Delivering Conference Participant Sound Level Indicators in RTP Streams
draft-ivov-avt-slic-00

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Abstract

This document describes a mechanism for RTP-level mixers in audio conferences to deliver information about the sound level information on the individual participants. Such sound level indicators are transported in the same RTP packets as the audio data they pertain to.

Table of Contents

1.  Introduction
2.  Terminology
3.  Protocol Operation
4.  Header Format
5.  Signaling Information
6.  Security Considerations
7.  IANA Considerations
8.  Open Issues
9.  Acknowledgments
10.  Appendix: An alternative approach
11.  References
    11.1.  Normative References
    11.2.  Informative References
§  Authors' Addresses

1.  Introduction

The Framework for Conferencing with the Session Initiation Protocol (SIP) defined in RFC 4353 (Rosenberg, J., “A Framework for Conferencing with the Session Initiation Protocol (SIP),” February 2006.) [RFC4353] presents an overall architecture for multi-party conferencing. Among others, the framework borrows from RTP (Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications,” July 2003.) [RFC3550] and extends the concept of a mixer entity "responsible for combining the media streams that make up a conference, and generating one or more output streams that are delivered to recipients". Every participant would hence receive, in a flat single stream, media originating from all the others.

Using such centralized mixer-based architectures simplifies support for conference calls on the client side since they would hardly differ from one-to-one conversations. However, the method also introduces a few limitations. The flat nature of the streams that a mixer would output and send to participants makes it difficult for users to identify the original source of what they are hearing.

Mechanisms that allow the mixer to send to participants cues on current speakers (e.g. the CSRC fields in RTP (Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications,” July 2003.) [RFC3550]) only work for speaking/silent binary indications. There are, however, a number of use cases where one would require more detailed information. Possible examples include the presence of background chat/noise/music/typing, someone breathing noisily in their microphone, or other cases where identifying the source of the disturbance would make it easy to remove it (e.g. by sending a private IM to the concerned party asking them to mute their microphone). A more advanced scenario could involve an intense discussion between multiple participants that the user does not personally know. Sound level information would help better recognize the speakers by associating with them complex (but still human readable) characteristics like loudness and speed for example.

One way of presenting such information in a user friendly manner would be for a conferencing client to attach sound level indicators to the corresponding participant related components in the user interface as displayed in Figure 1.

                      ------------------------
                     |                        |
                     |  00:42 |  Weekly Call  |
                     |                        |
                     |------------------------|
                     |                        |
                     | Alice |======    | (S) |
                     |                        |
                     | Bob   |=         |     |
                     |                        |
                     | Carol |          | (M) |
                     |                        |
                     | Dave  |===       |     |
                     |                        |
                     |________________________|

Displaying detailed speaker information to the user by including sound level for every participant.

Implementing a user interface like the above requires analysis of the media sent from other participants. In a conventional audio conference this is only possible for the mixer since all other conference participants are generally receiving a single, flat audio stream and have therefore no immediate way of determining individual sound levels.

This document specifies an RTP extension header that allows such mixers to deliver sound level information to conference participants by including it directly in the RTP packets transporting the corresponding audio data.

2.  Terminology

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 (Bradner, S., “Key words for use in RFCs to Indicate Requirement Levels,” March 1997.) [RFC2119].

3.  Protocol Operation

According to RFC 3550 (Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications,” July 2003.) [RFC3550] a mixer is expected to include in outgoing RTP packets a list of identifiers (CSRC IDs) indicating the sources that contributed to the resulting stream. The presence of such CSRC IDs allows an RTP client to determine, in a binary way, the active speaker(s) in any given moment. RTCP also provides a basic mechanism to map the CSRC IDs to user identities through the CNAME field. More advanced mechanisms, may exist depending on the signaling protocol used to establish and control a conference. In the case of the Session Initiation Protocol (Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, “SIP: Session Initiation Protocol,” June 2002.) [RFC3261] for example, the Event Package for Conference State (Rosenberg, J., Schulzrinne, H., and O. Levin, “A Session Initiation Protocol (SIP) Event Package for Conference State,” August 2006.) [RFC4575] defines a <src-id> tag which binds CSRC IDs to media streams and SIP URIs.

This document describes an RTP header extension that allows mixers to indicate the sound-level of every conference participant (CSRC) in addition to simply indicating their on/off status. This new header extension is based on the "General Mechanism for RTP Header Extensions" (Singer, D. and H. Desineni, “A General Mechanism for RTP Header Extensions,” July 2008.) [RFC5285].

Each instance of this header contains a list of one-octet sound level values (see Section 4 (Header Format)). Such values indicate sound level on a 0 to 255 scale where 0 is silence (i.e. same as omitting the corresponding source id from the CSRC list) and 255 corresponds to a threshold accepted by the mixer implementation as the maximum sound level that a participant is likely to reach during a conference.

Every sound level value pertains to the CSRC identifier located at the corresponding position in the CSRC list. In other words, the first value would indicate the sound level of the conference participant represented by the first CSRC identifier in that packet and so forth. The number and order of these values MUST therefore match the number and order of the CSRC IDs present in the same packet.

When encoding sound level information, a mixer SHOULD include in a packet information that corresponds to the audio data being transported in that same packet. It is important that these values follow the actual stream as closely as possible. Therefore a mixer SHOULD also calculate the values after the original contributing stream has undergone possible processing such as level normalization, and noise reduction for example.

Note that in some cases a mixer may be sending an RTP audio stream that only contains sound level information and no actual audio. Updating a (web) interface conference module may be one reason for this to happen.

It may sometimes happen that a conference involves more than a single mixer. In such cases each of the mixers MAY choose to relay the CSRC list and sound-level information they receive from peer mixers (as long as the total CSRC count remains below 16). Given that the maximum sound level is not precisely defined by this specification, it is likely that in such situations average sound levels would be perceptibly different for the participants located behind the different mixers.

4.  Header Format

The sound level indicators are delivered to the receivers in-band using the "General Mechanism for RTP Header Extensions" (Singer, D. and H. Desineni, “A General Mechanism for RTP Header Extensions,” July 2008.) [RFC5285]. The payload of this extension (the transmitted list of sound level values) is a sequence of 8-bit unsigned integers.

The form of the sound level indicators extension block is as follows:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |  ID   |  len  |    level 1    |    level 2    |    level 3   ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Note that use of the two-byte header defined in RFC 5285 (Singer, D. and H. Desineni, “A General Mechanism for RTP Header Extensions,” July 2008.) [RFC5285] follows the same rules the only change being the length of the ID and len fields.

5.  Signaling Information

The URI for declaring the sound level header extension in an SDP extmap attribute and mapping it to a local extension header identifier is "urn:ietf:params:rtp-hdrext:csrc-sound-level". There is no additional setup information needed for this extension (i.e. no extensionattributes).

An example attribute line in the SDP, for a conference might be:

        a=extmap:7 urn:ietf:params:rtp-hdrext:csrc-sound-level

The above mapping will most often be provided per media stream (in the media-level section(s) of SDP, i.e., after an "m=" line) or globally if there is more than one stream containing sound level indicators in a session.

Presence of the above attribute in the SDP description of a media stream indicates that some or all RTP packets in that stream would contain the sound level information RTP extension header.

Conferencing clients that support sound level indicators and have no mixing capabilities SHOULD always include the direction parameter in the "extmap" attribute setting it to "recvonly". Conference focus entities with mixing capabilities MAY omit the direction or set it to "sendrecv" in SDP offers. Such entities SHOULD set it to "sendonly" in SDP answers to offers with a "recvonly" parameter and to "sendrecv" when answering other "sendrecv" offers.

The following Figure 2 and Figure 3 show two example offer/answer exchanges between a conferencing client and a focus, and between two conference focus entities.

  v=0
  o=alice 2890844526 2890844526 IN IP6 host.example.com
  c=IN IP6 host.example.com
  t=0 0
  m=audio 49170 RTP/AVP 0 4
  a=rtpmap:0 PCMU/8000
  a=rtpmap:4 G723/8000
  a=extmap:1/recvonly urn:ietf:params:rtp-hdrext:csrc-sound-level

  v=0
  i=A Seminar on the session description protocol
  o=conf-focus 2890844730 2890844730 IN IP6 focus.example.net
  c=IN IP6 focus.example.net
  t=0 0
  m=audio 52543 RTP/AVP 0
  a=rtpmap:0 PCMU/8000
  a=extmap:1/sendonly urn:ietf:params:rtp-hdrext:csrc-sound-level

A client-initiated example SDP offer/answer exchange negotiating an audio stream with one-way flow of of sound level information.

  v=0
  i=Un seminaire sur le protocole de description des sessions
  o=fr-focus 2890844730 2890844730 IN IP6 focus.fr.example.net
  c=IN IP6 focus.fr.example.net
  t=0 0
  m=audio 49170 RTP/AVP 0
  a=rtpmap:0 PCMU/8000
  a=extmap:1/sendrecv urn:ietf:params:rtp-hdrext:csrc-sound-level

  v=0
  i=A Seminar on the session description protocol
  o=us-focus 2890844526 2890844526 IN IP6 focus.us.example.net
  c=IN IP6 focus.us.example.net
  t=0 0
  m=audio 52543 RTP/AVP 0
  a=rtpmap:0 PCMU/8000
  a=extmap:1/sendrecv urn:ietf:params:rtp-hdrext:csrc-sound-level

An example SDP offer/answer exchange between two conference focus entities with mixing capabilities negotiating an audio stream with bidirectional flwo of sound level information.

6.  Security Considerations

1. This document defines a means of attributing sound level to a particular participant in a conference. An attacker may try to modify the content of RTP packets in a way that would make sound activity from one participant appear as coming from another.
2. Furthermore, the fact that sound level values would not be protected even in an SRTP session may be of concern in some cases where the activity of a particular participant in a conference is confidential.
3. Both of the above are concerns that stem from the design of the RTP protocol itself. It is therefore important that according to the needs of a particular scenario, implementors and deployers consider use of a lower level security and authentication mechanism.

7.  IANA Considerations

This document defines a new extension URI that, if approved, would need to be added to the RTP Compact Header Extensions sub-registry of the Real-Time Transport Protocol (RTP) Parameters registry, according to the following data:

        Extension URI: urn:ietf:params:rtp-hdrext:csrc-sound-level
        Description:   Sound level indicators
        Contact:       emcho@sip-communicator.org
        Reference:     RFC XXXX

8.  Open Issues

At the time of writing of this document the authors have no clear view on how and if the following list of issues should be address here:

1. Specific sound level mappings. The current version of this specification treats sound level indicators as referable to any scale chosen by the mixer. The only limitations consist in making sure that the value of 0 should correspond to participant inactivity/silence and the value 255x to a level that would appear to users as loud but still attainable. It is however possible to map specific levels (e.g. measured in dBm) with the purpose of achieving cross-mixer uniformity of these values. An obvious tradeoff here is the increased complexity of implementation that would require mixers to convert sound level to whatever specific unit they use for internal estimation, which could be non-trivial in a number of cases.
2. Sound levels in video streams. This specification allows use of sound level values in "silent" audio streams that don't otherwise carry any payload thus allowing their delivery within systems where the various focus/mixer components communicate with each other as conference participants. The same train of thought may very well justify sound level transport in video streams.

9.  Acknowledgments

Roni Even, Ingemar Johansson, and several others provided helpful feedback over the dispatch mailing list.

SIP Communicator's participation in this specification is funded by the NLnet Foundation.

10.  Appendix: An alternative approach

The problem statement (Ivov, E. and E. Marocco, “Dispatching Sound Level Indicators in Conferences (Problem Statement),” May 2009.) [I‑D.ivov‑dispatch‑slic‑ps] preceding this document originally favored a slightly different resolution approach that the authors feel may still be relevant and therefore worth publishing here.

A very simple way for a mixer to use the CSRC fields as a transport means for sound level indication would be to extend their meaning over a series of packets rather than a single one. This way it could be specified that the sound-level of a particular participant, represented on a zero to ten scale, corresponds to the number of occurrences of its CSRC identifier in the ten most recent RTP packets received from the mixer.

For example, consider a conference call with four participants: Alice, Bob, Carol, and Dave. At a certain point in time Alice has a sound level of 6/10, Bob 1/10, Carol is silent or in other words 0/10 and Dave has a level of 3/10. In order to describe this state the mixer could have sent the last ten RTP packets with the following CSRC configuration:

A possible representation of a particular sound level configuration through the presence/absence of CSRC IDs in subsequent RTP packets.

The graphical interface of a user agent involved in such a conference (like the one sketched in Figure 1) would then display correct sound levels just showing for each participant as many ticks as were the occurrencies of the respective CSRC in the previous ten RTP packets.

The algorithm for encoding sound level information this way is relatively simple. In order to determine whether or not to include a particular CSRC a mixer should:

• include the CSRC if the sound level of the participant in the current packet is greater than the number of occurrencies of that same CSRC in the nine previous packets;
• omit the CSRC if the sound level of the participant in the current packet is lower than or equal to the number of occurrencies of that same CSRC in the nine previous packets.

There are several advantages to using this approach, the most obvious being its simplicity as well as the fact that sound level information is transported together with the parts of the audio stream that it actually concerns which should make synchronization straightforward.

The technique would also work with other signaling protocols using RTP such as XMPP's (Saint-Andre, P., Ed., “Extensible Messaging and Presence Protocol (XMPP): Core,” October 2004.) [RFC3920] Jingle extensions for example.

One of the first disadvantages that come to mind with this approach is the fact that mixer would not be able to indicate level in a single packet but would have to distribute it over a succession of up to ten packets which would reduce the reactivity of the representation.

It is probably worth mentioning, however, that a granularity that allows switching from a level of zero to ten and back to zero again in an instant manner is not of much use anyway since such UI updates would be barely perceptible to the user. Still, this is a UI decision and making it on a protocol level may bring some inconveniences.

Another possible problem would come from implementations using CSRC presence in a binary way to determine current speaker. When running against a mixer that supports sound level indication such implementations may appear to be jumpy as the participants that they are designating as active may be changing status too rapidly.

11.  References

11.1. Normative References

11.2. Informative References

Authors' Addresses