Internet Engineering Task Force                     Gonzalo Camarillo
Internet draft                                             Jan Holler
                                                    Goran AP Eriksson
                                                             Ericsson
                                                            June
                                                            July 2001
                                                 Expires December 2001
                                       <draft-ietf-mmusic-fid-02.txt> January 2002
                                       <draft-ietf-mmusic-fid-03.txt>

                     Grouping of m media lines in SDP

Status of this Memo

   This document is an Internet-Draft and is in full conformance with
      all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
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   http://www.ietf.org/ietf/1id-abstracts.txt
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Abstract

   This document defines two SDP attributes: "groupe" "group" and "mid". They
   allow to group together several "m" lines for two different
   purposes: for lip synchronization and for receiving media from a
   single flow (several media streams), encoded in different formats
   during a particular session, in different ports and host interfaces.

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                    Grouping of m media lines in SDP

TABLE OF CONTENTS

   1     Terminology................................................2
   2     Media stream identification attribute........................2
   2   Groupe attribute.............................................2 attribute......................2
   3     Group attribute............................................2
   4     Lip Synchronization (LS).....................................3
   4 (LS)...................................3
   5     Flow Identification (FID)....................................3
   4.1 (FID)..................................3
   5.1   SIP and cellular access......................................3
   4.2 access....................................4
   5.2   DTMF tones...................................................4
   5 tones.................................................4
   5.3   Media flow definition........................................4
   6 definition......................................5
   5.4   FID semantics................................................4
   7 semantics..............................................5
   5.4.1 Interactions of "groupe" "group" with other media level attributes...5
   8 attributes..6
   5.4.2 Media in parallel..........................................7
   5.4.3 DTMF tones encoded as telephony events.....................8
   6     Usage of the "groupe" "group" attribute in SIP.......................6
   8.1 SIP......................8
   6.1   Media alignment............................................9
   6.2   Mid value in responses.....................................9
   6.3   Group value in responses...................................9
   6.4   Backward compatibility.......................................6
   8.2 Caller compatibility....................................10
   6.4.1 Client does not support fid..................................6
   8.3 Callee "group"...........................11
   6.4.2 Server does not support fid..................................6 "group"...........................11
   7     Acknoledgements...........................................11
   8     References................................................11
   9   Acknoledgements..............................................7
   10   References..................................................7
   11     Authors³ Addresses..........................................7

1. Addresses........................................12

1 Terminology

   In this document, the key words "MUST", "MUST NOT", "REQUIRED",
   "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
   and "OPTIONAL" are to be interpreted as described in RFC 2119 [1]
   and indicate requirement levels for compliant implementations.

2. Media stream identification attribute

   A new "media stream identification" media attribute is defined. It
   is used for identifying media streams within a session description.
   Its formatting in SDP [2] is described by the following BNF:

         mid-attribute      = "a=mid:" identification-tag
         identification-tag = token

   The identification tag is MUST be unique within the SDP session
   description.

2.

3. Group attribute

   A new "group" session level attribute is defined. It is used for
   grouping together different media streams. Its formatting in SDP is
   described by the following BNF:

         groupe-attribute

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                    Grouping of media lines in SDP

         group-attribute    = "a=groupe:" "a=group:" semantics space
                              2*(space identification-tag)
         semantics          = "LS" | "FID"

   This document defines two standard semantics: LS (Lip
   Synchronization) and FID (Flow Identification). If in the future it
   was needed to standardize further semantics they would need to be
   defined in a standards track document. However, defining new
   semantics apart from LS and FID is discouraged. Instead, it is
   RECOMMENDED to use other session description mechanisms such as
   SDPng [1].

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                      Grouping of m lines in SDP [3].

   There might MAY be several "a=groupe" "a=group" lines in a session description.
   "a=groupe"

   "a=group" lines that contain identification-tags that are not
   present in the session description are MUST be simply ignored. The
   application acts as if the "a=groupe" "a=group" line did not exist.

3.

4. Lip Synchronization (LS)

   The play out of media streams that are grouped together using LS
   semantics have to MUST be synchronized. Synchronization is typically
   performed using RTCP, which provides enough information to map time
   stamps from the different streams into a wall clock.

   The following example shows a session description where the audio
   and the video stream have to be synchronized.

         v=0
         o=Laura 289083124 289083124 IN IP4 first.example.com
         t=0 0
         c=IN IP4 131.160.1.112
         a=groupe:LS
         a=group:LS 1 2
         m=audio 30000 RTP/AVP 0
         a=mid:1
         m=video 30002 RTP/AVP 31
         a=mid:2

4.
         m=audio 30004 RTP/AVP 0
         a=mid:3

   Note that although the third media stream is not present in the
   group line it still contains an mid attribute (mid:3). All the "m"
   lines of a session description that uses "group" MUST be identified
   with an "mid" attribute regardless of whether they appear or not in
   the group line(s).

5. Flow Identification (FID)

   An "m" line in an SDP session description defines a media stream.
   However, SDP does not define what a media stream is. To find the

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                    Grouping of media lines in SDP

   definition of a media stream we have to go to the RTSP
   specification. The RTSP RFC [2] [4] defines a media stream as "a single
   media instance, e.g., an audio stream or a video stream as well as a
   single whiteboard or shared application group. When using RTP, a
   stream consists of all RTP and RTCP packets created by a source
   within an RTP session".

   This definition assumes that a single audio (or video) stream maps
   into an RTP session. To find the definition of an RTP session we go
   to the RTP specification. The RTP RFC [3] [5] defines an RTP session as
   follows: "For each participant, the session is defined by a
   particular pair of destination transport addresses (one network
   address plus a port pair for RTP and RTCP)".

   However,

   While the previous definitions cover the most common cases, there
   are situations where a single media instance, (e.g., an audio stream
   or a video stream) is sent using more than one RTP session. Two
   examples (among many others) of this kind of situation are cellular
   systems using SIP [4] [6] and systems receiving DTMF tones on a
   different host than the voice.

4.1

5.1 SIP and cellular access

   Systems using a cellular access and SIP as a signalling protocol
   need to receive media over the air. During a session the media can
   be encoded using different codecs. The encoded media has to traverse
   the radio interface. The radio interface is generally characterized
   by being bit error prone and associated with relatively high packet

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                      Grouping of m lines in SDP
   transfer delays. In addition, radio interface resources in a
   cellular environment are scarce and thus expensive, which calls for
   special measures in providing a highly efficient transport [5]. [7]. In
   order to get an appropriate speech quality in combination with an
   efficient transport, precise knowledge of codec properties are
   required so that a proper radio bearer for the RTP session can be
   configured before transferring the media. These radio bearers are
   dedicated bearers per media type, i.e. codec.

   Cellular systems typically configure different radio bearers on
   different port numbers. Therefore, incoming media has to have
   different destination port numbers for the different possible codecs
   in order to be routed properly to the correct radio bearer. Thus,
   this is an example in which several RTP sessions are used to carry a
   single media instance (the encoded speech from the sender).

4.2

5.2 DTMF tones

   Some voice sessions include DTMF tones. Sometimes the voice handling
   is performed by a different host than the DTMF handling. [6] [8]
   contains several examples of how application servers in the network
   gather DTMF tones for the user while the user receives the encoded
   speech on his user agent. In this situations it is necessary to
   establish two RTP sessions: one for the voice and the other for the

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                    Grouping of media lines in SDP

   DTMF tones. Both RTP sessions are logically part of the same media
   instance.

5.

5.3 Media flow definition

   The previous examples show that the definition of a media stream in
   [2] has to be updated.
   [4] do not cover some scenarios. It cannot be assumed that a single
   media instance maps into a single RTP session. Therefore, we
   introduce the definition of a media flow:

   Media flow consists of a single media instance, e.g., an audio
   stream or a video stream as well as a single whiteboard or shared
   application group. When using RTP, a media flow comprises one or
   more RTP sessions.

   For instance, in a two party call where the voice exchanged can be
   encoded using GSM or PCM, the receiver wants to receive GSM on a
   port number and PCM on a different port number. Two RTP sessions
   will be established, one carrying GSM and the other carrying PCM.

   At any particular moment just one codec is in use. Therefore, at any
   moment one of the RTP sessions will not transport any voice. Here
   the systems are dealing with a single media flow, but two RTP
   sessions.

6.

5.4 FID semantics

   Several "m" lines grouped together using FID semantics form a media
   flow. A media agent handling a media flow that comprises several "m"

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                      Grouping of m lines in SDP
   lines sends media to different destinations (IP address/port number)
   depending on the codec used at any moment. If several "m" lines
   contain the codec used media is sent to different destinations in
   parallel.

   For instance, a SIP user agent receives an INVITE with the following
   body:

         v=0
         o=Laura 289083124 289083124 IN IP4 second.example.com
         t=0 0
         c=IN IP4 131.160.1.112
         a=groupe:FID
         a=group:FID 1 2 3
         m=audio 30000 RTP/AVP 0 3
         a=rtpmap:3 GSM/8000
         a=mid:1
         m=audio 30002 RTP/AVP 8 97
         a=rtpmap:97 AMR/8000
         a=fmtp:97 mode-set=0,2,5,7; mode-change-period=2; mode-change-
      neighbor; maxframes=1
         a=mid:2
         m=audio 30004 RTP/AVP 0 8
         a=mid:3

   At a particular point of time, if

   This would be the media agent is sending PCM u-
   law (payload 0) it SDP sent by a terminal using a cellular access.
   The terminal supports GSM on port 30000 and AMR on port 30002. When
   the remote party sends GSM it will send RTP packets to ports 30000 and 30004 (first
   and third "m" lines). If it port number
   30000. When AMR is sending PCM A-law (payload 8) it
   sends RTP the codec chosen, packets will be sent to ports 30002 and 30004 (second and third "m"
   lines). port

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                    Grouping of media lines in SDP

   30002. Note that if several "m" lines with the same fid value contain remote party can switch between both codecs
   dynamically in the
   same codec middle of the session.

   In the previous example a system receives media agent MUST send media over several RTP sessions
   at on the same time.

7 IP
   address on different port numbers. The following example shows how a
   system can receive different codecs on different IP addresses.

         v=0
         o=Laura 289083124 289083124 IN IP4 third.example.com
         t=0 0
         c=IN IP4 131.160.1.112
         a=group:FID 1 2
         m=audio 20000 RTP/AVP 0
         c=IN IP4 131.160.1.111
         a=rtpmap:0 PCMU/8000
         a=mid:1
         m=audio 30002 RTP/AVP 97
         a=rtpmap:97 AMR/8000
         a=fmtp:97 mode-set=0,2,5,7; mode-change-period=2; mode-change-
      neighbor; maxframes=1
         a=mid:2

   The cellular terminal of this example only supports the AMR codec.
   However, many current IP phones only support PCM (payload 0). In
   order to be able to interoperate with them, the cellular terminal
   uses a transcoder whose IP address is 131.160.1.111. The cellular
   terminal includes in its SDP support for PCM at that IP address.
   Remote systems will send AMR directly to the terminal but PCM will
   be sent to the transcoder. The transcoder will be configured (using
   whatever method) to convert the incoming PCM audio to AMR and send
   it to the terminal.

5.4.1 Interactions of "groupe" "group" with other media level attributes

   Media level attributes affect a media stream defined by an "m" line.
   The presence of "groupe" "group" does not modify this behavior.

   For instance, a

   This property can be used for different purposes. The example below
   shows one possible use of this. A SIP user agent receives an INVITE
   with the following body:

         v=0
         o=Laura 289083124 289083124 IN IP4 third.example.com forth.example.com
         t=0 0
         c=IN IP4 131.160.1.112
         a=groupe:FID
         a=group:FID 1 2
         m=audio 30000 RTP/AVP 0
         a=mid:1
         m=audio 30002 RTP/AVP 8
         a=recvonly
         a=mid:2

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                    Grouping of media lines in SDP

   The media agent knows that at a certain moment it can send either
   PCM u-law to port number 30000 or PCM A-law to port number 30002.
   However, the media agent also knows that the other end will only
   send PCM u-law (payload 0).

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                      Grouping of m lines in SDP

   Note that the "groupe" "group" attribute used with FID semantics allows to
   express uni-directional codecs for a bi-directional media flow, as
   it is shown in the example above.

8. Usage

5.4.2 Media in parallel

   It can happen that different "m" lines grouped together using FID
   semantics contain the same codec. The SDP below shows one example of
   this situation:

         v=0
         o=Laura 289083124 289083124 IN IP4 fifth.example.com
         t=0 0
         c=IN IP4 131.160.1.112
         a=groupe:FID 1 2 3
         m=audio 30000 RTP/AVP 0
         a=mid:1
         m=audio 30002 RTP/AVP 8
         a=mid:2
         m=audio 20000 RTP/AVP 0 8
         c=IN IP4 131.160.1.111
         a=recvonly
         a=mid:3

   If several "m" lines contain the "groupe" attribute in SIP

   SIP [4] is an application layer protocol for establishing,
   terminating and modifying multimedia sessions. SIP carries session
   descriptions codec used at a certain point of
   time media MUST be sent to different destinations in the bodies parallel.

   At a particular point of time, if the SIP messages but media agent is independent
   from sending PCM u-
   law (payload 0) it sends RTP packets to 131.160.1.112 on port 30000
   and to 131.160.1.111 on port 20000 (first and third "m" lines). If
   it is sending PCM A-law (payload 8) it sends RTP packets to
   131.160.1.112 on port 30002 and to 131.160.1.111 on port 20000
   (second and third "m" lines).

   The system that generated the protocol used for describing sessions. SDP [7] above supports PCM u-law on port
   30000 and PCM A-law on port 30002. Besides, it uses an application
   server whose IP address is one of
   the protocols 131.160.1.111 that can be used for this purpose.

   Appendix B records all the
   conversation. That is why the application server always receives a
   copy of [4] describes the usage audio stream regardless of SDP in relation to SIP. It
   states: "The caller the codec being used at any
   given moment (it receives both u-law and callee align their media description so A-law).

   Note that if several "m" lines grouped together using FID semantics
   contain the same codec the media agent MUST send media over several
   RTP sessions at the same time.

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                    Grouping of media lines in SDP

5.4.3 DTMF tones encoded as telephony events

   DTMF tones can be transmitted using a regular voice codec or can be
   transmitted as telephony events. The RTP payload for DTMF tones
   treated as telephone events is described in RFC 2833 [9]. Below
   there is an example of an SDP session description using FID
   semantics and this payload type.

         v=0
         o=Laura 289083124 289083124 IN IP4 sixth.example.com
         t=0 0
         c=IN IP4 131.160.1.112
         a=group:FID 1 2
         m=audio 30000 RTP/AVP 0
         a=mid:1
         m=audio 20000 RTP/AVP 97
         c=IN IP4 131.160.1.111
         a=rtpmap:97 telephone-events
         a=mid:2

   The remote party would send PCM encoded voice (payload 0) to
   131.160.1.112 and DTMF tones encoded as telephony events to
   131.160.1.111. Note that only voice or DTMF is sent at a particular
   point of time. When DTMF tones are sent the first media stream does
   not carry any data and when voice is sent there is no data in the
   second media stream. FID semantics provide different destinations
   for alternative codecs.

   Some systems implement the RTP payload defined in RFC 2833, but when
   they send DTMF tones they do not mute the voice channel. Therefore,
   effectively they are sending two copies of the same DTMF tone:
   encoded as voice and encoded as a telephony event. When the receiver
   gets both copies it typically uses the telephony event rather than
   the tone encoded as voice. FID semantics MUST NOT be used in this
   context to group both media streams since such a system is not using
   alternative codecs but rather different parallel encodings for the
   same information.

6. Usage of the "group" attribute in SIP

   SDP descriptions are used by several different protocols, SIP among
   them. We include a section about SIP because the "group" attribute
   will most likely be used mainly by SIP systems.

   SIP [6] is an application layer protocol for establishing,
   terminating and modifying multimedia sessions. SIP carries session
   descriptions in the bodies of the SIP messages but is independent
   from the protocol used for describing sessions. SDP [2] is one of
   the protocols that can be used for this purpose.

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                    Grouping of media lines in SDP

6.1 Media alignment

   Appendix B of [6] describes the usage of SDP in relation to SIP. It
   states: "The caller and callee align their media description so that
   the nth media stream ("m=" line) in the caller³s session description
   corresponds to the nth media stream in the callee³s description."

   The presence of the "group" attribute in an SDP session description
   does not modify this behavior.

   Since the "mid" attribute provides a means to label "m" lines it
   would be possible to perform media alignment using "mid" labels
   rather than matching nth "m" lines. However this would not bring any
   gain and would add complexity to implementations. Therefore SIP
   systems MUST perform media alignment matching nth lines regardless
   of the presence of the "group" or "mid" attributes.

6.2 Mid value in responses

   The "mid" attribute is an identifier for a particular media stream.
   Therefore, the "mid" value in the response MUST be the same as the
   "mid" value in the request. Besides, subsequent requests such as re-
   INVITEs MUST use the same "mid" value for the already existing media
   streams.

6.3 Group value in responses

   The "group" attribute in a response will typically be the same as
   the one received in the request. However, there are situations when
   both are different. In these situations the "group" value to be used
   in the session is the one present in the response.

        Note the "group value in the response" really refers to the
        "group" value in the last SDP exchanged between both parties.
        That is, if in the establishment of a particular session
        (INVITE-200 OK-ACK) SDPs are present in the 200 OK and in the
        ACK (not in the INVITE), the "group" value to be used during
        the session will be the one in the ACK.

   The example below shows how the callee refuses a media stream
   offered by the caller setting its port number to zero. The "mid"
   value corresponding to that media stream is removed from the "group"
   value in the response.

   SDP in the INVITE from caller to callee:

         v=0
         o=Laura 289083124 289083124 IN IP4 seventh.example.com
         t=0 0
         c=IN IP4 131.160.1.112
         a=group:FID 1 2 3
         m=audio 30000 RTP/AVP 0
         a=mid:1

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                    Grouping of media lines in SDP

         m=audio 30002 RTP/AVP 8
         a=mid:2
         m=audio 30004 RTP/AVP 3
         a=mid:3

   SDP in the INVITE from callee to caller:

         v=0
         o=Bob 289083125 289083125 IN IP4 fifth.example.com
         t=0 0
         c=IN IP4 131.160.1.113
         a=group:FID 1 3
         m=audio 20000 RTP/AVP 0
         a=mid:1
         m=audio 0 RTP/AVP 8
         a=mid:2
         m=audio 20002 RTP/AVP 3
         a=mid:3

   Note that although the media stream was refused the "mid" value was
   still included.

6.4 Backward compatibility

   An application that wants to be compliant to this specification MUST
   support both "group" and "mid". Supporting just one of them would be
   useless.

   A SIP entity that receives a request that contains "group" and "mid"
   attributes, understands them and it is willing to use the grouping
   semantics offered returns a response that also contains "group" and
   "mid" attributes. This way, the client that issued the request knows
   that the server understood this extension.

   Note that grouping of m lines is always requested by the nth media stream ("m=" line) in the caller³s session description
   corresponds to issuer of
   the nth media stream in request (the client), never by the callee³s description."

   The presence issuer of the "groupe" attribute response (the
   server). Since there is no response to a response in an SDP session description
   does SIP, a server
   that requested grouping in a response would not modify this behavior.

8.1 Backward compatibility know whether the
   "group" attribute was accepted by the client or not. A server that
   wants to group media lines should issue another request after having
   responded to the first one (a re-INVITE for instance).

   This document does not define any SIP "Require" header. Therefore,
   if one of the SIP user agents does not understand the "groupe" "group"
   attribute the standard SDP fall back mechanism is used.

   A system client that understands the "groupe" attribute MUST add an "mid"
   attribute does not want to every "m" line perform grouping of media lines in any SDP a
   session description SHOULD NOT add "mid" lines either. The presence of "mid"
   lines would not be of any use for the server. Even if the server can
   see that the client supported "mid" (and obviously "group" also) it
   generates.

8.2 Caller
   would be impossible to know which particular semantics are supported
   (LS or/and FID).

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                    Grouping of media lines in SDP

6.4.1 Client does not support "groupe" "group"

   This situation does not represent a problem. The SDP in the INVITE
   will problem because grouping
   requests is always performed by clients, not contain any "mid" attribute. The callee knows that by servers. If the
   caller
   client does not support "groupe".

8.3 Callee "group" this attribute will just not be
   used.

6.4.2 Server does not support "groupe" "group"

   The callee server will ignore the "groupe" "group" attribute, since it does not
   understand it. it (it will also ignore the "mid" attribute). For LS
   semantics, the callee server might decide to perform or to not perform
   synchronization between media streams.

   For FID semantics, the callee server will consider that the session
   comprises several media streams.

   Different implementations would behave in different ways.

   In the case of audio and different "m" lines for different codecs an
   implementation might decide to act as a mixer with the different
   incoming RTP sessions, which is the correct behavior.

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                      Grouping of m lines in SDP

   An implementation might also decide to refuse the request (e.g. 488
   Not acceptable here or 606 Not Acceptable) because it contains
   several "m" lines. In this case, the callee server does not support the
   type of session that the caller wanted to establish. In case the
   caller
   client is willing to establish a simpler session anyway, he should
   re-try the request without "groupe" "group" attribute and only one "m" line
   per flow.

9.

7. Acknowledgments

   The authors would like to thank Jonathan Rosenberg, Adam Roach and
   Orit Levin for their feedback on this document.

10.

8. References

   [1] S. Bradner, "Key words for use in RFCs to Indicate Requirement
   Levels", RFC 2119, IETF; March 1997.

   [2] M. Handley/V. Jacobson, "SDP: Session Description Protocol", RFC
   2327, IETF; April 1998.

   [3] D. Kutscher/J. Ott/C. Bormann, "Session Description and
   Capability Negotiation", draft-ietf-mmusic-sdpng-00.txt, IETF; April
   2001. Work in progress.

   [2]

   [4] H. Schulzrinne/A. Rao/R. Lanphier, "Real Time Streaming Protocol
   (RTSP)", RFC 2326, IETF; April 1998.

   [3]

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                    Grouping of media lines in SDP

   [5] H. Schulzrinne/S. Casner/R. Frederick/V. Jacobson, "RTP: A
   Transport Protocol for Real-Time Applications", RFC 1889, IETF;
   January 1996.

   [4]

   [6] M. Handley/H. Schulzrinne/E. Schooler/J. Rosenberg, "SIP:
   Session Initiation Protocol", RFC 2543, IETF; Mach 1999.

   [5]

   [7] L. Westberg/M. Lindqvist, "Realtime Traffic over Cellular Access
   Networks", draft-westberg-realtime-cellular-03.txt, draft-westberg-realtime-cellular-04.txt, IETF; November
   2000. June 2001.
   Work in progress.

   [6]

   [8] J. Rosemberg/P.Mataga/H.Schulzrinne, Rosenberg/P.Mataga/H.Schulzrinne, "An Applcation Application Server
   Component Architecture for SIP", draft-rosenberg-sip-app-components-
   00.txt, IETF; November 2000. Work in progress.

   [7] M. Handley/V. Jacobson, "SDP: Session Description Protocol",

   [9] H. Schulzrinne/S. Petrack, "RTP Payload for DTMF Digits,
   Telephony Tones and Telephony Signals", RFC
   2327, 2833, IETF; April 1998.

11. May 2000.

9. Authors³ Addresses

   Gonzalo Camarillo
   Ericsson
   Advanced Signalling Research Lab.
   FIN-02420 Jorvas
   Finland
   Phone: +358 9 299 3371
   Fax: +358 9 299 3052
   Email: Gonzalo.Camarillo@ericsson.com

   Jan Holler
   Ericsson Research

Camarillo/Holler/Eriksson                                            7
                      Grouping of m lines in SDP
   S-16480 Stockholm
   Sweden
   Phone: +46 8 58532845
   Fax: +46 8 4047020
   Email: Jan.Holler@era.ericsson.se

   Goran AP Eriksson
   Ericsson Research
   S-16480 Stockholm
   Sweden
   Phone: +46 8 58531762
   Fax: +46 8 4047020
   Email: Goran.AP.Eriksson@era.ericsson.se

Camarillo/Holler/Eriksson                                            8                                           12