitted
together fits into one MTU. This should work fine for most
interactive audio and text use cases as both the codec bit-rates
and the framing intervals normally allow for this requirement to
hold. This payload format also don't increase the packet rate, as
original data and redundant data are sent together. This format
does not allow perfect recovery, only recovery of information
deemed necessary for audio, for example the sequence number of the
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original data is lost.
RTP Retransmission Format: The RTP Retransmission Payload format
[RFC4588] can be used to pro-actively send redundant packets using
either SSRC or session multiplexing. By using different SSRCs or
a different session for the redundant packets the RTCP receiver
reports will be correct. The retransmission payload format is
used to recover the packets original data thus enabling a perfect
recovery.
Duplication Grouping Semantics in the Session Description Protocol:
This [I-D.begen-mmusic-redundancy-grouping] is proposal for new
SDP signalling to indicate media stream duplication using
different RTP sessions, or different SSRCs to separate the source
and the redundant copy of the stream.
6.2.2. Block Based FEC
Block based redundancy collects a number of source packets into a
data block for processing. The processing results in some number of
repair packets that is then transmitted to the other end allowing the
receiver to attempt to recover some number of lost packets in the
block. The benefit of block based approaches is the overhead which
can be lower than 100% and still recover one or more lost source
packet from the block. The optimal block codes allows for each
received repair packet to repair a single loss within the block.
Thus 3 repair packets that are received should allow for any set of 3
packets within the block to be recovered. In reality one commonly
don't reach this level of performance for any block sizes and number
of repair packets, and taking the computational complexity into
account there are even more trade-offs to make among the codes.
One result of the block based approach is the extra delay, as one
needs to collect enough data together before being able to calculate
the repair packets. In addition sufficient amount of the block needs
to be received prior to recovery. Thus additional delay are added on
both sending and receiving side to ensure possibility to recover any
packet within the block.
The redundancy overhead and the transmission pattern of source and
repair data can be altered from block to block, thus allowing a
adaptive process adjusting to meet the actual amount of loss seen on
the network path and reported in RTCP.
The alternatives that exist for block based FEC with RTP are the
following:
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RTP Payload Format for Generic Forward Error Correction: This RTP
payload format [RFC5109] defines an XOR based recovery packet.
This is the simplest processing wise that an block based FEC
scheme can be. It also results in some limited properties, as
each repair packet can only repair a single loss. To handle
multiple close losses a scheme of hierarchical encodings are need.
Thus increasing the overhead significantly.
Forward Error Correction (FEC) Framework: This framework
[I-D.ietf-fecframe-framework] defines how not only RTP packets but
how arbitrary packet flows can be protected. Some solutions
produced or under development in FECFRAME WG are RTP specific.
There exist alternatives supporting block codes such as Reed-
Salomon and Raptor.
6.2.3. Recommendations for FEC
(tbd)
7. WebRTC Use of RTP: Rate Control and Media Adaptation
WebRTC will be used in very varied network environment with a
hetrogenous set of link technologies, including wired and wireless,
interconnecting peers at different topological locations resulting in
network paths with widely varying one way delays, bit-rate capacity,
load levels and traffic mixes. In addition individual end-points
will open one or more WebRTC sessions between one or more peers.
Each of these session may contain different mixes of media and data
flows. Assymetric usage of media bit-rates and number of media
streams is also to be expected. A single end-point may receive zero
to many simultanous media streams while itself transmitting one or
more streams.
The WebRTC application is very dependent from a quality perspective
on the media adapation working well so that an end-point doesn't
transmit significantly more than the path is capable of handling. If
it would, the result would be high levels of packet loss or delay
spikes causing media degradations.
WebRTC applications using more than a single media stream of any
media type or data flows has an additional concern. In this case the
different flows should try to avoid affecting each other negatively.
In addition in case there is a resource limiation, the available
resources needs to be shared. How to share them is something the
application should prioritize so that the limiation in quality or
capabilities are the ones that provide the least affect on the
application.
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This hetrogenous situation results in a requirement to have
functionality that adapts to the available capacity and that competes
fairly with other network flows. If it would not compete fairly
enough WebRTC could be used as an attack method for starving out
other traffic on specific links as long as the attacker is able to
create traffic across a specific link. This is not far-fetched for a
web-service capable of attracting large number of end-points and use
the service, combined with BGP routing state a server could pick
client pairs to drive traffic to specific paths.
The above estalish a clear need based on several reasons why there
need to be a well working media adaptation mechanism. This mechanism
also have a number of requirements on what services it should provide
and what performance it needs to provide.
The biggest issue is that there are no standardised and ready to use
mechanism that can simply be included in WebRTC Thus there will be
need for the IETF to produce such a specification. Therefore the
suggested way forward is to specify requirements on any solution for
the media adaptation. These requirements is for now proposed to be
documented in this specification. In addition a proposed detailed
solution will be developed, but is expected to take longer time to
finalize than this document.
7.1. Congestion Control Requirements
Requirements for congestion control of WebRTC sessions are discussed
in [I-D.jesup-rtp-congestion-reqs].
Implementations are REQUIRED to implement the RTP circuit breakers
described in [I-D.perkins-avtcore-rtp-circuit-breakers].
7.2. RTCP Limiations
Experience with the congestion control algorithms of TCP [RFC5681],
TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown
that feedback on packet arrivals needs to be sent roughly once per
round trip time. We note that the capabilities of real-time media
traffic to adapt to changing path conditions may be less rapid than
for the elastic applications TCP was designed for, but frequent
feedback is still required to allow the congestion control algorithm
to track the path dynamics.
The total RTCP bandwidth is limited in its transmission rate to a
fraction of the RTP traffic (by default 5%). RTCP packets are larger
than, e.g., TCP ACKs (even when non-compound RTCP packets are used).
The media stream bit rate thus limits the maximum feedback rate as a
function of the mean RTCP packet size.
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Interactive communication may not be able to afford waiting for
packet losses to occur to indicate congestion, because an increase in
playout delay due to queuing (most prominent in wireless networks)
may easily lead to packets being dropped due to late arrival at the
receiver. Therefore, more sophisticated cues may need to be reported
-- to be defined in a suitable congestion control framework as noted
above -- which, in turn, increase the report size again. For
example, different RTCP XR report blocks (jointly) provide the
necessary details to implement a variety of congestion control
algorithms, but the (compound) report size grows quickly.
In group communication, the share of RTCP bandwidth needs to be
shared by all group members, reducing the capacity and thus the
reporting frequency per node.
Example: assuming 512 kbit/s video yields 3200 bytes/s RTCP
bandwidth, split across two entities in a point-to-point session. An
endpoint could thus send a report of 100 bytes about every 70ms or
for every other frame in a 30 fps video.
7.3. Legacy Interop Limitations
Congestion control interoperability with most type of legacy devices,
even using an translator could be difficult. There are numerous
reasons for this:
No RTCP Support: There exist legacy implementations that does not
even implement RTCP at all. Thus no feedback at all is provided.
RTP/AVP Minimal RTCP Interval of 5s: RTP [RFC3550] under the RTP/AVP
profile specifies a recommended minimal fixed interval of 5
seconds. Sending RTCP report blocks as seldom as 5 seconds makes
it very difficult for a sender to use these reports and react to
any congestion event.
RTP/AVP Scaled Minimal Interval: If a legacy device uses the scaled
minimal RTCP compound interval, the "RECOMMENDED value for the
reduced minimum in seconds is 360 divided by the session bandwidth
in kilobits/second" ([RFC3550], section 6.2). The minimal
interval drops below a second, still several times the RTT in
almost all paths in the Internet, when the session bandwidht
becomes 360 kbps. A session bandwidth of 1 Mbps still has a
minimal interval of 360 ms. Thus, with the exception for rather
high bandwidth sessions, getting frequent enough RTCP Report
Blocks to report on the order of the RTT is very difficult as long
as the legacy device uses the RTP/AVP profile.
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RTP/AVPF Supporting Legacy Device: If a legacy device supports RTP/
AVPF, then that enables negotation of important parameters for
frequent reporting, such as the "trr-int" parameter, and the
possibility that the end-point supports some useful feedback
format for congestion control purpose such as TMMBR [RFC5104].
It has been suggested on the RTCWEB mailing list that if
interoperating with really limited legacy devices an WebRTC end-point
may not send more than 64 kbps of media streams, to avoid it causing
massive congestion on most paths in the Internet when communicating
with a legacy node not providing sufficient feedback for effective
congestion control. This warrants further discussion as there is
clearly a number of link layers that don't even provide that amount
of bit-rate consistently, and that assumes no competing traffic.
8. WebRTC Use of RTP: Performance Monitoring
RTCP does contains a basic set of RTP flow monitoring points like
packet loss and jitter. There exist a number of extensions that
could be included in the set to be supported. However, in most cases
which RTP monitoring that is needed depends on the application, which
makes it difficult to select which to include when the set of
applications is very large.
Exposing some metrics in the WebRTC API should be considered allowing
the application to gather the measurements of interest. However,
security implications for the different data sets exposed will need
to be considered in this.
9. IANA Considerations
This memo makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
10. Security Considerations
RTP and its various extensions each have their own security
considerations. These should be taken into account when considering
the security properties of the complete suite. We currently don't
think this suite creates any additional security issues or
properties. The use of SRTP [RFC3711] will provide protection or
mitigation against all the fundamental issues by offering
confidentiality, integrity and partial source authentication. A
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mandatory to implement media security solution will be required to be
picked. We currently don't discuss the key-management aspect of SRTP
in this memo, that needs to be done taking the WebRTC communication
model into account.
The guidelines in [I-D.ietf-avtcore-srtp-vbr-audio] apply when using
variable bit rate (VBR) audio codecs, for example Opus or the Mixer
audio level header extensions.
Security considerations for the WebRTC work are discussed in
[I-D.ietf-rtcweb-security].
11. Acknowledgements
The authors would like to thank Harald Alvestrand, Cary Bran, and
Cullen Jennings for valuable feedback.
12. References
12.1. Normative References
[I-D.holmberg-mmusic-sdp-bundle-negotiation]
Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
Using Session Description Protocol (SDP) Port Numbers",
draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in
progress), October 2011.
[I-D.ietf-avtcore-srtp-encrypted-header-ext]
Lennox, J., "Encryption of Header Extensions in the Secure
Real-Time Transport Protocol (SRTP)",
draft-ietf-avtcore-srtp-encrypted-header-ext-00 (work in
progress), June 2011.
[I-D.ietf-avtcore-srtp-vbr-audio]
Perkins, C. and J. Valin, "Guidelines for the use of
Variable Bit Rate Audio with Secure RTP",
draft-ietf-avtcore-srtp-vbr-audio-03 (work in progress),
July 2011.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-02 (work
in progress), September 2011.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for RTC-Web",
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draft-ietf-rtcweb-security-01 (work in progress),
October 2011.
[I-D.jesup-rtp-congestion-reqs]
Jesup, R. and H. Alvestrand, "Congestion Control
Requirements For Real Time Media",
draft-jesup-rtp-congestion-reqs-00 (work in progress),
March 2012.
[I-D.lennox-rtcweb-rtp-media-type-mux]
Lennox, J. and J. Rosenberg, "Multiplexing Multiple Media
Types In a Single Real-Time Transport Protocol (RTP)
Session", draft-lennox-rtcweb-rtp-media-type-mux-00 (work
in progress), October 2011.
[I-D.perkins-avtcore-rtp-circuit-breakers]
Perkins, C. and V. Singh, "RTP Congestion Control: Circuit
Breakers for Unicast Sessions",
draft-perkins-avtcore-rtp-circuit-breakers-00 (work in
progress), March 2012.
[I-D.westerlund-avtcore-multiplex-architecture]
Westerlund, M., Burman, B., and C. Perkins, "RTP
Multiplexing Architecture",
draft-westerlund-avtcore-multiplex-architecture-00 (work
in progress), October 2011.
[RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP
Payload Format Specifications", BCP 36, RFC 2736,
December 1999.
[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.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth",
RFC 3556, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[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.
[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.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, December 2007.
[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.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, 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.
[RFC6464] Lennox, J., Ivov, E., and E. Marocco, "A Real-time
Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication", RFC 6464, December 2011.
[RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time
Transport Protocol (RTP) Header Extension for Mixer-to-
Client Audio Level Indication", RFC 6465, December 2011.
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12.2. Informative References
[I-D.begen-mmusic-redundancy-grouping]
Begen, A., Cai, Y., and H. Ou, "Duplication Grouping
Semantics in the Session Description Protocol",
draft-begen-mmusic-redundancy-grouping-01 (work in
progress), June 2011.
[I-D.cbran-rtcweb-data]
Bran, C. and C. Jennings, "RTC-Web Non-Media Data
Transport Requirements", draft-cbran-rtcweb-data-00 (work
in progress), July 2011.
[I-D.ietf-fecframe-framework]
Watson, M., Begen, A., and V. Roca, "Forward Error
Correction (FEC) Framework",
draft-ietf-fecframe-framework-15 (work in progress),
June 2011.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
September 1997.
[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.
[RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control
(TFRC): The Small-Packet (SP) Variant", RFC 4828,
April 2007.
[RFC4867] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie,
"RTP Payload Format and File Storage Format for the
Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband
(AMR-WB) Audio Codecs", RFC 4867, April 2007.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
January 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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[RFC5404] Westerlund, M. and I. Johansson, "RTP Payload Format for
G.719", RFC 5404, January 2009.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, September 2009.
[RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP
Control Protocol (RTCP)", RFC 5968, September 2010.
[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for
Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263, June 2011.
Authors' Addresses
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
Joerg Ott
Aalto University
School of Electrical Engineering
Espoo 02150
Finland
Email: jorg.ott@aalto.fi
Perkins, et al. Expires September 13, 2012 [Page 27]