AVTCORE WG M. Westerlund
Internet-Draft Ericsson
Updates: 3550, 3551 (if approved) C. Perkins
Intended status: Standards Track University of Glasgow
Expires: March 18, 2016 J. Lennox
Vidyo
September 15, 2015
Sending Multiple Types of Media in a Single RTP Session
draft-ietf-avtcore-multi-media-rtp-session-10
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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Internet-Drafts are draft documents valid for a maximum of six months
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This Internet-Draft will expire on March 18, 2016.
Copyright Notice
Copyright (c) 2015 IETF Trust and the persons identified as the
document authors. All rights reserved.
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to this document. Code Components extracted from this document must
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described in the Simplified BSD License.
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 . . . . . 6
5.1. Allowing Multiple Media Types in an RTP Session . . . . . 6
5.2. Demultiplexing media types within an RTP session . . . . 7
5.3. Per-SSRC Media Type Restrictions . . . . . . . . . . . . 8
5.4. RTCP Considerations . . . . . . . . . . . . . . . . . . . 8
6. Extension Considerations . . . . . . . . . . . . . . . . . . 9
6.1. RTP Retransmission Payload Format . . . . . . . . . . . . 9
6.2. RTP Payload Format for Generic FEC . . . . . . . . . . . 10
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 . . . . . . . . . . . . . . . . . 13
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. Section 5 discusses issues arising from the base RTP and
RTCP specification when using multiple types of media in a single RTP
session, while Section 6 considers the impact of RTP extensions. We
discuss signalling in Section 7. Finally, security considerations
are discussed in Section 8.
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", "image",
"application", and "message". [RFC4566] [RFC6466]
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
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3. increased risk that a subset of the transport layer flows will
fail to be established, thus preventing the application from
communicating.
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 the above issues, it might seem appropriate for RTP-based
applications to send all their media streams bundled into one RTP
session, running over a single transport layer flow. However, this
is prohibited by the RTP specification, because the design of 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. Various RTP extensions also make
assumptions about SSRC use and RTCP reporting 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 one media type in certain circumstances, and gives
guidance on when it is safe to send multiple media types in a single
RTP session.
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
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flexibility. Further guidance is given in
[I-D.ietf-avtcore-multiplex-guidelines] and
[I-D.ietf-dart-dscp-rtp].
Compatible RTCP Behaviour: The RTCP timing rules enforce a single
RTCP reporting interval for all participants in an RTP session.
Flows with very different media sending rate or RTCP feedback
requirements cannot be multiplexed together, since this leads to
either excessive or insufficient RTCP for some flows, depending
how the RTCP session bandwidth, and hence reporting interval, is
configured. For example, it is likely not feasible to find a
single RTCP configuration that simultaneously suits both a low-
rate audio flow with no feedback and a high-quality video flow
with sophisticated RTCP-based feedback needs, making it difficult
to combine these into a single RTP session.
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 multiparty RTP sessions: If it is desired to
send multiple types of media in a multiparty RTP session, then all
participants in that session need to support sending multiple type
of media in a single RTP session. It is not possible, in the
general case, to implement a gateway that can interconnect an
endpoint using multiple types of media sent using separate RTP
sessions, with one or more endpoints that send multiple types of
media in a single RTP session.
One reason for this is that the same SSRC value can safely be used
for different streams in multiple RTP sessions, but when collapsed
to a single RTP session there is an SSRC collision. This would
not be an issue, since SSRC collision detection will resolve the
conflict, except that some RTP payload formats and extensions use
matching SSRCs to identify related flows, and break when a single
RTP session is used.
A middlebox that remaps SSRC values when combining multiple RTP
sessions into one also needs to be aware of all possible RTCP
packet types that might be used, so that it can remap the SSRC
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values in those packets. This is impossible to do without
restricting the set of RTCP packet types that can be used to those
that are known by the middlebox. Such a middlebox might also have
difficulty due to differences in configured RTCP bandwidth and
other parameters between the RTP sessions.
Finally, the use of a middlebox that translates SSRC values can
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.
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
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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.
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 [RFCXXXX] is used,
and the application conforms to the applicability constraints.
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 types within an RTP session
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.
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It is important to note that the RTP payload type is never used to
distinguish media streams. The RTP packets are demultiplexed into
media streams based on their SSRC, then the RTP payload type is used
to select the correct media decoding pathway for each media stream.
5.3. Per-SSRC Media Type Restrictions
An SSRC in an RTP session can change between media formats of the
same type, subject to certain restrictions [RFC7160], but MUST NOT
change media type during its lifetime. For example, an SSRC can
change between different audio formats, but 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
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].
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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
The RTP Retransmission Payload Format [RFC4588] can operate in either
SSRC-multiplexed mode or session-multiplex mode.
In SSRC-multiplexed mode, retransmitted RTP packets are sent in the
same RTP session as the original packets, but use a different SSRC
with the same RTCP SDES CNAME. If each endpoint sends only a single
original RTP stream and a single retransmission RTP stream in the
session, this is sufficient. If an endpoint sends multiple original
and retransmission RTP streams, as would occur when sending multiple
media types in a single RTP session, then each original RTP stream
and the retransmission RTP stream have to be associated using
heuristics. By having retransmission requests outstanding for only
one SSRC not yet mapped, a receiver can determine the binding between
original and retransmission RTP stream. Another alternative is the
use of different RTP payload types, allowing the signalled "apt"
(associated payload type) parameter of the RTP retransmission payload
format to be used to associate retransmitted and original packets.
Session-multiplexed mode sends the retransmission RTP stream in a
separate RTP session to the original RTP stream, but using the same
SSRC for each, with association being done by matching SSRCs between
the two sessions. This is unaffected by the use of multiple media
types in a single RTP session, since each media type will be sent
using a different SSRC in the original RTP session, and the same
SSRCs can be used in the retransmission session, allowing the streams
to be associated. This can be signalled using SDP with the BUNDLE
[I-D.ietf-mmusic-sdp-bundle-negotiation] and FID grouping [RFC5888]
extensions. These SDP extensions require each "m=" line to only be
included in a single FID group, but the RTP retransmission payload
format uses FID groups to indicate the m= lines that form an original
and retransmission pair. Accordingly, when using the BUNDLE
extension to allow multiple media types to be sent in a single RTP
session, each original media source (m= line) that is retransmitted
needs a corresponding m= 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.
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An example SDP fragment showing the grouping structures is provided
in Figure 1. This example is not legal SDP and only the most
important attributes have 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
6.2. RTP Payload Format for Generic FEC
The RTP Payload Format for Generic Forward Error Correction (FEC)
[RFC5109] (and its predecessor [RFC2733]) can either send the FEC
stream as a separate RTP stream, or it can send the FEC combined with
the original RTP stream as a redundant encoding [RFC2198].
When sending FEC as a separate stream, the RTP Payload Format for
generic FEC requires that FEC stream to be sent in a separate RTP
session to the original stream, using the same SSRC, with the FEC
stream being associated by matching the SSRC between sessions. The
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RTP session used for the original streams can include multiple RTP
streams, and those RTP stream can use multiple media types. The
repair session only needs one RTP Payload type to indicate FEC data,
irrespective of the number of FEC streams sent, since the SSRC is
used to associate the FEC streams with the original streams. Hence,
it is RECOMMENDED that FEC stream use the "application/ulpfec" media
type for [RFC5109], and the "application/parityfec" media type for
[RFC2733]. It is legal, but NOT RECOMMENDED, to send FEC streams
using media specific payload format names (e.g., if an original RTP
session contains audio and video flows, for the associated FEC RTP
session where to use the "audio/ulpfec" and "video/ulpfec" payload
formats), since this unnecessarily uses up RTP payload type values,
and adds no value for demultiplexing since there might be multiple
streams of the same media type).
The combination of an original RTP session using multiple media types
with a associated generic FEC session can be signalled using SDP with
the BUNDLE extension [I-D.ietf-mmusic-sdp-bundle-negotiation]. In
this case, the RTP session carrying the FEC streams will be its own
BUNDLE group. The m= line for each original stream and the m= line
for the corresponding FEC stream are grouped using the SDP grouping
framework with either the FEC [RFC4756] or the FEC-FR [RFC5956]
grouping. This is similar to the situation that arises for RTP
retransmission with session multiplexing discussed in Section 6.1.
The Source-Specific Media Attributes [RFC5576] specification defines
an SDP extension (the "FEC" semantic of the "ssrc-group" attribute)
to signal FEC relationships between multiple RTP streams within a
single RTP session. This cannot be used with generic FEC, since the
FEC repair packets need to have the same SSRC value as the source
packets being protected. 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].
When the FEC is sent as a redundant encoding, the considerations in
Section 6.3 apply.
6.3. RTP Payload Format for Redundant Audio
The RTP Payload Format for Redundant Audio [RFC2198] can be used to
protect audio streams. It can also be used along with the generic
FEC payload format to send original and repair data in the same RTP
packets. Both are compatible with RTP sessions containing multiple
media types.
This payload format requires each different redundant encoding use a
different RTP payload type number. When used with generic FEC in
sessions that contain multiple media types, this requires each media
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type use a different payload type for the FEC stream. For example,
if audio and text are sent in a single RTP session with generic ULP
FEC sent as a redundant encoding for each, then payload types need to
be assigned for FEC using the audio/ulpfec and text/ulpfec payload
formats. If multiple original payload types of used in the session,
different redundant payload types need to be allocated for each one.
This has potential to rapidly exhaust the available RTP payload type
numbers.
7. Signalling
Establishing a single RTP session using multiple media types requires
signalling. This signalling has to:
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. ensure RTP session level parameters, for example the RTCP RR and
RS bandwidth modifiers, the RTP/AVPF trr-int parameter, transport
protocol, RTCP extensions in use, and any security parameters,
are consistent across the session; and
4. ensure that RTP and RTCP functions that can be bound to a
particular media type are reused where possible, rather than
configuring multiple code-points for the same thing.
When using SDP signalling, the BUNDLE extension
[I-D.ietf-mmusic-sdp-bundle-negotiation] is used to signal RTP
sessions containing multiple media types.
8. Security Considerations
RTP provides a range of strong security mechanisms that can be used
to secure sessions [RFC7201], [RFC7202]. The majority of these are
independent of the type of media sent in the RTP session, however it
is important to check that the security mechanism chosen is
compatible with all types of media sent within the session.
Sending multiple media types in a single RTP session will generally
require that all use the same security mechanism, whereas media sent
using different RTP sessions can be secured in different ways. When
different media types have different security requirements, it might
be necessary to send them using separate RTP sessions to meet those
different requirements. This can have significant costs in terms of
resource usage, session set-up time, etc.
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9. IANA Considerations
This memo makes no request of IANA.
10. Acknowledgements
The authors would like to thank Christer Holmberg, Gunnar Hellstroem,
Charles Eckel, and Tolga Asveren for their 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-08 (work in progress),
July 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-23 (work in progress), July 2015.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003,
<http://www.rfc-editor.org/info/rfc3551>.
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.
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[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-08 (work in progress), July
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.
[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,
DOI 10.17487/RFC2198, September 1997,
<http://www.rfc-editor.org/info/rfc2198>.
[RFC2733] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format
for Generic Forward Error Correction", RFC 2733,
DOI 10.17487/RFC2733, December 1999,
<http://www.rfc-editor.org/info/rfc2733>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <http://www.rfc-editor.org/info/rfc4566>.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
DOI 10.17487/RFC4588, July 2006,
<http://www.rfc-editor.org/info/rfc4588>.
[RFC4756] Li, A., "Forward Error Correction Grouping Semantics in
Session Description Protocol", RFC 4756,
DOI 10.17487/RFC4756, November 2006,
<http://www.rfc-editor.org/info/rfc4756>.
[RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, DOI 10.17487/RFC5109, December
2007, <http://www.rfc-editor.org/info/rfc5109>.
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[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
<http://www.rfc-editor.org/info/rfc5576>.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888,
DOI 10.17487/RFC5888, June 2010,
<http://www.rfc-editor.org/info/rfc5888>.
[RFC5956] Begen, A., "Forward Error Correction Grouping Semantics in
the Session Description Protocol", RFC 5956,
DOI 10.17487/RFC5956, September 2010,
<http://www.rfc-editor.org/info/rfc5956>.
[RFC6466] Salgueiro, G., "IANA Registration of the 'image' Media
Type for the Session Description Protocol (SDP)",
RFC 6466, DOI 10.17487/RFC6466, December 2011,
<http://www.rfc-editor.org/info/rfc6466>.
[RFC7160] Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple
Clock Rates in an RTP Session", RFC 7160,
DOI 10.17487/RFC7160, April 2014,
<http://www.rfc-editor.org/info/rfc7160>.
[RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP
Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
<http://www.rfc-editor.org/info/rfc7201>.
[RFC7202] Perkins, C. and M. Westerlund, "Securing the RTP
Framework: Why RTP Does Not Mandate a Single Media
Security Solution", RFC 7202, DOI 10.17487/RFC7202, April
2014, <http://www.rfc-editor.org/info/rfc7202>.
Authors' Addresses
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
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Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
Jonathan Lennox
Vidyo, Inc.
433 Hackensack Avenue
Seventh Floor
Hackensack, NJ 07601
US
Email: jonathan@vidyo.com
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