Network Working Group E. Ivov
Internet-Draft SIP Communicator
Intended status: Informational E. Marocco
Expires: April 29, 2010 Telecom Italia
October 26, 2009
A Real-Time Transport Protocol (RTP) Extension Header for Mixer-to-
client Audio Level Indication
draft-ivov-avt-slic-02
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Abstract
This document describes a mechanism for RTP-level mixers in audio
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conferences to deliver information about the audio level of the
individual participants. Such audio level indicators are transported
in the same RTP packets as the audio data they pertain to.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Protocol Operation . . . . . . . . . . . . . . . . . . . . . . 4
4. Header Format . . . . . . . . . . . . . . . . . . . . . . . . 6
5. Audio level encoding . . . . . . . . . . . . . . . . . . . . . 6
6. Signaling Information . . . . . . . . . . . . . . . . . . . . 7
7. Security Considerations . . . . . . . . . . . . . . . . . . . 9
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
9. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 10
10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 10
11. Appendix: Design choices . . . . . . . . . . . . . . . . . . . 10
11.1. SIP event package for conference state . . . . . . . . . 10
11.2. The RTP Control Protocol (RTCP) . . . . . . . . . . . . . 11
11.3. Encoding levels in the payload . . . . . . . . . . . . . 11
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 12
12.1. Normative References . . . . . . . . . . . . . . . . . . 12
12.2. Informative References . . . . . . . . . . . . . . . . . 12
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 13
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1. Introduction
The Framework for Conferencing with the Session Initiation Protocol
(SIP) defined in RFC 4353 [RFC4353] presents an overall architecture
for multi-party conferencing. Among others, the framework borrows
from RTP [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 [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. Audio 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 audio level indicators
to the corresponding participant related components in the user
interface as displayed in Figure 1.
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________________________
| |
| 00:42 | Weekly Call |
|________________________|
| |
| |
| Alice |====== | (S) |
| |
| Bob |= | |
| |
| Carol | | (M) |
| |
| Dave |=== | |
| |
|________________________|
Figure 1: Displaying detailed speaker information to the user by
including audio 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
audio levels.
This document specifies an RTP extension header that allows such
mixers to deliver audio 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 [RFC2119].
3. Protocol Operation
According to RFC 3550 [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
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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 [RFC3261] for example,
the Event Package for Conference State [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 audio-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" [RFC5285].
Each instance of this header contains a list of one-octet audio
levels expressed in -dBov, with values from 0 to 127 representing 0
to -127 dBov(see Section 4 and Section 5).
Every audio 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 audio 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 audio 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 audio 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 audio-level information they receive from peer mixers (as
long as the total CSRC count remains below 16). Given that the
maximum audio level is not precisely defined by this specification,
it is likely that in such situations average audio levels would be
perceptibly different for the participants located behind the
different mixers.
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4. Header Format
The audio level indicators are delivered to the receivers in-band
using the "General Mechanism for RTP Header Extensions" [RFC5285].
The payload of this extension is an ordered sequence of 8-bit audio
level indicators encoded as per Section 5.
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 |0| level 1 |0| level 2 |0| level 3 ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: Audio level indicators extension format
The 4-bit len field is the number minus one of data bytes (i.e. audio
level values) transported in this header extension element following
the one-byte header. Therefore, the value zero in this field
indicates that one byte of data follows. A value of 15 is not
allowed by this specification and it MUST NOT be used as the RTP
header can carry a maximum of 15 CSRC IDs. The maximum value allowed
is therefore 14 indicating a following sequence of 15 audio level
values.
Note that use of the two-byte header defined in RFC 5285 [RFC5285]
follows the same rules the only change being the length of the ID and
len fields.
5. Audio level encoding
Audio level indicators are encoded in the same manner as audio noise
level in the RTP Payload Comfort Noise specification [RFC3389] and
audio level in the RTP Extension Header for Client-to-mixer Audio
Level Notification [I-D.lennox-avt-rtp-audio-level-exthdr]
specification. The magnitude of the audio level is packed into the
least significant bits of one audio-level byte with the most
significant bit unused and always set to 0 as shown below in
Figure 3.
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0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|0| level |
+-+-+-+-+-+-+-+-+
Figure 3: Audio Level Encoding
The audio level is expressed in -dBov, with values from 0 to 127
representing 0 to -127 dBov. dBov is the level, in decibels, relative
to the overload point of the system, i.e. the maximum-amplitude
signal that can be handled by the system without clipping. (Note:
Representation relative to the overload point of a system is
particularly useful for digital implementations, since one does not
need to know the relative calibration of the analog circuitry.) For
example, in the case of u-law (audio/pcmu) audio [ITU.G.711], the 0
dBov reference would be a square wave with values +/- 8031. (This
translates to 6.18 dBm0, relative to u-law's dBm0 definition in Table
6 of G.711.)
6. Signaling Information
The URI for declaring the audio level header extension in an SDP
extmap attribute and mapping it to a local extension header
identifier is "urn:ietf:params:rtp-hdrext:csrc-audio-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-audio-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 audio 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 audio level information RTP extension header.
Conferencing clients that support audio 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
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"sendrecv" when answering other "sendrecv" offers.
The following Figure 4 and Figure 5 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-audio-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-audio-level
A client-initiated example SDP offer/answer exchange negotiating an
audio stream with one-way flow of of audio level information.
Figure 4
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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-audio-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-audio-level
An example SDP offer/answer exchange between two conference focus
entities with mixing capabilities negotiating an audio stream with
bidirectional flwo of audio level information.
Figure 5
7. Security Considerations
1. This document defines a means of attributing audio 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 audio
activity from one participant appear as coming from another.
2. Furthermore, the fact that audio 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 and they would probably also apply when using
CSRC identifiers the way they were specified in RFC 3550
[RFC3550]. 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.
8. 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
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to the following data:
Extension URI: urn:ietf:params:rtp-hdrext:csrc-audio-level
Description: Mixer-to-client audio level indicators
Contact: emcho@sip-communicator.org
Reference: RFC XXXX
9. 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. Audio levels in video streams. This specification allows use of
audio 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 audio level transport in video streams.
10. Acknowledgments
Roni Even, Ingemar Johansson, Michael Ramalho and several others
provided helpful feedback over the dispatch mailing list.
SIP Communicator's participation in this specification is funded by
the NLnet Foundation.
11. Appendix: Design choices
During discussions on the subject of audio levels the decision to
transport audio levels in RTP packets, rather than another protocol
was questioned several times which is why the authors find it worth
explaining here. The following subsections describe alternative
mechanisms for delivering audio levels and the reasons why authors
decided not to use them.
11.1. SIP event package for conference state
RFC 4575 [RFC4575] defines a conference event package for tightly
coupled conferences using the Session Initiation Protocol (SIP)
events framework. It allows for the delivery of various conference
related details such as conference descriptions, participant count
and identity. The document also provides a way of indicating who the
speakers are at any given moment by specifying a mechanism for
mapping conference participants to RTP SSRC/CSRC identifiers. All
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these details are dispatched in an asynchronous manner using the SIP
events framework, or, in other words, through NOTIFY SIP requests
following an initial SUBSCRIBE from a participant.
Contrary to "plain" active speaker infomation, where significant
changes only occur once every several seconds, audio level in human
speech is obviously a very time sensitive characteristic which would
require frequent updates (i.e. approximately once every 50-100 ms).
In order for the update of the user interface to appear "natural" to
the user, audio level information would probably have to be delivered
for every one or two RTP packets. Using RFC 4575 [RFC4575] or SIP in
general for this would generate traffic on the (often low-bandwidth)
signalling path comparable to, if not exceeding, the media itself.
It may also prove relatively hard for client developers to
synchronize the information they receive from SIP messages with the
one they obtain from the media flows.
It is probably also worth mentioning that the use of RFC 4575
[RFC4575] for such a feature would make the mechanism incompatible
with non-SIP signaling protocols like, for example, XMPP [RFC3920]
and its Jingle extensions.
11.2. The RTP Control Protocol (RTCP)
Similar to using SIP, delivering audio levels through RTCP would
cause bandidth and synchronization issues. Furthermore the RTP
specification [RFC3550] explicitly recommends that the fraction of
the session bandwidth added for RTCP be fixed at 5% which could not
be sufficient for the transport of audio level indicators.
11.3. Encoding levels in the payload
Given the content specific nature of audio levels, it has been
suggested that audio level information be encoded and transmitted as
part of the payload. While this is indeed a feasible approach,
implementing it would require a substantial effort. In order to
implement support for such a feature, client developers would need to
explicitly handle it in all individual codec modules of their
application. Compared to RTP extensions, the mechanism would
therefore represent a substantial additional effort without offering
any meaningful advantages.
12. References
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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.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
12.2. Informative References
[I-D.lennox-avt-rtp-audio-level-exthdr]
Lennox, J., "A Real-Time Transport Protocol (RTP) Header
Extension for Client-to- Mixer Audio Level Indication",
draft-lennox-avt-rtp-audio-level-exthdr-01 (work in
progress), October 2009.
[ITU.G.711]
International Telecommunications Union, "Pulse Code
Modulation (PCM) of Voice Frequencies", ITU-
T Recommendation G.711, November 1988.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, September 2002.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3920] Saint-Andre, P., Ed., "Extensible Messaging and Presence
Protocol (XMPP): Core", RFC 3920, October 2004.
[RFC4353] Rosenberg, J., "A Framework for Conferencing with the
Session Initiation Protocol (SIP)", RFC 4353,
February 2006.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
Initiation Protocol (SIP) Event Package for Conference
State", RFC 4575, August 2006.
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Authors' Addresses
Emil Ivov
SIP Communicator
Strasbourg 67000
France
Email: emcho@sip-communicator.org
Enrico Marocco
Telecom Italia
Via G. Reiss Romoli, 274
Turin 10148
Italy
Email: enrico.marocco@telecomitalia.it
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