Network Working Group B. Burman
Internet-Draft M. Westerlund
Intended status: Standards Track Ericsson
Expires: April 21, 2016 S. Nandakumar
M. Zanaty
Cisco
October 19, 2015

Using Simulcast in SDP and RTP Sessions
draft-ietf-mmusic-sdp-simulcast-03

Abstract

In some application scenarios it may be desirable to send multiple differently encoded versions of the same media source in different 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 of a new media level SDP attribute that expresses capability to send and/or receive simulcast RTP streams. RTP/RTCP identification using either payload types or a separately defined method for RTP stream configuration are defined.

Status of This Memo

This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.

Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at http://datatracker.ietf.org/drafts/current/.

Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."

This Internet-Draft will expire on April 21, 2016.

Copyright Notice

Copyright (c) 2015 IETF Trust and the persons identified as the document authors. All rights reserved.

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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(s) 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].

This document describes a few scenarios where it is motivated to use simulcast, and also defines the needed SDP signaling for it.

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 Stream:
One Encoded Stream or Dependent Stream from a set of concurrently transmitted Encoded Streams and optional Dependent Streams, all sharing a common Media Source, as defined in [I-D.ietf-avtext-rtp-grouping-taxonomy]. Decoding a Dependent Stream also requires the related (Dependent and) Encoded Stream(s), but in the context of simulcast that is considered a property of the Dependent Stream constituting the simulcast stream. For example, HD and thumbnail video simulcast versions of a single Media Source sent concurrently as separate RTP Streams.
Simulcast Format:
Different formats of a simulcast stream serve the same purpose as alternative RTP payload types in non-simulcast SDP, to allow multiple alternative media formats for a given RTP Stream. As for multiple RTP payload types on the m-line, any one of the alternative formats can be used at a given point in time, but not more than one (based on RTP timestamp), and what format is used can change dynamically from one RTP packet to another. For example, if all participants in a group video call can decode H.264 and H.265 video, but only some can encode H.265, both H.264 and H.265 can be kept as alternative formats, and the format may dynamically switch between H.264 and H.265 as different participants become active speaker.

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.

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:

Codec:
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.
Sampling:
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.
Bitrate:
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 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:

REQ-1:
Identification. It must be possible to identify a set of simulcasted RTP streams as originating from the same media source:
REQ-1.1:
In SDP signaling.
REQ-1.2:
On RTP/RTCP level.
REQ-2:
Transport usage. The solution must work when using:
REQ-2.1:
Legacy SDP with separate media transports per SDP media description.
REQ-2.2:
Bundled [I-D.ietf-mmusic-sdp-bundle-negotiation] SDP media descriptions.
REQ-3:
Capability negotiation. It must be possible that:
REQ-3.1:
Sender can express capability of sending simulcast.
REQ-3.2:
Receiver can express capability of receiving simulcast.
REQ-3.3:
Sender can express maximum number of simulcast streams that can be provided.
REQ-3.4:
Receiver can express maximum number of simulcast streams that can be received.
REQ-3.5:
Sender can detail the characteristics of the simulcast streams that can be provided.
REQ-3.6:
Receiver can detail the characteristics of the simulcast streams that it prefers to receive.
REQ-4:
Distinguishing features. It must be possible to have different simulcast streams use different codec parameters, as can be expressed by SDP format values and RTP payload types.
REQ-5:
Compatibility. It must be possible to use simulcast in combination with other RTP mechanisms that generate additional RTP streams:
REQ-5.1:
RTP Retransmission [RFC4588].
REQ-5.2:
RTP Forward Error Correction [RFC5109].
REQ-5.3:
Related payload types such as audio Comfort Noise and/or DTMF.
REQ-6:
Interoperability. The solution must be possible to use in:
REQ-6.1:
Interworking with non-simulcast legacy clients using a single media source per media type.
REQ-6.2:
WebRTC "Unified Plan" environment with a single media source per SDP media description.

5. Overview

As an overview, the above requirements are met by signaling simulcast capability and configurations in SDP [RFC4566]:

6. Detailed Description

This section further details the overview 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), the simulcast attribute defines a list of simulcast streams (separated by semicolons), each of which is a list of simulcast formats (separated by commas). The meaning of the attribute on SDP session level is undefined and MUST NOT be used. The ABNF [RFC5234] for this attribute is:

sc-attr     = "a=simulcast:" 1*2( WSP sc-str-list ) [WSP sc-pause-list]
sc-str-list = sc-dir WSP sc-id-type "=" sc-alt-list *( ";" sc-alt-list )
sc-pause-list = "paused=" sc-alt-list
sc-dir      = "send" / "recv"
sc-id-type  = "pt" / "rid" / token
sc-alt-list = sc-id *( "," sc-id )
sc-id       = fmt / rid-identifier / token
; WSP defined in [RFC5234]
; fmt, token defined in [RFC4566]
; rid-identifier defined in [I-D.pthatcher-mmusic-rid]

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 on the same line.

Two simulcast stream identification methods are defined; "pt" using RTP payload type (SDP format), and "rid" using an additional RTP-level identification mechanism [I-D.pthatcher-mmusic-rid]. Different identification methods MUST NOT be used for different directions on a single "a=simulcast" line. Implementations that support both identification methods MAY include one "a=simulcast" line for each identification method for the same "m="-line. Multiple "a=simulcast" lines with the same identification method MUST NOT be used for a single "m="-line.

Attribute parameters are grouped by direction and consist of a listing of simulcast stream identifications to be used. The number of (non-alternative, see below) identifications in the list sets a limit to the number of supported simulcast streams in that direction. The order of the listed simulcast versions in the "send" direction suggests a proposed order of preference, in decreasing order: the stream listed first is the most preferred Section 3.1, and subsequent streams have progressively lower preference. The order of the listed simulcast streams in the "recv" direction expresses a preference which simulcast streams that are preferred, with the leftmost being most preferred. This can be of importance if the number of actually sent simulcast streams have to be reduced for some reason.

Formats that have explicit dependencies [RFC5583] [I-D.pthatcher-mmusic-rid] to other formats (even in the same media description) MAY be listed as different simulcast streams.

Alternative simulcast formats MAY be specified as part of the attribute parameters by expressing each simulcast stream as a comma-separated list of alternative format identifiers. In this case, there MUST NOT be any capability restriction in what alternative formats can be used across different simulcast streams, like requiring all simulcast streams to use the same codec format alternative. The order of the format alternatives within a simulcast stream is significant; the alternatives are listed from (left) most preferred to (right) least preferred. For the use of simulcast, this overrides the normal codec preference as expressed by format type ordering on the "m="-line, using regular SDP rules. This is to enable a separation of general codec preferences and simulcast stream configuration preferences.

A simulcast stream 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 streams of the media source, but rather a specific way of generating the RTP stream of a single simulcast stream with varying RTP payload type. Instead, only a single simulcast stream identification MUST be used per simulcast stream or alternative simulcast format (if there are such) in the SDP. The used simulcast stream identification SHOULD be the codec format 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.

If RTP stream pause/resume [I-D.ietf-avtext-rtp-stream-pause] is supported, the optional "paused=" parameter MAY be used in conjunction with "rid" simulcast stream identification to specify that a certain simulcast stream is initially paused already from start of the RTP session. In this case, support for RTP stream pause/resume MUST also be included under the same "m="-line listing "a=simulcast". Initially paused simulcast streams MUST NOT be used with "pt" identification. Initially paused simulcast streams are resumed as described by the RTP pause/resume specification.

An initially paused simulcast stream in "send" direction MUST be considered equivalent to an unsolicited locally paused stream, and be handled accordingly.

An initially paused simulcast stream in "recv" direction SHOULD cause the remote RTP sender to put the stream as unsolicited locally paused, unless there are other RTP stream receivers that do not mark the simulcast stream as initially paused. The reason to require an initially paused "recv" stream to be considered locally paused by the remote RTP sender, instead of making it equivalent to implicitly sending a pause request, is because the pausing RTP sender cannot know which SSRC owns the restriction when TMMBR/TMMBN are used for pause/resume signaling since the RTP receiver's SSRC in send direction is not known yet.

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 stream 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.

If multiple simulcast formats 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 stream.

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 simulcast stream identifications 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. Note that, like all other use of SDP format tags ("pt:") for the send direction in Offer/Answer, format tags related to the simulcast stream identification send direction in an offer 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 streams and/or alternative formats in the offer MUST be prepared to receive RTP streams for any of those simulcast streams and/or alternative formats from the answerer.

An answerer that receives an offer with simulcast containing an "a=simulcast" attribute listing alternative formats for simulcast streams MAY keep all the alternatives in the answer, but it MAY also choose to remove any non-desirable alternatives per simulcast stream 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 streams, MAY reduce the number of simulcast streams in the answer, but MUST NOT add simulcast streams.

An offerer that receives an answer where some simulcast formats 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. Similarly, the answerer MUST be prepared to receive any of the kept receive direction alternatives, and MAY send any of the kept send direction alternatives in the answer.

The offerer and answerer MUST NOT send more than a single alternative format at a time (based on RTP timestamps) per simulcast stream, but MAY change format on a per-RTP packet basis. This corresponds to the existing (non-simulcast) SDP offer/answer case when multiple formats are included on the "m="-line in the SDP answer.

An offerer that receives an answer where some of the simulcast streams 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 streams.

Simulcast streams or formats using undefined simulcast stream identifications MUST NOT be used as valid simulcast streams by an RTP stream receiver.

An offerer that is capable of using both simulcast stream identification methods MAY include one "a=simulcast" line per identification method in the offer. Note that it is in general not expected that the "pt" identification method will provide feature parity with the "rid" method, and the different "a=simulcast" lines can therefore express different use of simulcast functionality. However, for some configurations the different identification methods can be equivalent.

An answerer receiving an offer listing both simulcast stream identification methods MUST choose only one and remove the other from the answer. An answerer not supporting a simulcast stream identification method in the offer MUST remove the non-supported "a=simulcast" line from the answer, possibly falling back to not using simulcast at all.

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 Streams

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 and (optionally) RID [I-D.pthatcher-mmusic-rid], as specified in the SDP "a=simulcast" attribute [sec-cap] parameters. When using BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation] with 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 and RID relations can be used.

BUNDLE's MID is an RTCP source description (SDES) item. To ensure rapid initial reception, required to correctly process the RTP streams, it is also defined as an RTP header extension [RFC5285].

6.3. Signaling Examples

These examples describe a 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. This is defined below using the "imageattr" [RFC6236]. Media formats (RTP payload types) are used as simulcast stream identification. Alice's Offer:

v=0
o=alice 2362969037 2362969040 IN IP4 192.0.2.156
s=Simulcast Enabled Unified Plan Client
t=0 0
c=IN IP4 192.0.2.156
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 pt=97;98 recv pt=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 streams can differ in video resolution.

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.

v=0
o=server 823479283 1209384938 IN IP4 192.0.2.2
s=Answer to Simulcast Enabled Unified Plan Client
t=0 0
c=IN IP4 192.0.2.43
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 pt=97;98 send pt=97

Figure 4: Unified Plan Simulcast Answer

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

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 restricted to a single media source per media description.

The first two simulcast streams for the first media source use different codecs, H264-SVC [RFC6190] and H264 [RFC6184]. These two simulcast streams 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 stream for the first media source. RID is used as simulcast stream identification, reducing the number of media formats needed. Only the highest fidelity simulcast stream are sent from start, the lower fidelity streams being initially paused.

The second media source is offered with three different simulcast streams. All video streams of this second media source are loss protected by RTP retransmission [RFC4588]. RID is used as simulcast stream identification. Also here, all but the highest fidelity simulcast stream are initially paused.

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. Although using many different simulcast streams in this example, use of RID as simulcast stream identification enables use of a low number of RTP payload types. Note that the use of both BUNDLE and RID recommends using the RTP header extension [RFC5285] for carrying these fields.

v=0
o=fred 238947129 823479223 IN IP4 192.0.2.125
s=Offer from Simulcast Enabled Multi-Source Client
t=0 0
c=IN IP4 192.0.2.125
a=group:BUNDLE foo bar zen

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

m=video 49600 RTP/AVPF 100 101 103
a=mid:bar
a=rtpmap:100 H264-SVC/90000
a=rtpmap:101 H264/90000
a=rtpmap:103 VP8/90000
a=fmtp:100 profile-level-id=42400d; max-fs=3600; max-mbps=108000; \
    mst-mode=NI-TC
a=fmtp:101 profile-level-id=42c00d; max-fs=3600; max-mbps=54000
a=fmtp:103 max-fs=900; max-fr=30
a=rid:1 send pt=100;max-width=1280;max-height=720;max-fr=60;depend=2
a=rid:2 send pt=101;max-width=1280;max-height=720;max-fr=30
a=rid:3 send pt=101;max-width=640;max-height=360
a=rid:4 send pt=103;max-width=640;max-height=360
a=depend:100 lay bar:101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:rid
a=rtcp-fb:* ccm pause nowait
a=simulcast: send rid=1;2;4,3 paused=2,3,4

m=video 49602 RTP/AVPF 96 104
a=mid:zen
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=rid:5 send pt=96;max-fs=921600;max-fr=30
a=rid:6 send pt=96;max-fs=614400;max-fr=15
a=rid:7 send pt=96;max-fs=230400;max-fr=30
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:rid
a=rtcp-fb:* ccm pause nowait
a=simulcast: send rid=5;6;7 paused=6,7

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 stream, rather than applying equal treatment to all streams. For example, lower bit-rate streams may be prioritized over higher bit-rate streams to minimize congestion or packet losses in the low bit-rate streams. Thus, there is a benefit to use a simulcast solution that supports QoS as good as possible.
NAT/FW Traversal:
Using multiple RTP sessions incurs more cost for NAT/FW traversal unless they can re-use the same transport flow, which can be achieved by Multiplexing Negotiation Using SDP Port Numbers [I-D.ietf-mmusic-sdp-bundle-negotiation].

8. Limitations

The chosen approach has a few limitations that are described in this section. Some relate to the use of a single RTP session for all simulcast formats of a media source, while others relate to the two different simulcast stream identification methods.

8.1. Single RTP Session

The limitations in this section come from sending all simulcast streams related to a media source under the same SDP media description, which also means they are sent in the same RTP session.

It is not possible to use different simulcast streams on different transports, limiting the possibilities to apply different QoS to different simulcast streams. When using unicast, QoS mechanisms based on individual packet marking are feasible, since they do not require separation of simulcast streams into different RTP sessions to apply different QoS.

It is not possible to separate different simulcast streams into different multicast groups to allow a multicast receiver to pick the stream 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.

8.2. SDP Format Identification

The limitations in this section come from and thus apply only when using SDP format (RTP payload type) as simulcast stream identification method.

The available RTP payload type number space may not be sufficient when many different media formats and/or simulcast streams are used in the SDP. This can be particularly prominent when BUNDLE is used, and for any technology that adds to the number of required RTP payload types in a multiplicative way, such as for example adding RTP retransmission [RFC4588] and Forward Error Correction [RFC5109]. Flexible FEC Scheme [I-D.ietf-payload-flexible-fec-scheme] can be used for RTP retransmissions and would avoid the double consumption of the PT space that RTP Retransmission [RFC4588] causes.

Only existing SDP attributes and parameters can be used to define codec configuration for a simulcast format. Any codec that does not define a sufficient set of codec parameters in "a=fmtp", or can make use of other SDP attributes, may not be capable of expressing the desired simulcast format dimensions [sec-diverse-receivers] with necessary precision, or not at all. One example of this is the ability to separate simulcast formats by bandwidth for codecs lacking a codec-specific bandwidth parameter, since the SDP "b="-line covers all RTP payload types listed on an "m="-line.

A simulcast stream signaled as initially paused is not possible to resume by a remote peer, because it cannot know which target SSRC to use in the RESUME message [I-D.ietf-avtext-rtp-stream-pause].

8.3. RID Identification

The limitations in this section come from and thus apply only when using RID as simulcast stream identification method.

Use of the additional "a=rid"-line in SDP and the corresponding RID RTCP SDES item and RTP header extension requires some additional implementation complexity, and incurs some extra bandwidth cost to carry the RID RTCP SDES item and RTP header extension.

9. IANA Considerations

This document requests to register a new SDP attribute, simulcast.

Formal registrations to be written.

10. Security Considerations

The simulcast capability, 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 streams irrespective of their number. There may however be a large number of unwanted RTP streams that will impact the share of 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.

11. Contributors

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

12. Acknowledgements

13. References

13.1. Normative References

[I-D.ietf-avtext-rtp-stream-pause] Burman, B., Akram, A., Even, R. and M. Westerlund, "RTP Stream Pause and Resume", Internet-Draft draft-ietf-avtext-rtp-stream-pause-10, September 2015.
[I-D.pthatcher-mmusic-rid] Thatcher, P., Zanaty, M., Nandakumar, S., Burman, B., Roach, A. and B. Campen, "RTP Payload Format Constraints", Internet-Draft draft-pthatcher-mmusic-rid-02, October 2015.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997.
[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.
[RFC4566] Handley, M., Jacobson, V. and C. Perkins, "SDP: Session Description Protocol", RFC 4566, DOI 10.17487/RFC4566, July 2006.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error Correction", RFC 5109, DOI 10.17487/RFC5109, December 2007.
[RFC5234] Crocker, D. and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", STD 68, RFC 5234, DOI 10.17487/RFC5234, January 2008.
[RFC7104] Begen, A., Cai, Y. and H. Ou, "Duplication Grouping Semantics in the Session Description Protocol", RFC 7104, DOI 10.17487/RFC7104, January 2014.

13.2. 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-09, September 2015.
[I-D.ietf-avtcore-rtp-topologies-update] Westerlund, M. and S. Wenger, "RTP Topologies", Internet-Draft draft-ietf-avtcore-rtp-topologies-update-10, 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", Internet-Draft draft-ietf-avtext-rtp-grouping-taxonomy-08, 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)", Internet-Draft draft-ietf-mmusic-sdp-bundle-negotiation-23, July 2015.
[I-D.ietf-payload-flexible-fec-scheme] Singh, V., Begen, A., Zanaty, M. and G. Mandyam, "RTP Payload Format for Flexible Forward Error Correction (FEC)", Internet-Draft draft-ietf-payload-flexible-fec-scheme-01, October 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-vp8-17, September 2015.
[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.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, DOI 10.17487/RFC3264, June 2002.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389, September 2002.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V. and R. Hakenberg, "RTP Retransmission Payload Format", RFC 4588, DOI 10.17487/RFC4588, July 2006.
[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals", RFC 4733, DOI 10.17487/RFC4733, December 2006.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, DOI 10.17487/RFC5117, January 2008.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July 2008.
[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.
[RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding Dependency in the Session Description Protocol (SDP)", RFC 5583, DOI 10.17487/RFC5583, July 2009.
[RFC6184] Wang, Y., Even, R., Kristensen, T. and R. Jesup, "RTP Payload Format for H.264 Video", RFC 6184, DOI 10.17487/RFC6184, May 2011.
[RFC6190] Wenger, S., Wang, Y., Schierl, T. and A. Eleftheriadis, "RTP Payload Format for Scalable Video Coding", RFC 6190, DOI 10.17487/RFC6190, May 2011.
[RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image Attributes in the Session Description Protocol (SDP)", RFC 6236, DOI 10.17487/RFC6236, May 2011.

Appendix A. Changes From Earlier Versions

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

A.1. Modifications Between WG Version -02 and -03

  • Removed text on multicast / broadcast from use cases, since it is not supported by the solution.
  • Removed explicit references to unified plan draft.
  • Added possibility to initiate simulcast streams in paused mode.
  • Enabled an offerer to offer multiple stream identification (pt or rid) methods and have the answerer choose which to use.
  • Added a preference indication also in send direction offers.
  • Added a section on limitations of the current proposal, including identification method specific limitations.

A.2. Modifications Between WG Version -01 and -02

  • Relying on the new RID solution for codec constraints and configuration identification. This has resulted in changes in syntax to identify if pt or RID is used to describe the simulcast stream.
  • Renamed simulcast version and simulcast version alternative to simulcast stream and simulcast format respectively, and improved definitions for them.
  • Clarification that it is possible to switch between simulcast version alternatives, but that only a single one be used at any point in time.
  • Changed the definition so that ordering of simulcast formats for a specific simulcast stream do have a preference order.

A.3. Modifications Between WG Version -00 and -01

  • No changes. Only preventing expiry.

A.4. Modifications Between Individual Version -00 and WG Version -00

  • Added this appendix.

Authors' Addresses

Bo Burman Ericsson Kistavagen 25 SE-164 80 Stockholm, Sweden EMail: bo.burman@ericsson.com
Magnus Westerlund Ericsson Farogatan 2 SE-164 80 Stockholm, Sweden Phone: +46 10 714 82 87 EMail: magnus.westerlund@ericsson.com
Suhas Nandakumar Cisco 170 West Tasman Drive San Jose, CA 95134 USA EMail: snandaku@cisco.com
Mo Zanaty Cisco 170 West Tasman Drive San Jose, CA 95134 USA EMail: mzanaty@cisco.com