Internet Engineering Task Force G. Hellstrom
Internet-Draft Omnitor
Intended status: BCP A. van Wijk
Expires: September 15, 2011 Real-Time Text Taskforce (R3TF)
March 14, 2011
Text media handling in RTP based real-time conferences
draft-hellstrom-text-conference-04
Abstract
This memo specifies methods for text media handling in multi-party
calls, where the text is carried by the RTP protocol. Real-time text
is carried in a time-sampled mode according to RFC 4103. Centralized
multi-party handling of real-time text is achieved through a media
control unit coordinating multiple RTP text streams into one single
stream RTP session, identifying each stream with its own CSRC.
Identification for the streams are provided through the RTCP
messages. This mechanism enables the receiving application to
present the received real-time text medium in different ways
according to user preferences. Some presentation related features
are also described explaining suitable variations of transmission and
presentation of text. Call control features are described for the
SIP environment, while the transport mechanisms should be suitable
for any IP based call control environment using RTP transport. Two
alternative methods using a single RTP stream and source
identification inline in the text stream are also described.
Status of this Memo
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provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on September 15, 2011.
Copyright Notice
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Requirements Language . . . . . . . . . . . . . . . . . . . 3
2. Centralized conference model . . . . . . . . . . . . . . . . . 3
2.1. Coordination of text RTP streams . . . . . . . . . . . . . 4
2.2. Session control of multi-party sessions . . . . . . . . . . 4
3. Identification of the source of text . . . . . . . . . . . . . 5
4. Presentation of multi-party text . . . . . . . . . . . . . . . 5
4.1. Associating identities with text streams . . . . . . . . . 6
5. Transmission of text from each user . . . . . . . . . . . . . . 6
6. Presentation level source indicator . . . . . . . . . . . . . . 6
7. Mixing for conference-unaware user agents . . . . . . . . . . . 7
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 7
9. Security Considerations . . . . . . . . . . . . . . . . . . . . 8
10. Congestion considerations . . . . . . . . . . . . . . . . . . . 8
11. Normative References . . . . . . . . . . . . . . . . . . . . . 8
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 9
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1. Introduction
Real-time text is a medium in real-time conversational sessions.
Text entered by participants in a session is transmitted in a time-
sampled fashion, so that no specific user action is needed to cause
transmission. This gives a direct flow of text that is suitable in a
real-time conversational setting. The real-time text medium can be
combined with other media in multimedia sessions.
A number of multimedia sessions can be combined in a multi-party
session. This memo specifies how the real-time text streams are
handled in such multi-party sessions.
The description is mainly focused on the transport level, but also
describes a few presentation level features.
Transport of real-time text is specified in RFC 4103 [RFC4103] RTP
Payload for text conversation. It makes use of RFC 3550 [RFC3550]
Real Time Protocol, for transport, and is usually used in the SIP
Session Initiation Protocol RFC 3261 [RFC3261] environment, even if
it is also used in other call control environments. Call control
aspects in this specification are explained with examples from SIP.
The specifications about how to handle multi-party text transport,
identification and presentation are valid also for other call control
environments where RTP and RTCP are used.
A very brief overview of functions for both real-time and messaging
text handling in multi-party sessions is described in RFC 4597
[RFC4579] Conferencing Scenarios. This specification builds on that
description and indicates what existing protocol mechanisms should be
used to implement multi-party handling of text in real-time sessions.
1.1. 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].
2. Centralized conference model
In the centralized conference model, one function co-ordinates the
sessions with participants in the multi-party session. This function
also controls media mixer functions for the media appearing in the
session. The central function is common for control of all media,
while the media mixers may work differently for each medium.
The central function is called the Focus UA and may be co-located in
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an advanced terminal including multi-party control functions, or it
may be located in a separate location. Many variants exist for
setting up sessions including the multipoint control centre, It is
not within scope of this description to describe these, but rather
the media specific handling in the mixer required to handle multi-
party calls.
The main principle for handling real-time text media in a centralized
conference is that one RTP session for real-time text is established
between the multipoint media control centre and each participant who
is going to have real-time text exchange with the others.
2.1. Coordination of text RTP streams
The preferred way of coordinating text RTP streams is that within
each RTP session, text from all participants are transmitted from the
media mixer in the same RTP stream, thus all using the same
destination address/port combination, and the same RTP SSRC as
described in Section 7.1 and 7.3 of RTP RFC 3550 [RFC3550] about the
Mixer function. The source of the primary text in each RTP packet is
identified by the CSRC parameter, containing the SSRC of the initial
source of text.
The mixer MUST NOT transmit redundant levels of text from one source
together with primary text from another source. Thus, when there is
text available for primary or redundant transmission from more than
one source, the mixer MUST buffer text from other sources until all
the redundant transmissions of a packet from one selected source has
been transmitted. Without this restriction, there would be no way to
decide with what source to associate text recovered from the
redundant information in case of packet loss.
The identification of the source is made through the RTCP SDES CNAME
and NAME packets as described in RTP[RFC3550].
This method enables the receiver to freely select display
characteristics of the text conversation.
2.2. Session control of multi-party sessions
General session control aspects for multi-party sessions are
described in RFC 4575 [RFC4575] A Session Initiation Protocol (SIP)
Event Package for Conference State, and RFC 4579 [RFC4579] Session
Initiation Protocol (SIP) Call Control - Conferencing for User
Agents. The nomenclature of these specifications are used here.
The procedures for the mixer-based model shall only be applied if a
capability exchange for mixer-based real-time text transmission has
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been completed. Capability for the mixer-based model is indicated by
both the focus and the user agent by the media tag rtt-mixer=rtp-mix
3. Identification of the source of text
The Focus UA co-ordinates the media flow. Real-time text media from
different sources are combined in one text media session by the Focus
UA. The main principle is that the Focus UA SHOULD act as an RTP
Mixer as described in RTP Section 7.1 [RFC3550].
The RTP text stream from each participant who transmits text is
allocated one unique CSRC. The CSRC is used by the receiver to
identify text packets originating from one source. Each RTP packet
MUST contain text from only one source.
The redundancy mechanism for increased robustness used by the RFC
4103 transport makes use of the RTP sequence number for detection of
loss. The RTP Mixer mechanism maintains a separate CSRC for each
source RTP stream in the combined RTP session. Therefore the RTP
Mixer mechanism can be used for conveying text from multiple sources
to one destination, with maintained possibility to detect and recover
loss and identify text from the different sources.
As soon as a new member is added to the RTP session, its
characteristics shall be transmitted in RTCP SDES CNAME and NAME
reports according to section 6.5 in RFC 3550.
The RTCP SDES report, SHOULD contain identification of the source
represented by the CSRC identifier. This identification MUST contain
the CNAME field and MAY contain the NAME field and other defined
fields of the SDES report.
A focus UA SHOULD primarily convey SDES information received from the
sources of the session members. When such information is not
available, the focus UA SHOULD compose CSRC, CNAME and NAME
information from available information from the SIP session with the
participant.
4. Presentation of multi-party text
All session participants MUST observe the CSRC field of incoming text
RTP packets, and make note of what source they came from in order to
be able to present text in a way that makes it easy to read text from
each participant in a session, and get information about the source
of the text.
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4.1. Associating identities with text streams
A source identity SHOULD be composed from available information
sources and displayed together with the text as indicated in ITU-T
T.140 Appendix[T.140].
The source should primarily be the NAME field from incoming SDES
packets. If this information is not available, and the session is a
two-party session, then the T.140 source identity SHOULD be composed
from the SIP session participant information. For multi-party
sessions the source identity may be composed by local information if
sufficient information is not available in the session.
Applications may abbreviate the presented source identity to a
suitable form for the available display.
5. Transmission of text from each user
UAs participating in sessions with real-time text, SHOULD send SDES
packets in RTCP giving values to appropriate identification fields.
The CNAME field SHALL be included in SDES packets.
The NAME field should be given a value that is suitable as an
identifier of text from the user of the UA.
6. Presentation level source indicator
In certain application environments, it may be known to be unsuitable
to use the CSRC identification on the RTP level as the base for
identificating the source of text. In such cases, an inline coding
of the source of text SHOULD be applied in the data stream itself,
and an RTP mixer function normally without CSRC identification used
for coordinating the sources of text into one RTP stream.
The support of this mixer type is indicated by the SIP header rtt-
mix=t140, both by the focus and the user agent.
Information uniquely identifying each user in the multi-party session
SHALL then be placed as the parameter value "cn" in the T.140
application protocol function with the function code "c". The
identifier shall thus be formatted like this: SOS c cn field contents
ST, where SOS and ST are coded as specified in ITU-T T.140 [T.140].
The cn parameter shall be kept short so that it can be repeated in
the transmission without concerns for network load.
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The information otherwise conveyed in the NAME field of an SDES
packet SHOULD then be placed as the parameter value in the T.140
application protocol function with the function code "n".
A T.140 application protocol function with the function code "c" MUST
be included in the text in the beginning of text when the source of
the text changes. A T.140 application protocol function with the
function code "c" MAY be repeated in the text from the same the
source. A T.140 application protocol function with the function code
"n" MAY be included in the text to further provide identification of
the transmitting party. This information SHOULD also be provided in
the SDES name field. A receiving UA SHOULD separate text from the
different sources and identify and display them accordingly.
In this case, the mixer can use the redundancy transmission function
of RFC 4103 without restrictions.
7. Mixing for conference-unaware user agents
Multi-party real-time text contents can be transmitted to conference-
unaware user agents if source labeling and formatting of the text is
performed by a mixer. This method has the limitations that the
format of source identification is purely controlled by the mixer,
and that only one source at a time is allowed to present in real-
time. Other sources need to be stored temporarily waiting for an
appropriate moment to switch the source of transmitted text.
This method is used when no exchange of the rtt-mixer media tag has
occurred in the session setup. Support for the method can however be
expressed by the focus by the SIP media tag rtt-mixer=text-mixer.
8. IANA Considerations
This document Introduces the SIP media tag rtt-mixer, with a comma-
separated parameter list containing the following possible values:
rtp-mixer
t140-mixer
text-mixer
rtp-mixer indicates capability for using the RTP-mixer based
presentation of multi-party text. t140-mixer indicates capability for
using the T.140 control code source indicators in a mixer. text-mixer
indicates capability for using text-level control over formatting and
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presentation of multi-party text presentation.
9. Security Considerations
The security considerations valid for RFC 4103 and RFC 3550 are valid
also for the multi-party sessions with text.
10. Congestion considerations
The congestion considerations described in RFC 4103 are valid also
for multi-party use of the real-time text RTP transport. A risk for
congestion may appear if a number of conference participants are
active transmitting text simultaneously, because this multi-party
transmission method does not allow multiple sources of text to
contribute to the same packet.
In situations of risk for congestion, the Focus UA MAY combine
packets from the same source to increase the transmission interval
per source up to one second. Local conference policy in the Focus UA
may be used to decide on which streams shall be selected for such
transmission frequency reduction.
11. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[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.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, June 2005.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
Initiation Protocol (SIP) Event Package for Conference
State", RFC 4575, August 2006.
[RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol
(SIP) Call Control - Conferencing for User Agents",
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BCP 119, RFC 4579, August 2006.
[T.140] ITU-T, "Protocol for multimedia application text
conversation", 1998,
<http://www.itu.int/rec/T-REC-T.140/en>.
Authors' Addresses
Gunnar Hellstrom
Omnitor
Box 92054
Stockholm SE-120 06
SE
Phone: +46 858900056
Fax: +46 858900051
Email: gunnar.hellstrom@omnitor.se
URI: www.omnitor.se
Arnoud van Wijk
Real-Time Text Taskforce (R3TF)
NL
Fax: +31 412614000
Email: arnoud@realtimetext.org
URI: www.realtimetext.org
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