ETSI TR 126 980 V17.0.0 (2022-05)

Universal Mobile Telecommunications System (UMTS); LTE; 5G; Multimedia telephony over IP Multimedia Subsystem (IMS); Media handling aspects of multi-stream multiparty conferencing for Multimedia Telephony Service for IMS (MTSI) (3GPP TR 26.980 version 17.0.0 Release 17)

RTR/TSGS-0426980vh00

General Information

• Status: Not Published
• Technical Committee: 3GPP SA 4 - Codec
• Current Stage: 12 - Completion
• Completion Date: 04-May-2022

Standards Content (Sample)

TECHNICAL REPORT
Universal Mobile Telecommunications System (UMTS);
LTE;
5G;
Multimedia telephony over IP Multimedia Subsystem (IMS);
Media handling aspects of multi-stream multiparty
conferencing for Multimedia Telephony Service for IMS (MTSI)
(3GPP TR 26.980 version 17.0.0 Release 17)

3GPP TR 26.980 version 17.0.0 Release 17 1 ETSI TR 126 980 V17.0.0 (2022-05)

Reference
RTR/TSGS-0426980vh00
Keywords
5G,LTE,UMTS
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ETSI
3GPP TR 26.980 version 17.0.0 Release 17 2 ETSI TR 126 980 V17.0.0 (2022-05)
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ETSI
3GPP TR 26.980 version 17.0.0 Release 17 3 ETSI TR 126 980 V17.0.0 (2022-05)
Contents
Intellectual Property Rights . 2
Legal Notice . 2
Modal verbs terminology . 2
Foreword . 6
Introduction . 6
1 Scope . 7
2 References . 7
3 Definitions and abbreviations . 9
3.1 Definitions . 9
3.2 Abbreviations . 10
4 Overview . 10
5 Media Handling in Current 3GPP Conferencing . 10
6 Use cases . 12
6.1 Overview . 12
6.2 Use case A: Transcoding Free Continuous Presence. 12
6.2.1 Problem Description . 13
6.2.2 Proposed Solution . 13
6.3 Use case B: Screen Sharing . 14
6.3.1 Problem Description . 14
6.3.2 Proposed Solution . 15
6.4 Use Case C: Bandwidth Handling . 17
6.4.1 Problem Description . 17
6.4.2 Proposed Solution . 17
6.4.2.1 Single-stream to multi-stream . 18
6.4.2.2 Multi-stream to single-stream . 18
6.4.2.3 Multi-stream to multi-stream without bandwidth restriction . 18
6.4.2.4 Multi-stream to multi-stream with minor bandwidth restrictions. 19
6.4.2.5 Multi-stream to multi-stream with severe bandwidth restriction . 20
6.4.2.6 Multi-stream UE to multi-stream UE in point-to-point . 20
6.5 Use Case D: Active Speaker Override . 21
6.5.1 Problem Description . 21
6.5.2 Proposed Solution . 22
6.6 Use Case E: Pausing Unused Streams . 23
6.6.1 Problem Description . 24
6.6.2 Proposed Solution . 25
6.7 Use Case F: Conference Rate Adaptation Considerations . 26
6.7.1 Problem Description . 26
6.7.2 Proposed Solution . 26
6.8 Use Case G: Multi-stream Audio Steering or Panning . 29
6.9 Use Case H: De-jitter Buffer Handling . 30
6.10 Use Case I: Audio Spatialization based on head-tracking . 31
6.11 Use Case J: Mixing at the Rendering Device – Media Handling, Distribution via Multi-Unicast . 32
6.11.1 Concurrent Codec Capabilities Exc hange . 32
6.11.1.1 Concurrent Decoding . 32
6.11.1.2 Concurrent Encoding . 33
6.11.1.3 Further Considerations in Concurrency . 34
6.11.1.4 Prioritizing and Ignoring of Received Media Streams . 34
6.11.1.5 Conclusions . 34
6.12 Use Case K: Mixing at the Rendering Device – Media Handling, Distribution via Multicast . 35
6.12.1 Session Establishment. 35
6.12.1.1 In the presence of a Conference Focus . 35
6.12.2 Media Handling . 35
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6.12.2.1 In the absence of a Conference Focus for media handling . 35
6.13 Use Case L: Mixing at the Rendering Device – Media Handling, Distribution via Single Source Multi-

Unicas t . 36
6.13.1 Session Establishment. 38
6.13.2 Media Handling . 38
6.13.3 Multi-stream Concurrent Encoding/Decoding Capabilities . 38
6.13.3.1 Querying and Capabilities Exchange – SIP OPTIONS . 39
6.13.3.2 Common/Preferred Codec Identification and Usage . 42
6.13.3.2.1 General . 42
6.13.3.2.2 Common/Preferred Codec Information Exchange. 42
6.13.3.2.3 Indicating the Common and Preferred Codec Selection Through SDP . 43
6.13.4 MSMTSI MRF Handling with Reduced m-lines . 43
6.13.4.1 Introduction . 43
6.13.4.2 RTP Stream Selective Forwarding . 43
6.14 Use Case M: Provisioning of Talker ID . 45
6.14.1 Use Case Description . 45
6.14.2 Problem Description . 45
6.14.3 Suggested Solution . 46
6.15 Use Case N: Mixing at the Rendering Device – Media Handling, Concurrent Codec Capabilities
Exchange . 47
6.15.1 Format of the Concurrent Codec Capabilities Information . 47
6.15.1.1 Using Current SDP Parameters . 47
6.15.1.2 Using Compact SDP Parameters . 48
6.15.1.2.1 SDP Line Compression . 49
6.15.1.2.2 SDP Line Compression for Terminal Performing Trimming of Streams . 51
6.15.1.2.3 Conclusions . 51
6.15.2 Protocol for Concurrent Codec Capabilities Exchange (CCCEx). 51
6.15.2.1 Recommended Requirements for the CCCEx . 51
6.15.2.2 Potential Solution for the CCCEx . 51
6.15.3 Examples of Concurrent Codec Capabilities (CCC) Usage . 51
6.15.4 Compact CCC SDP Parameter for Session Initiation . 57
6.15.4.1 Introduction . 57
6.15.4.2 Compression Gains . 57
6.15.4.3 Offer-Answer Rules . 60
6.15.4.4 Conclusions . 61
6.16 Use Case O: Media Distribution Without Conference Focus – Session Establishment Aspects . 61
6.16.1 Session Establishment Without a Conference Focus in Multi-unicast Topology . 61
6.17 Use Case P: Codec Migration. 62
6.17.1 General . 62
6.17.2 Problem Description . 62
6.17.3 Proposed Solution . 62
6.17.3.1 General . 62
6.17.3.2 Codec Fall-back . 63
6.17.3.3 Transcoding . 63
6.17.3.4 Codec Simulcast . 63
6.17.3.5 Recommended Requirements for Codec Simulcast . 65
7 Conclusion . 65
Annex A: SDP examples for Multi-stream Multiparty Conference Media Handling . 66
A.1 General . 66
A.2 MSMTSI video offer/answer examples. 66
A.2.1 MSMTSI offer/answer towards an MTSI client . 66
A.2.2 MSMTSI answer from an MSMTSI MRF . 68
A.2.3 MSMTSI answer from an MSMTSI client in terminal. 71
A.2.4 MSMTSI Offer and Answer Using Codec Simulcast . 72
A.3 MSMTSI audio offer/answer examples. 74
A.3.1 MSMTSI offer with multi-stream audio support . 74
A.3.2 MSMTSI answer with multi-stream audio support . 76
A.3.3 MSMTSI CCCEx SDP offer/answer example . 77
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Annex B: QoS for Multi-stream Multiparty Conference Media Handling . 81
B.1 General . 81
B.2 QoS for MSMTSI video offer/answer examples . 81
Annex C: Technical Background . 83
C.1 General . 83
C.2 Simulcast Stream Identification . 83
C.2.1 General . 83
C.2.2 Codec Identification in SDP . 83
C.2.3 Sampling Identification in SDP . 83
C.2.4 Bandwidth Identification in SDP . 84
C.2.5 Simulcast Usage for WebRTC . 84
C.2.5.1 General . 84
C.2.5.2 RTP Payload Type Uniqueness . 84
C.2.5.3 RTP Payload Type Depletion . 84
C.2.6 Conclusion . 85
Annex D: Change history . 86
History . 87

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3GPP TR 26.980 version 17.0.0 Release 17 6 ETSI TR 126 980 V17.0.0 (2022-05)
Foreword
rd
This Technical Report has been produced by the 3 Generation Partnership Project (3GPP).
The contents of the present document are subject to continuing work within the TSG and may change following formal
TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an
identifying change of release date and an increase in version number as follows:
Version x.y.z
where:
x the first digit:
1 presented to TSG for information;
2 presented to TSG for approval;
3 or greater indicates TSG approved document under change control.
y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections,
updates, etc.
z the third digit is incremented when editorial only changes have been incorporated in the document.
Introduction
Media handling of Multimedia Telephony Service over IMS, MTSI, are based on 3GPP SA4 TS 26.114 [1]. MTSI
media handling is referred to by GSMA in GSMA PRD IR.92 [21] also known as VoLTE and GSMA PRD IR.94 [22]
also known as video over LTE. MTSI clients can connect to conferencing IMS communication services. 3GPP
conducted a study as part of a work item on Multi-stream Multiparty Conferencing Media-Handling for MTSI. The
work objective is to specify an increment to MTSI client media-handling specification TS 26.114 to enable a mass-
market multiparty communication service with excellent multiparty user experience and media quality. Such Operator
communication service evolution would match proprietary communication services in quality with excellent efficiency
and device reach.
The present document captures the study inputs, discussions and findings of the Multi-stream Multiparty Conference
Media Handling (MMCMH) work item. It describes a set of use cases with corresponding problem descriptions and
potential solutions. The conclusion gives recommendations as to what needs to be normatively specified to achieve the
MMCMH work item objectives.
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1 Scope
The Technical Report provides a study on the media handling aspects of Multi-stream Multiparty Conferencing for
MTSI. The study focuses on enabling
- support to receive multi-stream audio/video at the terminals in a multiparty conferencing,
- support for at least two video contents, e.g. one main and one presentation,
- talker ID provisioning,
- compatibility with MTSI TS 26.114 and GSMA IR.94 (Video over LTE) [22] and GSMA IR.92 (VoLTE)
[21], and
- applicability to both mobile and fixed access terminals
High-level use cases along with current limitations and recommendations are documented in the present document.
2 References
The following documents contain provisions which, through reference in this text, constitute provisions of the present
document.
- References are either specific (identified by date of publication, edition number, version number, etc.) or
non-specific.
- For a specific reference, subsequent revisions do not apply.
- For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including
a GSM document), a non-specific reference implicitly refers to the latest version of that document in the same
Release as the present document.
[1] 3GPP TS 26.114: "IP Multimedia Subsystem (IMS); Multimedia Telephony, Media handling and
interaction".
[2] Void.
[3] 3GPP TS 24.147: "Conferencing Using IP Multimedia Core Network; Stage 3".
[4] 3GPP TS 24.605: "CONF Using IP Multimedia Core Network".
[5] 3GPP TS 22.228: "Service Requirements for IP Multimedia Core Network; Stage 1".
[6] 3GPP TS 23.218:"IP Multimedia Session Handling; IP Multimedia Call Model; Stage 2".
[7] 3GPP TS 24.228: "Signalling Flows for IP Multimedia Call Control Based on SIP and SDP;
Stage 3".
[8] 3GPP TS 24.229: "IP Multimedia Call Control Protocol Based on SIP and SDP; Stage 3".
[9] 3GPP TR 22.948: "IP Multimedia Subsystem Convergent Multimedia Conferencing".
[10] 3GPP TR 29.847: "SIP Conferencing Models, Flows and Protocols".
[11] GSM Association: "WebRTC Codecs DRAFT v1.3", September 2014,
http://www.gsma.com/newsroom/wp-content/uploads/WebRTC-Whitepaper-v13.pdf.
[12] IETF Internet Draft: "Using Simulcast in SDP and RTP Sessions", June 2016,
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-simulcast/ (WORK IN PROGRESS).
[13] IETF RFC 4796: "The Session Description Protocol (SDP) Content Attribute", February 2007,
https://datatracker.ietf.org/doc/rfc4796/.
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3GPP TR 26.980 version 17.0.0 Release 17 8 ETSI TR 126 980 V17.0.0 (2022-05)
[14] GSM Association PRD IR.39: "IMS Profile for High Definition Video Conferencing (HDVC)
v5.0", July 2014, http://www.gsma.com/newsroom/wp-content/uploads//IR.39-v5.0.pdf.
[15] IETF RFC 4582: "The Binary Floor Control Protocol (BFCP)", November 2006,
https://datatracker.ietf.org/doc/rfc4582/.
NOTE A new version of this RFC is in the process of being published. See [17].
[16] IETF RFC 4583: "Session Description Protocol (SDP) Format for Binary Floor Control Protocol
(BFCP) Streams", November 2006, https://datatracker.ietf.org/doc/rfc4583/.
NOTE: A new version of this RFC is in the process of being published. See [18].
[17] IETF Internet Draft, draft-ietf-bfcpbis-rfc4582bis-16: "The Binary Floor Control Protocol (BFCP),
November 2015, WORK IN PROGRESS, https://datatracker.ietf.org/doc/draft-ietf-bfcpbis-
rfc4582bis/.
[18] IETF Internet Draft, draft-ietf-bfcpbis-rfc4583bis-15: "Session Description Protocol (SDP) Format
for Binary Floor Control Protocol (BFCP) Streams", July 2016, WORK IN PROGRESS,
https://datatracker.ietf.org/doc/draft-ietf-bfcpbis-rfc4583bis/.
[19] IETF RFC 3550: "RTP: A Transport Protocol for Real-Time Applications", July 2003,
https://datatracker.ietf.org/doc/rfc3550/.
[20] IETF RFC 5104: "Codec Control Messages in the RTP Audio-Visual Profile with Feedback
(AVPF)", February 2008, https://datatracker.ietf.org/doc/rfc5104/.
[21] GSM Association PRD IR.92, "IR.92 IMS Profile for Voice and SMS", v9.0, April 2015,
http://www.gsma.com/newsroom/wp-content/uploads//IR.92-v9.0.pdf.
[22] GSM Association PRD IR.94: "IR.94 IMS Profile for Conversational Video Service", v8.0.1,
November 2014, http://www.gsma.com/newsroom/wp-content/uploads//IR.94-v8.01.pdf.
[23] IETF RFC 7728, "RTP Stream Pause and Resume", March 2016.
[24] 3GPP TS 24.173: "IMS Multimedia Telephony Communication Service and Supplementary
Services".
[25] IETF RFC 4353: "A Framework for Conferencing with the Session Initiation Protocol (SIP)",
February 2006, https://datatracker.ietf.org/doc/rfc4353/.
[26] IETF RFC 3515: "The Session Initiation Protocol (SIP) Refer Method", April 2003,
https://datatracker.ietf.org/doc/rfc3515/.
[27] IETF RFC 5939: "Session Description Protocol (SDP) Capability Negotiation", Sept. 2010.
[28] IETF RFC 6184: "RTP Payload Format for H.264 Video", May 2011,
https://www.ietf.org/rfc/rfc6184.txt.
[29] IETF RFC 7798, "RTP Payload Format for High Efficiency Video Coding (HEVC)", March 2016.
[30] ITU-T Recommendation H.264: "Advanced video coding for generic audiovisual services".
[31] IETF RFC 3261: SIP: "Session Initiation Protocol", June 2002.
[32] 3GPP TR 21.905: "Vocabulary for 3GPP Specifications".
[33] Void.
[34] Void.
[35] Void.
[36] IETF RFC 4575: "A Session Initiation Protocol (SIP) Event Package for Conference State",
August 2006.
[37] Void.
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[38] Void.
[39] Void.
[40] ITU-T Recommendation H.265 (04/2015): "High efficiency video coding".
[41] 3GPP TS 26.171: "Speech codec speech processing functions; Adaptive Multi-Rate - Wideband
(AMR-WB) speech codec; General description".
[42] 3GPP TS 26.441: "Codec for Enhanced Voice Services (EVS); General Overview".
[43] IETF RFC 3264: "An Offer/Answer Model with the Session Description Protocol (SDP)", J.
Rosenberg and H. Schulzrinne, July 2002.
[44] IETF Internet Draft, draft-ietf-mmusic-rid-07: "RTP Payload Format Constraints", July 2016
(WORK IN PROGRESS), https://datatracker.ietf.org/doc/draft-ietf-mmusic-rid/.
[45] IETF RFC 7656: "A Taxonomy of Semantics and Mechanisms for Real-Time Transport Protocol
(RTP) Sources", J. Lennox, K. Gross, S. Nandakumar, G. Salguiero, and B. Burman, November
2015.
3 Definitions and abbreviations
3.1 Definitions
For the purposes of the present document, the terms and definitions given in TR 21.905 [32] and the following apply.
A term defined in the present document takes precedence over the definition of the same term, if any, in
TR 21.905 [32].
Mode-set: Used for the AMR and AMR-WB codecs to identify the codec modes that can be used in a session. A mode-
set can include one or more codec modes.
MSMTSI client: A multi-stream capable MTSI client supporting multiple streams. An MTSI client may support
multiple streams, even of the same media type, without being an MSMTSI client. Such an MTSI client may, for
example, add a second video to an ongoing video telephony session as shown in TS 26.114 Annex A.11.
MSMTSI MRF: An MSMTSI client implemented by functionality included in the MRFC and the MRFP.
MSMTSI client in terminal: An MSMTSI client that is implemented in a terminal or UE. The term "MSMTSI client
in terminal" is used in the present document when entities such as MRFP, MRFC or media gateways are excluded.
MTSI client: A function in a terminal or in a network entity (e.g. a MRFP) that supports MTSI.
MTSI client in terminal: An MTSI client that is implemented in a terminal or UE. The term "MTSI client in terminal"
is used in the present document when entities such as MRFP, MRFC or media gateways are excluded.
MTSI media gateway (or MTSI MGW): A media gateway that provides interworking between an MTSI client and a
non MTSI client, e.g. a CS UE. The term MTSI media gateway is used in a broad sense, as it is outside the scope of the
current specification to make the distinction whether certain functionality should be implemented in the MGW or in the
MGCF.
Operational mode: Used for the EVS codec to distinguish between EVS Primary mode and EVS AMR-WB IO mode.
Simulcast: Simultaneously sending different encoded representations (simulcast formats) of a single media source (e.g.
originating from a single microphone or camera) in different simulcast streams.
Simulcast format: The encoded format used by a single simulcast stream, typically represented by an SDP format and
all SDP attributes that apply to that particular SDP format, indicated in RTP by the RTP header payload type field.
Simulcast stream: The RTP stream carrying a single simulcast format in a simulcast.
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3.2 Abbreviations
For the purposes of the present document, the abbreviations given in TR 21.905 [32] and the following apply.
An abbreviation defined in the present document takes precedence over the definition of the same abbreviation, if any,
in TR 21.905 [32].
AVC Advanced Video Coding
BFCP Binary Floor Control Protocol
CCM Codec Control Messages
LTE Long Term Evolution
MSRP Message Session Relay Protocol
MSMTSI Multi-Stream Multimedia Telephony Service for IMS
MTSI Multimedia Telephony Service for IMS
SDP Session Description Protocol
4 Overview
Clause 5 provides a high-level description of the media handling in current 3GPP conferencing. The rest of the
document is organized as follows. clause 6 describes the use cases analysed in this study. Clause 7 provides the
conclusion and recommendations for further standardization efforts. Annex A includes some SDP offer/answer
examples.
5 Media Handling in Current 3GPP Conferencing
The current 3GPP specifications mentioning conferencing or group communication is mainly focusing on (SIP)
signalling aspects, and there is very little on media handling aspects. Those specifications include (list not intended to
be exhaustive) [3], [4], [5], [6], [7], [8], [9] and [10].

Figure 1. Existing Conference Architecture Example.
This briefly summarizes a few things regarding IMS conferences that are already specified, and that have an impact on
media handling:
- A centralized conference with the MRFP as conference focus is assumed [3], where media handling is not
explicitly described:
- MRFP is assumed to be an RTP "mixer" in IETF RFC 3550 [19] sense:
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- One possible implicit assumption is that the conference focus always transcodes (decodes, mixes, and re-
encodes) media individually towards every participant.
- Another possibility is to switch the video RTP stream untouched from one participant to another, and
possibly to all other participants.
- It is not described which video stream the MRFP should distribute to the different participants:
- One possible and reasonable assumption is that the video from some "active speaker" is distributed to other
participants, which would require some "active speaker" decision in the MRFP that in turn could be based on
speech activity analysis of the audio streams from every participant.
- If "active speaker" is distributed, it is common on the market to not distribute media from that "active
speaker" to itself, but rather the previous "active speaker" (as depicted in Figure 1).
- Another possible assumption is that all, or at least most, participant videos are re-sized, composed, and
transcoded into a checkerboard layout.
- A third possible assumption is that some type of floor control, e.g. based on Binary Floor Control Protocol
(BFCP) [3] and [15], is used, where the usage details in that case are so far left unspecified.
- When MRFP is not transcoding, when changing from forwarding one participant's video to another
participant's video, and since encoded video typically makes use of temporal redundancy, this change can
only be made at a point in the video stream that does not depend on any previous part of that video stream – a
so called "intra" picture. When deciding to make a change of forwarded video, the MRFP can trigger the UE
to send such intra picture by issuing an RTCP CCM FIR command to the UE, as described in RFC 5104, and
make the actual switch only when that intra picture arrives to the MRFP. Timing, reliability and bandwidth
aspects of FIR transmission are described in RFC 5104. A MTSI UE is already required to support and react
on FIR. When changing to a new source and if the new source is inactive (on hold or not established) then
that stream has to be activated before the MRFP can switch to it.
- SIP conference call control includes three allowed options [3] and [8]:
- Each participating UE calls in to conference (SIP INVITE).
- The originating UE calls in to conference and requests it to call out to other participants (SIP INVITE with
recipient list).
- A UE has an ongoing point-to-point or three-party call that is moved into a conference (SIP REFER).
- MRFC always includes "isFocus" tag in its SIP signalling [3] (regardless if it is a SIP request or response),
which lets the UE know that it is signalling with a conference and not another UE.
- The conference may optionally make use of explicit floor control through Binary Floor Control Protocol (BFCP)
[3] and [15]:
- The use of a floor control protocol allows explicitly, and even manually, controlling which participant's video
is distributed to others by the MRFP.
- The use of this "application" media stream is negotiated through SDP [16].
- TCP transport of BFCP is assumed, possibly because this was until fairly recently the only specified transport
in IETF, but many BFCP implementations on the market instead use UDP in a straightforward way, and there
is well progressed work in IETF to describe this in an update to the BFCP RFC [17].
The MMCMH work item objectives include enabling multi-stream audio/video support at the terminals. In addition, as
specified in the MMCMH WI objectives, the conference focus and the terminals may receive stereo streams for further
processing and rendering. The clauses below present the multi-stream audio and video use cases, where the terminals
receive and decode the multiple streams of audio/video and thumbnails, and render them at the terminal that is
potentially transcoder-free. Conferencing using IP multimedia core network and with conference focus mixing (e.g.
with MRFP) are addressed in 3GPP TS 24.147 and IETF RFC 4353.
ETSI
3GPP TR 26.980 version 17.0.0 Release 17 12 ETSI TR 126 980 V17.0.0 (2022-05)
6 Use cases
6.1 Overview
This clause contains multimedia group communication use cases that enables multi-stream video and audio support at
the terminals.
6.2 Use case A: Transcoding Free Continuous Presence
When calling in to a group video call, the user is able to see video from more than a single one of the other participants
in the call, which is commonly referred to as "continuous presence". This is typically desirable in a group
communication for a user to be able to see the reactions of more than a single participant.
In contrast, when receiving video from one participant at a time, several different approaches to choose that single
participant are possible, subject to implementation in the conference focus. It may be that the active speaker is chosen,
based on conference focus analysis of some (unspecified) voice activity measure of all participants. It may be based on
a chair person's explicit and manual control of the conference focus, typically requiring a floor control protocol, such as
for example BFCP, [15]. It can also be based on other approaches, such as for example an automatic, timed round-robin
among all participants.
The participant layout or the number of participants simultaneously visible in a continuous presence layout is neither
specified nor specifically restricted in this use case. An implementation will however
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