AVTCORE WG M. Westerlund
Internet-Draft Ericsson
Updates: 3550, 3551 (if approved) C. Perkins
Intended status: Standards Track University of Glasgow
Expires: January 08, 2016 J. Lennox
Vidyo
July 07, 2015
Sending Multiple Types of Media in a Single RTP Session
draft-ietf-avtcore-multi-media-rtp-session-08
Abstract
This document specifies how an RTP session can contain RTP Streams
with media from multiple media types such as audio, video, and text.
This has been restricted by the RTP Specification, and thus this
document updates RFC 3550 and RFC 3551 to enable this behaviour for
applications that satisfy the applicability for using multiple media
types in a single RTP session.
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
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This Internet-Draft will expire on January 08, 2016.
Copyright Notice
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document authors. All rights reserved.
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to this document. Code Components extracted from this document must
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Background and Motivation . . . . . . . . . . . . . . . . . . 3
4. Applicability . . . . . . . . . . . . . . . . . . . . . . . . 4
5. Using Multiple Media Types in a Single RTP Session . . . . . 7
5.1. Allowing Multiple Media Types in an RTP Session . . . . . 7
5.2. Demultiplexing Media Streams . . . . . . . . . . . . . . 8
5.3. Per-SSRC Media Type Restrictions . . . . . . . . . . . . 8
5.4. RTCP Considerations . . . . . . . . . . . . . . . . . . . 9
6. Extension Considerations . . . . . . . . . . . . . . . . . . 9
6.1. RTP Retransmission Payload Format . . . . . . . . . . . . 9
6.2. RTP Payload Format for Generic FEC . . . . . . . . . . . 11
6.3. RTP Payload Format for Redundant Audio . . . . . . . . . 11
7. Signalling . . . . . . . . . . . . . . . . . . . . . . . . . 12
8. Security Considerations . . . . . . . . . . . . . . . . . . . 12
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 13
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 13
11.1. Normative References . . . . . . . . . . . . . . . . . . 13
11.2. Informative References . . . . . . . . . . . . . . . . . 14
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 15
1. Introduction
The Real-time Transport Protocol [RFC3550] was designed to use
separate RTP sessions to transport different types of media. This
implies that different transport layer flows are used for different
media streams. For example, a video conferencing application might
send audio and video traffic RTP flows on separate UDP ports. With
increased use of network address/port translation, firewalls, and
other middleboxes it is, however, becoming difficult to establish
multiple transport layer flows between endpoints. Hence, there is
pressure to reduce the number of concurrent transport flows used by
RTP applications.
This memo updates [RFC3550] and [RFC3551] to allow multiple media
types to be sent in a single RTP session in certain cases, thereby
reducing the number of transport layer flows that are needed. It
makes no changes to RTP behaviour when using multiple RTP streams
containing media of the same type (e.g., multiple audio streams or
multiple video streams) in a single RTP session, however
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[I-D.ietf-avtcore-rtp-multi-stream] provides important clarifications
to RTP behaviour in that case.
This memo is structured as follows. Section 2 defines terminology.
Section 3 further describes the background to, and motivation for,
this memo and Section 4 describes the scenarios where this memo is
applicable. (tbd: fixme)
2. Terminology
The terms Encoded Stream, Endpoint, Media Source, RTP Session, and
RTP Stream are used as defined in
[I-D.ietf-avtext-rtp-grouping-taxonomy]. We also define the
following terms:
Media Type: The general type of media data used by a real-time
application. The media type corresponds to the value used in the
<media> field of an SDP m= line. The media types defined at the
time of this writing are "audio", "video", "text", "application",
and "message".
Quality of Service (QoS): Network mechanisms that are intended to
ensure that the packets within a flow or with a specific marking
are transported with certain properties.
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 [RFC2119].
3. Background and Motivation
RTP was designed to support multimedia sessions, containing multiple
types of media sent simultaneously, by using multiple transport layer
flows. The existence of network address translators, firewalls, and
other middleboxes complicates this, however, since a mechanism is
needed to ensure that all the transport layer flows needed by the
application can be established. This has three consequences:
1. increased delay to establish a complete session, since each of
the transport layer flows needs to be negotiated and established;
2. increased state and resource consumption in the middleboxes, that
can lead to unexpected behaviour when middlebox resource limits
are reached; and
3. increased risk that a subset of the transport layer flows will
fail to be established, thus preventing the application from
communicating.
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Using fewer transport layer flows can hence be seen to reduce the
risk of communication failure, and can lead to improved reliability
and performance.
One of the benefits of using multiple transport layer flows is that
it makes it easy to use network layer quality of service (QoS)
mechanisms to give differentiated performance for different flows.
However, we note that many RTP-using application don't use network
QoS features, and don't expect or desire any separation in network
treatment of their media packets, independent of whether they are
audio, video or text. When an application has no such desire, it
doesn't need to provide a transport flow structure that simplifies
flow based QoS.
Given this, it might seem desirable for RTP-based applications to
send all their media streams bundled into one RTP session, that runs
on a single transport layer flow. Unfortunately, this is prohibited
by the RTP specification, since RTP makes certain assumptions that
can be incompatible with sending multiple media types in a single RTP
session. Specifically, the RTP control protocol (RTCP) timing rules
assume that all RTP media flows in a single RTP session have broadly
similar RTCP reporting and feedback requirements, which can be
problematic when different types of media are multiplexed together.
Certain RTP extensions also make assumptions that are incompatible
with sending different media types in a single RTP session.
This memo updates [RFC3550] and [RFC3551] to allow RTP sessions to
contain more than just one media type, and gives guidance on when it
is safe to perform such multiplexing.
4. Applicability
This specification has limited applicability, and anyone intending to
use it MUST ensure that their application and use meets the following
criteria:
Equal treatment of media: The use of a single RTP session enforces
similar treatment on all types of media used within the session.
Applications that require significantly different network QoS or
RTCP configuration for different media streams are better suited
by sending those media streams on separate RTP session, using
separate transport layer flows for each, since that gives greater
flexibility. Further guidance is given in
[I-D.ietf-avtcore-multiplex-guidelines] and
[I-D.ietf-dart-dscp-rtp].
Compatible Media Requirements: The RTCP timing rules enforce a
single RTCP reporting interval for all participants in an RTP
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session. Flows with very different media requirements, for
example a low-rate audio flow with no feedback needs and a high-
quality video flow with different repair mechanisms, cannot be
multiplexed together since this results in either excessive or
insufficient RTCP for some flows, depending how the RTCP session
bandwidth, and hence reporting interval, is configured.
Signalled Support: The extensions defined in this memo are not
compatible with unmodified [RFC3550]-compatible endpoints. Their
use requires signalling and mutual agreement by all participants
within an RTP session. This requirement can be a problem for
signalling solutions that can't negotiate with all participants.
For declarative signalling solutions, mandating that the session
is using multiple media types in one RTP session can be a way of
attempting to ensure that all participants in the RTP session
follow the requirement. However, for signalling solutions that
lack methods for enforcing that a receiver supports a specific
feature, this can still cause issues.
Consistent support for multiple media types in a single RTP session:
In multiparty communication scenarios it is important to separate
two different cases. One case is where the RTP session contains
multiple participants in a common RTP session. This occurs for
example in Any Source Multicast (ASM) and Relay (Transport
Translator) topologies as defined in RTP Topologies
[I-D.ietf-avtcore-rtp-topologies-update]. It can also occur in
some implementations of RTP mixers that share the same SSRC/CSRC
space across all participants. The second case is when the RTP
session is terminated in a middlebox and the other participants
sources are projected or switched into each RTP session and
rewritten on RTP header level including SSRC mappings.
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For the first case, with a common RTP session or at least shared
SSRC/CSRC values, all participants in multiparty communication are
REQUIRED to support multiple media types in an RTP session. An
participant using two or more RTP sessions towards a multiparty
session can't be collapsed into a single session with multiple
media types. The reason is that in case of multiple RTP sessions,
the same SSRC value can be use in both RTP sessions without any
issues, but when collapsed to a single session there is an SSRC
collision. In addition some collisions can't be represented in
the multiple separate RTP sessions. For example, in a session
with audio and video, an SSRC value used for video will not show
up in the Audio RTP session at the participant using multiple RTP
sessions, and thus not trigger any collision handling. Thus any
application using this type of RTP session structure MUST have a
homogeneous support for multiple media types in one RTP session,
or be forced to insert a translator node between that participant
and the rest of the RTP session.
For the second case of separate RTP sessions for each multiparty
participant and a central node it is possible to have a mix of
single RTP session users and multiple RTP session users as long as
one is willing to remap the SSRCs used by a participant with
multiple RTP sessions into non-used values in the single RTP
session SSRC space for each of the participants using a single RTP
session with multiple media types. It can be noted that this type
of implementation has to understand all types of RTP/RTCP
extension being used in the RTP sessions to correctly be able to
translate them between the RTP sessions. It might also suffer
issues due to differencies in configured RTCP bandwidth and other
parameters between the RTP sessions. It can also negatively
impact the possibility for loop detection, as SSRC/CSRC can't be
used to detect the loops, instead some other RTP stream or media
source identity name space that is common across all interconnect
parts are needed.
Ability to operate with limited payload type space: An RTP session
has only a single 7-bit payload type space for all its payload
type numbers. Some applications might find this space limiting
when media different media types and RTP payload formats are using
within a single RTP session.
Avoids incompatible Extensions: Some RTP and RTCP extensions rely on
the existence of multiple RTP sessions and relate media streams
between sessions. Others report on particular media types, and
cannot be used with other media types. Applications that send
multiple types of media into a single RTP session need to avoid
such extensions.
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5. Using Multiple Media Types in a Single RTP Session
This section defines what needs to be done or avoided to make an RTP
session with multiple media types function without issues.
5.1. Allowing Multiple Media Types in an RTP Session
Section 5.2 of "RTP: A Transport Protocol for Real-Time Applications"
[RFC3550] states:
For example, in a teleconference composed of audio and video media
encoded separately, each medium SHOULD be carried in a separate
RTP session with its own destination transport address.
Separate audio and video streams SHOULD NOT be carried in a single
RTP session and demultiplexed based on the payload type or SSRC
fields.
This specification changes both of these sentences. The first
sentence is changed to:
For example, in a teleconference composed of audio and video media
encoded separately, each medium SHOULD be carried in a separate
RTP session with its own destination transport address, unless
specification [RFCXXXX] is followed and the application meets the
applicability constraints.
The second sentence is changed to:
Separate audio and video media sources SHOULD NOT be carried in a
single RTP session, unless the guidelines specified in [RFCXXXX]
are followed.
Second paragraph of Section 6 in RTP Profile for Audio and Video
Conferences with Minimal Control [RFC3551] says:
The payload types currently defined in this profile are assigned
to exactly one of three categories or media types: audio only,
video only and those combining audio and video. The media types
are marked in Tables 4 and 5 as "A", "V" and "AV", respectively.
Payload types of different media types SHALL NOT be interleaved or
multiplexed within a single RTP session, but multiple RTP sessions
MAY be used in parallel to send multiple media types. An RTP
source MAY change payload types within the same media type during
a session. See the section "Multiplexing RTP Sessions" of RFC
3550 for additional explanation.
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This specifications purpose is to violate that existing SHALL NOT
under certain conditions. Thus this sentence also has to be changed
to allow for multiple media type's payload types in the same session.
The above sentence is changed to:
Payload types of different media types SHALL NOT be interleaved or
multiplexed within a single RTP session unless as specified and
under the restriction in Multiple Media Types in an RTP Session
[RFCXXXX]. Multiple RTP sessions MAY be used in parallel to send
multiple media types.
RFC-Editor Note: Please replace RFCXXXX with the RFC number of this
specification when assigned.
5.2. Demultiplexing Media Streams
When receiving packets from a transport layer flow, an endpoint will
first separate the RTP and RTCP packets from the non-RTP packets, and
pass them to the RTP/RTCP protocol handler. The RTP and RTCP packets
are then demultiplexed based on their SSRC into the different media
streams. For each media stream, incoming RTCP packets are processed,
and the RTP payload type is used to select the appropriate media
decoder.
This process remains the same irrespective of whether multiple media
types are sent in a single RTP session or not. It is important to
note that the RTP payload type is never used to demultiplex media
streams. Media streams are distinguished by SSRC, and the payload
type is then used to route data for a particular SSRC to the right
media decoder.
5.3. Per-SSRC Media Type Restrictions
An SSRC in an RTP session MUST NOT change media type during its
lifetime. For example, an SSRC cannot start sending audio, then
change to sending video. The lifetime of an SSRC ends when an RTCP
BYE packet for that SSRC is sent, or when it ceases transmission for
long enough that it times out for the other participants in the
session.
The main motivation is that a given SSRC has its own RTP timestamp
and sequence number spaces. The same way that you can't send two
encoded streams of audio on the same SSRC, you can't send one encoded
audio and one encoded video stream on the same SSRC. Each encoded
stream when made into an RTP stream needs to have the sole control
over the sequence number and timestamp space. If not, one would not
be able to detect packet loss for that particular encoded stream.
Nor can one easily determine which clock rate a particular SSRCs
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timestamp will increase with. For additional arguments why RTP
payload type based multiplexing of multiple media sources doesn't
work see [I-D.ietf-avtcore-multiplex-guidelines].
Within an RTP session where multiple media types have been configured
for use, an SSRC can only send one type of media during its lifetime
(i.e., it can switch between different audio codecs, since those are
both the same type of media, but cannot switch between audio and
video). Different SSRCs MUST be used for the different media
sources, the same way multiple media sources of the same media type
already have to do. The payload type will inform a receiver which
media type the SSRC is being used for. Thus the payload type MUST be
unique across all of the payload configurations independent of media
type that is used in the RTP session.
5.4. RTCP Considerations
When sending multiple types of media that have different rates in a
single RTP session, endpoints MUST follow the guidelines for handling
RTCP described in Section 7 of [I-D.ietf-avtcore-rtp-multi-stream].
6. Extension Considerations
This section outlines known issues and incompatibilities with RTP and
RTCP extensions when multiple media types are used in a single RTP
sessions. Future extensions to RTP and RTCP need to consider, and
document, any potential incompatibility.
6.1. RTP Retransmission Payload Format
SSRC-multiplexed RTP retransmission [RFC4588] is actually very
straightforward. Each retransmission RTP payload type is explicitly
connected to an associated payload type. If retransmission is only
to be used with a subset of all payload types, this is not a problem,
as it will be evident from the retransmission payload types which
payload types have retransmission enabled for them.
Session-multiplexed RTP retransmission is also possible to use where
an retransmission session contains the retransmissions of the
associated payload types in the source RTP session. The only
difference to the previous case is if the source RTP session is one
which contains multiple media types. This results in the
retransmission streams in the RTP session for the retransmission
having multiple associated media types.
When using SDP signalling for a multiple media type RTP session, i.e.
BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation], the session
multiplexed case do require some recommendations on how to signal
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this. To avoid breaking the semantics of the FID grouping as defined
by [RFC5888] each media line can only be included in one FID group.
FID is used by RTP retransmission to indicate the SDP media lines
that is a source and retransmission pair. Thus, for SDP using
BUNDLE, each original media source (m= line) that is retransmitted
needs a corresponding media line in the retransmission RTP session.
In case there are multiple media lines for retransmission, these
media lines will form a independent BUNDLE group from the BUNDLE
group with the source streams.
Below is an SDP example (Figure 1) which shows the grouping
structures. This example is not legal SDP and only the most
important attributes has been left in place. Note that this SDP is
not an initial BUNDLE offer. As can be seen there are two bundle
groups, one for the source RTP session and one for the
retransmissions. Then each of the media sources are grouped with its
retransmission flow using FID, resulting in three more groupings.
a=group:BUNDLE foo bar fiz
a=group:BUNDLE zoo kelp glo
a=group:FID foo zoo
a=group:FID bar kelp
a=group:FID fiz glo
m=audio 10000 RTP/AVP 0
a=mid:foo
a=rtpmap:0 PCMU/8000
m=video 10000 RTP/AVP 31
a=mid:bar
a=rtpmap:31 H261/90000
m=video 10000 RTP/AVP 31
a=mid:fiz
a=rtpmap:31 H261/90000
m=audio 40000 RTP/AVPF 99
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=0;rtx-time=3000
a=mid:zoo
m=video 40000 RTP/AVPF 100
a=rtpmap:100 rtx/90000
a=fmtp:199 apt=31;rtx-time=3000
a=mid:kelp
m=video 40000 RTP/AVPF 100
a=rtpmap:100 rtx/90000
a=fmtp:199 apt=31;rtx-time=3000
a=mid:glo
Figure 1: SDP example of Session Multiplexed RTP Retransmission
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6.2. RTP Payload Format for Generic FEC
The RTP Payload Format for Generic Forward Error Correction
[RFC5109], and also its predecessor [RFC2733], requires some
considerations, and they are different depending on what type of
configuration of usage one has.
Independent RTP Sessions, i.e. where source and repair data are sent
in different RTP sessions. As this mode of configuration requires
different RTP session, there has to be at least one RTP session for
source data, this session can be one using multiple media types. The
repair session only needs one RTP Payload type indicating repair
data, i.e. x/ulpfec or x/parityfec depending if RFC 5109 or RFC 2733
is used. The media type in this session is not relevant and can in
theory be any of the defined ones. It is RECOMMENDED that one uses
"Application".
If one uses SDP signalling with BUNDLE
[I-D.ietf-mmusic-sdp-bundle-negotiation], then the RTP session
carrying the FEC streams will be its own BUNDLE group. The media
line with the source stream for the FEC and the FEC stream's media
line will be grouped using media line grouping using the FEC or FEC-
FR [RFC5956] grouping. This is very similar to the situation that
arise for RTP retransmission with session multiplexing discussed
above inSection 6.1.
The RTP Payload Format for Generic Forward Error Correction [RFC5109]
and its predecessor [RFC2733] requires a separate RTP session unless
the FEC data is carried in RTP Payload for Redundant Audio Data
[RFC2198].
Note that the Source-Specific Media Attributes [RFC5576]
specification defines an SDP syntax (the "FEC" semantic of the "ssrc-
group" attribute) to signal FEC relationships between multiple RTP
streams within a single RTP session. However, this can't be used as
the FEC repair packets need to have the same SSRC value as the source
packets being protected. [RFC5576] does not normatively update and
resolve that restriction. There is ongoing work on an ULP extension
to allow it be use FEC RTP streams within the same RTP Session as the
source stream [I-D.lennox-payload-ulp-ssrc-mux].
6.3. RTP Payload Format for Redundant Audio
In stream, using RTP Payload for Redundant Audio Data [RFC2198]
combining repair and source data in the same packets. This is
possible to use within a single RTP session. However, the usage and
configuration of the payload types can create an issue. First of all
it might be necessary to have one payload type per media type for the
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FEC repair data payload format, i.e. one for audio/ulpfec and one
for text/ulpfec if audio and text are combined in an RTP session.
Secondly each combination of source payload and its FEC repair data
has to be an explicit configured payload type. This has potential
for making the limitation of RTP payload types available into a real
issue.
7. Signalling
Establishing an RTP session with multiple media types requires
signalling. This signalling needs to fulfil the following
requirements:
1. Ensure that any participant in the RTP session is aware that this
is an RTP session with multiple media types.
2. Ensure that the payload types in use in the RTP session are using
unique values, with no overlap between the media types.
3. Configure the RTP session level parameters, such as RTCP RR and
RS bandwidth, AVPF trr-int, underlying transport, the RTCP
extensions in use, and security parameters, commonly for the RTP
session.
4. RTP and RTCP functions that can be bound to a particular media
type SHOULD be reused when possible also for other media types,
instead of having to be configured for multiple code-points.
Note: In some cases one will not have a choice but to use
multiple configurations.
The signalling of multiple media types in one RTP session in SDP is
specified in "Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers"
[I-D.ietf-mmusic-sdp-bundle-negotiation].
8. Security Considerations
Having an RTP session with multiple media types doesn't change the
methods for securing a particular RTP session. One possible
difference is that the different media have often had different
security requirements. When combining multiple media types in one
session, their security requirements also have to be combined by
selecting the most demanding for each property. Thus having multiple
media types can result in increased overhead for security for some
media types to ensure that all requirements are meet.
Otherwise, the recommendations for how to configure and RTP session
do not add any additional requirements compared to normal RTP, except
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for the need to be able to ensure that the participants are aware
that it is a multiple media type session. If not that is ensured it
can cause issues in the RTP session for both the unaware and the
aware one. Similar issues can also be produced in an normal RTP
session by creating configurations for different end-points that
doesn't match each other.
9. IANA Considerations
This memo makes no request of IANA.
10. Acknowledgements
The authors would like to thank Christer Holmberg, Gunnar Hellstroem,
and Charles Eckel for the feedback on the document.
11. References
11.1. Normative References
[I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-07 (work in progress),
March 2015.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-22 (work in progress), June 2015.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[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.
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11.2. Informative References
[I-D.ietf-avtcore-multiplex-guidelines]
Westerlund, M., Perkins, C., and H. Alvestrand,
"Guidelines for using the Multiplexing Features of RTP to
Support Multiple Media Streams", draft-ietf-avtcore-
multiplex-guidelines-03 (work in progress), October 2014.
[I-D.ietf-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies", draft-
ietf-avtcore-rtp-topologies-update-10 (work in progress),
July 2015.
[I-D.ietf-avtext-rtp-grouping-taxonomy]
Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, "A Taxonomy of Semantics and Mechanisms for
Real-Time Transport Protocol (RTP) Sources", draft-ietf-
avtext-rtp-grouping-taxonomy-07 (work in progress), June
2015.
[I-D.ietf-dart-dscp-rtp]
Black, D. and P. Jones, "Differentiated Services
(DiffServ) and Real-time Communication", draft-ietf-dart-
dscp-rtp-10 (work in progress), November 2014.
[I-D.lennox-payload-ulp-ssrc-mux]
Lennox, J., "Supporting Source-Multiplexing of the Real-
Time Transport Protocol (RTP) Payload for Generic Forward
Error Correction", draft-lennox-payload-ulp-ssrc-mux-00
(work in progress), February 2013.
[I-D.westerlund-avtcore-transport-multiplexing]
Westerlund, M. and C. Perkins, "Multiplexing Multiple RTP
Sessions onto a Single Lower-Layer Transport", draft-
westerlund-avtcore-transport-multiplexing-07 (work in
progress), October 2013.
[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.
[RFC2733] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format
for Generic Forward Error Correction", RFC 2733, December
1999.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
Westerlund, et al. Expires January 08, 2016 [Page 14]
Internet-Draft Multiple Media Types in an RTP Session July 2015
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, December 2007.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC5956] Begen, A., "Forward Error Correction Grouping Semantics in
the Session Description Protocol", RFC 5956, September
2010.
Authors' Addresses
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
Westerlund, et al. Expires January 08, 2016 [Page 15]
Internet-Draft Multiple Media Types in an RTP Session July 2015
Jonathan Lennox
Vidyo, Inc.
433 Hackensack Avenue
Seventh Floor
Hackensack, NJ 07601
US
Email: jonathan@vidyo.com
Westerlund, et al. Expires January 08, 2016 [Page 16]
