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Versions: 00 01 02 03 04 draft-hellstrom-mmusic-multi-party-rtt

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


   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

   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
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   Internet-Drafts are draft documents valid for a maximum of six months
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   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 September 15, 2011.

Copyright Notice

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   Copyright (c) 2011 IETF Trust and the persons identified as the
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   described in the Simplified BSD License.

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",
   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

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 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,

Authors' Addresses

   Gunnar Hellstrom
   Box 92054
   Stockholm  SE-120 06

   Phone: +46 858900056
   Fax:   +46 858900051
   Email: gunnar.hellstrom@omnitor.se
   URI:   www.omnitor.se

   Arnoud van Wijk
   Real-Time Text Taskforce (R3TF)

   Fax:   +31 412614000
   Email: arnoud@realtimetext.org
   URI:   www.realtimetext.org

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