Network Working Group B. Burman
Internet-Draft M. Westerlund
Intended status: Standards Track Ericsson
Expires: July 24, 2015 S. Nandakumar
M. Zanaty
January 20, 2015

Using Simulcast in SDP and RTP Sessions


In some application scenarios it may be desirable to send multiple differently encoded versions of the same media source in independent RTP streams. This is called simulcast. This document discusses the best way of accomplishing simulcast in RTP and how to signal it in SDP. A solution is defined by making an extension to SDP, and using RTP/RTCP identification methods to relate RTP streams belonging to the same media source. The SDP extension consists a new media level SDP attribute that express capability to send and/or receive simulcast RTP streams. One part of the RTP/RTCP identification method is included as a reference to a separate document, since it is useful also for other purposes.

Status of This Memo

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This Internet-Draft will expire on July 24, 2015.

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Table of Contents

1. Introduction

Most of today's multiparty video conference solutions make use of centralized servers to reduce the bandwidth and CPU consumption in the endpoints. Those servers receive RTP streams from each participant and send some suitable set of possibly modified RTP streams to the rest of the participants, which usually have heterogeneous capabilities (screen size, CPU, bandwidth, codec, etc). One of the biggest issues is how to perform RTP stream adaptation to different participants' constraints with the minimum possible impact on both video quality and server performance.

Simulcast is defined in this memo as the act of simultaneously sending multiple different encoded streams of the same media source, e.g. the same video source encoded with different video encoder types or image resolutions. This can be done in several ways and for different purposes. This document focuses on the case where it is desirable to provide a media source as multiple encoded streams over RTP [RFC3550] towards an intermediary so that the intermediary can provide the wanted functionality by selecting which RTP stream to forward to other participants in the session, and more specifically how the identification and grouping of the involved RTP streams are done. From an RTP perspective, simulcast is a specific application of the aspects discussed in RTP Multiplexing Guidelines [I-D.ietf-avtcore-multiplex-guidelines].

The purpose of this document is to describe a few scenarios where it is motivated to use simulcast, and propose a suitable solution for SDP signaling and performing RTP simulcast.

2. Definitions

2.1. Terminology

This document makes use of the terminology defined in RTP Taxonomy [I-D.ietf-avtext-rtp-grouping-taxonomy], RTP Topology [RFC5117] and RTP Topologies Update [I-D.ietf-avtcore-rtp-topologies-update]. In addition, the following terms are used:

RTP Mixer:
An RTP middle node, defined in [RFC5117] (Section 3.4: Topo-Mixer), further elaborated and extended with other topologies in [I-D.ietf-avtcore-rtp-topologies-update] (Section 3.6 to 3.9).
RTP Switch:
A common short term for the terms "switching RTP mixer", "source projecting middlebox", and "video switching MCU" as discussed in [I-D.ietf-avtcore-rtp-topologies-update].
Simulcast version:
One encoded stream from the set of encoded streams that constitutes the simulcast for a single media source.
Simulcast version alternative:
One encoded stream being encoded in one of possibly multiple alternative ways to create a simulcast version.

2.2. Requirements Language

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [RFC2119].

3. Use Cases

Many use cases of simulcast as described in this document relate to a multi-party communication session where one or more central nodes are used to adapt the view of the communication session towards individual participants, and facilitate the media transport between participants. Thus, these cases targets the RTP Mixer type of topology.

There are two principle approaches for an RTP Mixer to provide this adapted view of the communication session to each receiving participant:

The use of simulcast relates to the latter approach, where it is more important to reduce the load on the RTP Mixer and/or minimize QoE impact than to achieve an optimal adaptation of resource usage.

A multicast/broadcast case where the receivers themselves selects the most appropriate simulcast version and tune in to the right media transport to receive that version is also considered [sec-multicast] . This enables large, heterogeneous receiver populations, when it comes to capabilities and the use of network path bandwidth resources.

3.1. Reaching a Diverse Set of Receivers

The media sources provided by a sending participant potentially need to reach several receiving participants that differ in terms of available resources. The receiver resources that typically differ include, but are not limited to:

This includes codec type (such as SDP MIME type) and can include codec configuration options (e.g. SDP fmtp parameters). A couple of codec resources that differ only in codec configuration will be "different" if they are somehow not "compatible", like if they differ in video codec profile, or the transport packetization configuration.
This relates to how the media source is sampled, in spatial as well as in temporal domain. For video streams, spatial sampling affects image resolution and temporal sampling affects video frame rate. For audio, spatial sampling relates to the number of audio channels and temporal sampling affects audio bandwidth. This may be used to suit different rendering capabilities or needs at the receiving endpoints, as well as a method to achieve different transport capabilities, bitrates and eventually QoE by controlling the amount of source data.
This relates to the amount of bits spent per second to transmit the media source as an RTP stream, which typically also affects the Quality of Experience (QoE) for the receiving user.

Letting the sending participant create a simulcast of a few differently configured RTP streams per media source can be a good tradeoff when using an RTP switch as middlebox, instead of sending a single RTP stream and using an RTP mixer to create individual transcodings to each receiving participant.

This requires that the receiving participants can be categorized in terms of available resources and that the sending participant can choose a matching configuration for a single RTP stream per category and media source.

For example, assume for simplicity a set of receiving participants that differ only in that some have support to receive Codec A, and the others have support to receive Codec B. Further assume that the sending participant can send both Codec A and B. It can then reach all receivers by creating two simulcasted RTP streams from each media source; one for Codec A and one for Codec B.

In another simple example, a set of receiving participants differ only in screen resolution; some are able to display video with at most 360p resolution and some support 720p resolution. A sending participant can then reach all receivers by creating a simulcast of RTP streams with 360p and 720p resolution for each sent video media source.

In more elaborate cases, the receiving participants differ both in available sampling and bitrate, and maybe also codec, and it is up to the RTP switch to find a good trade-off in which simulcasted stream to choose for each intended receiver. It is also the responsibility of the RTP switch to negotiate a good fit of simulcast streams with the sending participant.

The maximum number of simulcasted RTP streams that can be sent is mainly limited by the amount of processing and uplink network resources available to the sending participant.

3.2. Application Specific Media Source Handling

The application logic that controls the communication session may include special handling of some media sources. It is for example commonly the case that the media from a sending participant is not sent back to itself.

It is also common that a currently active speaker participant is shown in larger size or higher quality than other participants (the sampling or bitrate aspects of Section 3.1). Not sending the active speaker media back to itself means there is some other participant's media that instead has to receive special handling towards the active speaker; typically the previous active speaker. This way, the previously active speaker is needed both in larger size (to current active speaker) and in small size (to the rest of the participants), which can be solved with a simulcast from the previously active speaker to the RTP switch.

3.3. Receiver Adaptation in Multicast/Broadcast

When using broadcast or multicast technology to distribute real-time media streams to large populations of receivers, there can still be significant heterogeneity among the receiver population. This can depend on several factors:

Network Bandwidth:
The network paths to individual receivers will have variations in the bandwidth, thus putting different limits on the supported bit-rates that can be received.
Endpoint Capabilities:
The end point's hardware and software can have varying capabilities in relation to screen resolution, decoding capabilities, and supported media codecs.

To handle these variations, a transmitter of real-time media may want to apply simulcast to a media source and provide it as a set of different encoded streams, enabling the receivers to select the best fit from this set themselves. The end point capabilities will usually result in a single initial choice. However, the network bandwidth can vary over time, which requires a client to continuously monitor its reception to determine if the received RTP streams still fit within the available bandwidth. If not, another set of encoded streams from the ones offered in the simulcast will have to be chosen.

When using IP multicast, the level of granularity that the receiver can select from is decided by its ability to choose different multicast addresses. Thus, different simulcast versions need to be put on different media transports using different multicast addresses. If these simulcast versions are described using SDP, they need to be part of different SDP media descriptions, as SDP binds to transport on media description level.

3.4. Receiver Media Source Preferences

The application logic that controls the communication session may allow receiving participants to apply preferences to the characteristics of the RTP stream they receive, for example in terms of the aspects listed in Section 3.1. Sending a simulcast of RTP streams is one way of accommodating receivers with conflicting or otherwise incompatible preferences.

4. Requirements

The following requirements need to be met to support the use cases in previous sections:

Identification. It must be possible to identify a set of simulcasted RTP streams as originating from the same media source:
In SDP signaling.
On RTP/RTCP level.
Transport usage. The solution must work when using:
Legacy SDP with separate media transports per SDP media description.
Bundled SDP media descriptions.
Capability negotiation. It must be possible that:
Sender can express capability of sending simulcast.
Receiver can express capability of receiving simulcast.
Sender can express maximum number of simulcast versions that can be provided.
Receiver can express maximum number of simulcast versions that can be received.
Sender can detail the characteristics of the simulcast versions that can be provided.
Receiver can detail the characteristics of the simulcast versions that it prefers to receive.
Distinguishing features. It must be possible to have different simulcast versions use different codec parameters, as can be expressed by SDP format values and RTP payload types.
Compatibility. It must be possible to use simulcast in combination with other RTP mechanisms that generate additional RTP streams:
RTP Retransmission [RFC4588].
RTP Forward Error Correction [RFC5109].
Related payload types such as audio Comfort Noise and/or DTMF.
Interoperability. The solution must be possible to use in:
Interworking with non-simulcast legacy clients using a single media source per media type.
WebRTC "Unified Plan" environment with a single media source per SDP media description.

5. Proposed Solution Overview

The proposed solution consists of signaling simulcast capability and configurations in SDP [RFC4566]:

6. Proposed Solution

This section further details the signaling solution outlined above [sec-solution-overview].

6.1. Simulcast Capability

Simulcast capability is expressed as a new media level SDP attribute, "a=simulcast". For each desired direction (send/recv/sendrecv), the simulcast attribute defines a list of simulcast versions (separated by semicolons), each of which is a list of alternative RTP payload types (separated by commas) for that simulcast version. The meaning of the attribute on SDP session level is undefined and MUST NOT be used. There MUST be at most one "a=simulcast" attribute per media description. The ABNF [RFC5234] for this attribute is:

simulcast-attribute = "a=simulcast" 1*3( WSP sc-dir-list )
sc-dir-list         = sc-dir WSP sc-fmt-list *( ";" sc-fmt-list )
sc-dir              = "send" / "recv" / "sendrecv"
sc-fmt-list         = sc-fmt *( "," sc-fmt )
sc-fmt              = fmt
; WSP defined in [RFC5234]
; fmt defined in [RFC4566]

Figure 1: ABNF for Simulcast

There are separate and independent sets of parameters for simulcast in send and receive directions. When listing multiple directions, each direction MUST NOT occur more than once.

Attribute parameters are grouped by direction and consist of a listing of SDP format tokens (usually corresponding to RTP payload types), which describe the simulcast versions to be used. The number of (non-alternative, see below) formats in the list sets a limit to the number of supported simulcast versions in that direction. The order of the listed simulcast versions in the "send" direction is not significant. The order of the listed simulcast versions in the "recv" direction expresses a preference which simulcast versions that are preferred, with the leftmost being most preferred, if the number of actually sent simulcast versions have to be reduced for some reason.

Formats that have explicit dependencies [RFC5583] to other formats (even in the same media description) MAY be listed as different simulcast versions.

Alternative simulcast versions MAY be specified as part of the attribute parameters by expressing each simulcast version format as a comma-separated list of alternative values. In this case, all combinations of those alternatives MUST be supported. The order of the alternatives within a simulcast version is not significant; codec preference is expressed by format type ordering on the m-line, using regular SDP rules.

A simulcast version can use a codec defined such that the same RTP SSRC can change RTP payload type multiple times during a session, possibly even on a per-packet basis. A typical example can be a speech codec that makes use of Comfort Noise [RFC3389] and/or DTMF [RFC4733] formats. In those cases, such "related" formats MUST NOT be listed explicitly in the attribute parameters, since they are not strictly simulcast versions of the media source, but rather a specific way of generating the RTP stream of a single simulcast version with varying RTP payload type. Instead, only a single codec format MUST be used per simulcast version or simulcast version alternative (if there are such). The codec format SHOULD be the codec most relevant to the media description, if possible to identify, for example the audio codec rather than the DTMF. What codec format to choose in the case of switching between multiple equally "important" formats is left open, but it is assumed that in the presence of such strong relation it does not matter which is chosen.

Use of the redundant audio data [RFC2198] format could be seen as a form of simulcast for loss protection purposes, but is not considered conflicting with the mechanisms described in this memo and MAY therefore be used as any other format. In this case the "red" format, rather than the carried formats, SHOULD be the one to list as a simulcast version on the "a=simulcast" line.

6.1.1. Declarative Use

When used as a declarative media description, a=simulcast "recv" direction formats indicates the configured end point's required capability to recognize and receive a specified set of RTP streams as simulcast streams. In the same fashion, a=simulcast "send" direction requests the end point to send a specified set of RTP streams as simulcast streams. The "sendrecv" direction combines "send" and "recv" requirements, using the same format values for both.

If simulcast version alternatives are listed, it means that the configured end point MUST be prepared to receive any of the "recv" formats, and MAY send any of the "send" formats for that simulcast version.

6.1.2. Offer/Answer Use

An offerer wanting to use simulcast SHALL include the "a=simulcast" attribute in the offer. An offerer that receives an answer without "a=simulcast" MUST NOT use simulcast towards the answerer. An offerer that receives an answer with "a=simulcast" not listing a direction or without any formats in a specified direction MUST NOT use simulcast in that direction.

An answerer that does not understand the concept of simulcast will also not know the attribute and will remove it in the SDP answer, as defined in existing SDP Offer/Answer [RFC3264] procedures. An answerer that does understand the attribute and that wants to support simulcast in an indicated direction SHALL reverse directionality of the unidirectional direction parameters; "send" becomes "recv" and vice versa, and include it in the answer. If the offered direction is "sendrecv", the answerer MAY keep it, but MAY also change it to "send" or "recv" to indicate that it is only interested in simulcast for a single direction. Note that, like all other use of SDP format tags for the send direction in Offer/Answer, format tags related to the simulcast send direction in an offer ("send" or "sendrecv") are placeholders that refer to information in the offer SDP, and the actual formats that will be used on the wire (including RTP Payload Format numbers) depends on information included in the SDP answer.

An offerer listing a set of receive simulcast versions and/or alternatives in the offer MUST be prepared to receive RTP streams for any of those simulcast versions and/or alternatives from the answerer.

An answerer that receives an offer with simulcast containing an "a=simulcast" attribute listing alternative formats for simulcast versions MAY keep all the alternatives in the answer, but it MAY also choose to remove any non-desirable alternatives per simulcast version in the answer. The answerer MUST NOT add any alternatives that were not present in the offer.

An answerer that receives an offer with simulcast that lists a number of simulcast versions, MAY reduce the number of simulcast versions in the answer, but MUST NOT add simulcast versions.

An offerer that receives an answer where some simulcast version alternatives are kept MUST be prepared to receive any of the kept send direction alternatives, and MAY send any of the kept receive direction alternatives from the answer. This is similar to the case when the answer includes multiple formats on the m-line.

An offerer that receives an answer where some of the simulcast versions are removed MAY release the corresponding resources (codec, transport, etc) in its receive direction and MUST NOT send any RTP streams corresponding to the removed simulcast versions.

The media formats and corresponding characteristics of encoded streams used in a simulcast SHOULD be chosen such that they are different. If this difference is not required, RTP duplication [RFC7104] procedures SHOULD be considered instead of simulcast.

6.2. Relating Simulcast Versions

As long as there is only a single media source per SDP media description, simulcast RTP streams can be related on RTP level through the RTP payload type, as specified in the SDP "a=simulcast" attribute [sec-cap] parameters. When using BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation] to use multiple SDP media descriptions to specify a single RTP session, there is an identification mechanism that allows relating RTP streams back to individual media descriptions, after which the above RTP payload type relation can be used.

6.3. Signaling Examples

These examples are for a case of client to video conference service using a centralized media topology with an RTP mixer.

+---+      +-----------+      +---+
| A |<---->|           |<---->| B |
+---+      |           |      +---+
           |   Mixer   |
+---+      |           |      +---+
| F |<---->|           |<---->| J |
+---+      +-----------+      +---+

Figure 2: Four-party Mixer-based Conference

6.3.1. Unified Plan Client

Alice is calling in to the mixer with a simulcast-enabled Unified Plan client capable of a single media source per media type. The client can send a simulcast of 2 video resolutions and frame rates: HD 1280x720p 30fps and thumbnail 320x180p 15fps. Alice's Offer:

o=alice 2362969037 2362969040 IN IP4
s=Simulcast Enabled Unified Plan Client
t=0 0
c=IN IP4
m=audio 49200 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 49300 RTP/AVP 97 98
a=rtpmap:97 H264/90000
a=rtpmap:98 H264/90000
a=fmtp:97 profile-level-id=42c01f; max-fs=3600; max-mbps=108000
a=fmtp:98 profile-level-id=42c00b; max-fs=240; max-mbps=3600
a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720]
a=imageattr:98 send [x=320,y=180] recv [x=320,y=180]
a=simulcast send 97;98 recv 97

Figure 3: Unified Plan Simulcast Offer

The only thing in the SDP that indicates simulcast capability is the line in the video media description containing the "simulcast" attribute. The included format parameters indicates that sent simulcast versions can differ in video resolution and framerate.

The Answer from the server indicates that it too is simulcast capable. Should it not have been simulcast capable, the "a=simulcast" line would not have been present and communication would have started with the media negotiated in the SDP.

o=server 823479283 1209384938 IN IP4
s=Answer to Simulcast Enabled Unified Plan Client
t=0 0
c=IN IP4
m=audio 49672 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 49674 RTP/AVP 97 98
a=rtpmap:97 H264/90000
a=rtpmap:98 H264/90000
a=fmtp:97 profile-level-id=42c01f; max-fs=3600; max-mbps=108000
a=fmtp:98 profile-level-id=42c00b; max-fs=240; max-mbps=3600
a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720]
a=imageattr:98 send [x=320,y=180] recv [x=320,y=180]
a=simulcast recv 97;98 send 97

Figure 4: Unified Plan Simulcast Answer

Since the server is the simulcast media receiver, it reverses the direction of the "simulcast" attribute.

6.3.2. Multi-Source Client

Fred is calling in to the same conference as in the example above with a two-camera, two-display system, thus capable of handling two separate media sources in each direction, where each media source is simulcast-enabled in the send direction. Fred's client is a Unified Plan client, restricted to a single media source per media description.

The first two simulcast versions for the first media source use different codecs, H264-SVC [RFC6190] and H264 [RFC6184]. These two simulcast versions also have a temporal dependency. Two different video codecs, VP8 [I-D.ietf-payload-vp8] and H264, are offered as alternatives for the third simulcast version for the first media source.

The second media source is offered with three different simulcast versions. All video streams of this second media source are loss protected by RTP retransmission [RFC4588].

Fred's client is also using BUNDLE to send all RTP streams from all media descriptions in the same RTP session on a single media transport. There are not so many RTP payload types in this example that there is any risk of running out of payload types, but for the sake of making an example, it is assumed that one of the payload types cannot be kept unique across all media descriptions. Therefore, the SDP makes use of the mechanism (work in progress) in BUNDLE that identifies which media description an RTP stream belongs to (a new RTCP SDES item and RTP header extension [RFC5285] type carrying the a=mid value). That identification will make it possible to identify unambiguously also on RTP level which media source it is and thus what the related simulcast versions are, even though two separate RTP streams in the joint RTP session share RTP payload type.

o=fred 238947129 823479223 IN IP4
s=Offer from Simulcast Enabled Multi-Source Client
t=0 0
c=IN IP4
a=group:BUNDLE foo bar zen

m=audio 49200 RTP/AVP 99
a=rtpmap:99 G722/8000

m=video 49600 RTP/AVP 100 101 102 103
a=rtpmap:100 H264-SVC/90000
a=rtpmap:101 H264/90000
a=rtpmap:102 H264/90000
a=rtpmap:103 VP8/90000
a=fmtp:100 profile-level-id=42400d; max-fs=3600; max-mbps=108000; \
a=fmtp:101 profile-level-id=42c00d; max-fs=3600; max-mbps=54000
a=fmtp:102 profile-level-id=42c00d; max-fs=900; max-mbps=27000
a=fmtp:103 max-fs=900; max-fr=30
a=imageattr:100 send [x=1280,y=720] recv [x=1280,y=720]
a=imageattr:101 send [x=1280,y=720] recv [x=1280,y=720]
a=imageattr:102 send [x=640,y=360] recv [x=640,y=360]
a=imageattr:103 send [x=640,y=360] recv [x=640,y=360]
a=depend:100 lay bar:101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=simulcast sendrecv 100;101 send 103,102

m=video 49602 RTP/AVP 96 103 97 104 105 106
a=rtpmap:96 VP8/90000
a=fmtp:96 max-fs=3600; max-fr=30
a=rtpmap:104 rtx/90000
a=fmtp:104 apt=96;rtx-time=200
a=rtpmap:103 VP8/90000
a=fmtp:103 max-fs=900; max-fr=30
a=rtpmap:105 rtx/90000
a=fmtp:105 apt=103;rtx-time=200
a=rtpmap:97 VP8/90000
a=fmtp:97 max-fs=240; max-fr=15
a=rtpmap:106 rtx/90000
a=fmtp:106 apt=97;rtx-time=200
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=simulcast send 97;96;103

Figure 5: Fred's Multi-Source Simulcast Offer

  • Note: Empty lines in the SDP above are added only for readability and would not be present in an actual SDP.

7. Network Aspects

Simulcast is in this memo defined as the act of sending multiple alternative encoded streams of the same underlying media source. When transmitting multiple independent streams that originate from the same source, it could potentially be done in several different ways using RTP. A general discussion on considerations for use of the different RTP multiplexing alternatives can be found in Guidelines for Multiplexing in RTP [I-D.ietf-avtcore-multiplex-guidelines]. Discussion and clarification on how to handle multiple streams in an RTP session can be found in [I-D.ietf-avtcore-rtp-multi-stream].

The network aspects that are relevant for simulcast are:

Quality of Service:
When using simulcast it might be of interest to prioritize a particular simulcast version, rather than applying equal treatment to all versions. For example, lower bit-rate versions may be prioritized over higher bit-rate versions to minimize congestion or packet losses in the low bit-rate versions. Thus, there is a benefit to use a simulcast solution that supports QoS as good as possible. By separating simulcast versions into different RTP sessions and send those RTP sessions over different media transports, a simulcast version can be prioritized by existing flow based QoS mechanisms. When using unicast, QoS mechanisms based on individual packet marking are also feasible, which do not require separation of simulcast versions into different RTP sessions to apply different QoS. The proposed solution can be extended to support this functionality with an optional mid: prefix before the RTP payload types of a simulcast version, to describe simulcast across multiple media descriptions.
NAT/FW Traversal:
Using multiple RTP sessions will incur more cost for NAT/FW traversal unless they can re-use the same transport flow, which can be achieved by either one of multiplexing multiple RTP sessions on a single lower layer transport [I-D.westerlund-avtcore-transport-multiplexing] or Multiplexing Negotiation Using SDP Port Numbers [I-D.ietf-mmusic-sdp-bundle-negotiation]. If flow based QoS with any differentiation is desirable, the cost for additional transport flows is likely necessary.
Multiple RTP sessions will be required to enable combining simulcast with multicast. Different simulcast versions have to be separated to different multicast groups to allow a multicast receiver to pick the version it wants, rather than receive all of them. In this case, the only reasonable implementation is to use different RTP sessions for each multicast group so that reporting and other RTCP functions operate as intended. The proposed solution can be extended to support this functionality with an optional mid: prefix before the RTP payload types of a simulcast version, to describe simulcast across multiple media descriptions.

8. IANA Considerations

This document requests to register a new attribute, simulcast.

Formal registrations to be written.

9. Security Considerations

The simulcast capability and configuration attributes and parameters are vulnerable to attacks in signaling.

A false inclusion of the "a=simulcast" attribute may result in simultaneous transmission of multiple RTP streams that would otherwise not be generated. The impact is limited by the media description joint bandwidth, shared by all simulcast versions irrespective of their number. There may however be a large number of unwanted RTP streams that will impact the share of the bandwidth allocated for the originally wanted RTP stream.

A hostile removal of the "a=simulcast" attribute will result in simulcast not being used.

Neither of the above will likely have any major consequences and can be mitigated by signaling that is at least integrity and source authenticated to prevent an attacker to change it.

10. Contributors

Morgan Lindqvist and Fredrik Jansson, both from Ericsson, have contributed with important material to the first versions of this document. Robert Hansen, from Cisco, contributed significantly to subsequent versions.

11. Acknowledgements

12. References

12.1. Normative References

[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.
[RFC4566] Handley, M., Jacobson, V. and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error Correction", RFC 5109, December 2007.
[RFC5234] Crocker, D. and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", STD 68, RFC 5234, January 2008.
[RFC7104] Begen, A., Cai, Y. and H. Ou, "Duplication Grouping Semantics in the Session Description Protocol", RFC 7104, January 2014.

12.2. 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", Internet-Draft draft-ietf-avtcore-multiplex-guidelines-03, October 2014.
[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", Internet-Draft draft-ietf-avtcore-rtp-multi-stream-06, October 2014.
[I-D.ietf-avtcore-rtp-topologies-update] Westerlund, M. and S. Wenger, "RTP Topologies", Internet-Draft draft-ietf-avtcore-rtp-topologies-update-05, November 2014.
[I-D.ietf-avtext-rtp-grouping-taxonomy] Lennox, J., Gross, K., Nandakumar, S. and G. Salgueiro, "A Taxonomy of Grouping Semantics and Mechanisms for Real-Time Transport Protocol (RTP) Sources", Internet-Draft draft-ietf-avtext-rtp-grouping-taxonomy-04, January 2015.
[I-D.ietf-avtext-rtp-stream-pause] Akram, A., Even, R. and M. Westerlund, "RTP Stream Pause and Resume", Internet-Draft draft-ietf-avtext-rtp-stream-pause-05, October 2014.
[I-D.ietf-mmusic-sdp-bundle-negotiation] Holmberg, C., Alvestrand, H. and C. Jennings, "Negotiating Media Multiplexing Using the Session Description Protocol (SDP)", Internet-Draft draft-ietf-mmusic-sdp-bundle-negotiation-15, January 2015.
[I-D.ietf-payload-vp8] Westin, P., Lundin, H., Glover, M., Uberti, J. and F. Galligan, "RTP Payload Format for VP8 Video", Internet-Draft draft-ietf-payload-vp8-13, October 2014.
[I-D.roach-mmusic-unified-plan] Roach, A., Uberti, J. and M. Thomson, "A Unified Plan for Using SDP with Large Numbers of Media Flows", Internet-Draft draft-roach-mmusic-unified-plan-00, July 2013.
[I-D.westerlund-avtcore-transport-multiplexing] Westerlund, M. and C. Perkins, "Multiplexing Multiple RTP Sessions onto a Single Lower-Layer Transport", Internet-Draft draft-westerlund-avtcore-transport-multiplexing-07, 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.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, June 2002.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)", RFC 3389, September 2002.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V. and R. Hakenberg, "RTP Retransmission Payload Format", RFC 4588, July 2006.
[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals", RFC 4733, December 2006.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, January 2008.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP Header Extensions", RFC 5285, July 2008.
[RFC5576] Lennox, J., Ott, J. and T. Schierl, "Source-Specific Media Attributes in the Session Description Protocol (SDP)", RFC 5576, June 2009.
[RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding Dependency in the Session Description Protocol (SDP)", RFC 5583, July 2009.
[RFC6184] Wang, Y., Even, R., Kristensen, T. and R. Jesup, "RTP Payload Format for H.264 Video", RFC 6184, May 2011.
[RFC6190] Wenger, S., Wang, Y., Schierl, T. and A. Eleftheriadis, "RTP Payload Format for Scalable Video Coding", RFC 6190, May 2011.
[RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image Attributes in the Session Description Protocol (SDP)", RFC 6236, May 2011.

Appendix A. Changes From Earlier Versions

NOTE TO RFC EDITOR: Please remove this section prior to publication.

A.1. Modifications Between Individual Version -00 and WG Version -00

  • Added this appendix.

Authors' Addresses

Bo Burman Ericsson Kistavagen 25 SE-164 80 Stockholm, Sweden Phone: +46 10 714 13 11 EMail:
Magnus Westerlund Ericsson Farogatan 6 SE-164 80 Stockholm, Sweden Phone: +46 10 714 82 87 EMail:
Suhas Nandakumar Cisco 170 West Tasman Drive San Jose, CA 95134 USA EMail:
Mo Zanaty Cisco 170 West Tasman Drive San Jose, CA 95134 USA EMail: