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Internet Engineering Task Force                               SIPPING WG
Internet Draft                                              J. Rosenberg
                                                             dynamicsoft
                                                          H. Schulzrinne
                                                             Columbia U.
draft-ietf-sipping-conferencing-models-01.txt
July 1, 2002
Expires: January 2003


               Models for Multi Party Conferencing in SIP

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
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress".

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt

   To view the list Internet-Draft Shadow Directories, see
   http://www.ietf.org/shadow.html.


Abstract

   The Session Initiation Protocol (SIP) can support multi-party
   conferencing in many different ways. In this draft, we define the
   various multi-party conferencing models, and for each, discuss how
   they are used and then analyze their relative benefits and drawbacks.











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                           Table of Contents



   1          Introduction ........................................    4
   2          End System Mixing ...................................    4
   2.1        Inviting Users to Join ..............................    6
   2.2        Users Joining .......................................    7
   2.3        Scalability .........................................    7
   2.4        Location of Service Logic ...........................    8
   2.5        Discovering Participant Identities ..................    8
   3          Large-Scale Multicast Conferences ...................    8
   3.1        Inviting Users to Join ..............................    9
   3.2        Users Joining .......................................    9
   3.3        Scalability .........................................    9
   3.4        Location of Service Logic ...........................   10
   3.5        Discovering Participant Identities ..................   10
   4          Dial-In Conference Servers ..........................   10
   4.1        Inviting Users to Join ..............................   12
   4.2        Users Joining .......................................   13
   4.3        Scalability .........................................   13
   4.4        Location of Service Logic ...........................   14
   4.5        Discovering Participant Identities ..................   14
   5          Ad-hoc Centralized Conferences ......................   14
   5.1        Inviting Users to Join ..............................   16
   5.2        Users Joining .......................................   16
   5.3        Scalability .........................................   16
   5.4        Location of Service Logic ...........................   19
   5.5        Discovering Participant Identities ..................   19
   6          Dial-Out Conferences ................................   19
   6.1        Inviting Users to Join ..............................   19
   6.2        Users Joining .......................................   19
   6.3        Scalability .........................................   20
   6.4        Location of Service Logic ...........................   21
   6.5        Discovering Participant Identities ..................   21
   7          Centralized Signaling, Distributed Media ............   21
   7.1        Inviting Users to Join ..............................   24
   7.2        Users Joining .......................................   24
   7.3        Scalability .........................................   24
   7.4        Location of Service Logic ...........................   24
   7.5        Discovering Participant Identities ..................   25
   8          Summary of Models ...................................   25
   9          Security Considerations .............................   25
   10         Conclusion ..........................................   26
   11         Acknowledgements ....................................   26



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   12         Authors Addresses ...................................   26
   13         Normative References ................................   26
   14         Informative References ..............................   27
















































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1 Introduction

   The Session Initiation Protocol (SIP) [1] has been defined for the
   establishment, maintenance, and termination of calls between one or
   more users. However, despite its origins as a large scale multiparty
   conferencing protocol, SIP is used today primarily for point to point
   calls. This configuration is the focus of the SIP specification and
   most of its extensions. As a result, there is a lot of confusion
   about how SIP supports multi-party conferencing.

   We seek to remedy this problem by describing, in a consistent and
   complete fashion, the various multi-party conferencing models
   supported by standard SIP. For each model, we discuss:

        o How the model works.

        o How users are invited to join.

        o How users can join an existing conference without being
          invited

        o How well the model scales.

        o Which entities need to be aware of the model.

        o How participants learn about each other.

   We also identify missing pieces and reccomend standard activity to
   fill them in. This document itself does not define any new extensions
   of any kind. However, several scenarios discussed in the draft make
   use of existing extensions to SIP.

2 End System Mixing

   The first model we call "end system mixing". In this model, user A
   calls user B, and they have a conversation. At some point later, A
   decides to conference in user C. To do this, A calls C, using a
   completely separate SIP call. This call uses a different Call-ID,
   different tags, etc. There is no call set up directly between B and
   C. A receives media streams from both B and C, and mixes them. A
   sends a stream containing A's and C's streams to B, and a stream
   stream containing A's and B's streams to C.

   This model is depicted graphically in Figure 1.


   Basically, user A handles both signaling and media mixing. B and C
   are unaware of the multi-party call, from a SIP perspective at least.



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                                      +----------+
                                      |          |
                                   -- |          |
                                ---   |   B      |
                    SIP call ---      |          |
                          ---      .. |          |
                       ---       ..   +----------+
                     --       ...
                           ...
          +----------+   ..   RTP
          |          | ..
          |          |
          |   A      |  ..
          |          |    ..
          |          |      .. RTP
          +----------+        ..
                        --      ..
                          --      ..
                            ---     . +----------+
                               --     |          |
                                 --   |          |
                       SIP call    -- |          |
                                      |    C     |
                                      |          |
                                      +----------+




   Figure 1: Three Way Calling using End System Mixing


   From an RTP perspective, A is a mixer, and so the RTCP reports from A
   will contain SDES information that indicates the existence of an
   additional party in the media stream.

   Note that this model has the serious drawback that the conference
   ends when the mixing UA leaves the call.


        OPEN ISSUE: Another problem with this approach is that
        there is no specific way for A to determine when a
        signaling message it receives was meant just for it, or for
        the entire conference. For example, if B sends a REFER to



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        A, pointing to user D, was this REFER meant for A alone, or
        for A and C? If it was meant for A and C, presumably A
        would act upon the REFER and send it to C as well. C too
        would act on the REFER. This would cause two separate
        REFER-triggered INVITEs to get routed to D. How would D
        know that both INVITEs need to be mixed together as a
        conference? What if it cannot support this capability?

   Because the three-way calling approach works only for the most basic
   case, we do not recommend it as a general solution.

2.1 Inviting Users to Join

   Any user in the conference can invite another user to join, so long
   as they are capable of performing the required mixing and signaling
   functions. To invite a new user to join, a user in the conference
   simply calls them using normal SIP procedures. The only difference is
   that the stream sent to that new user contains the streams received
   from the other parties in the call.

   In fact, it is acceptable for complex connectivity graphs to be
   constructed, as a result of different users inviting other users to
   join. For example, take our case of A calling B, and then calling C.
   If, later on, C calls D, C will performing the mixing of the streams
   it gets from A (which actually contain media from A and B), along
   with its own stream, and send that to D. This results in a
   connectivity graph that looks like Figure 2.



A------B
|
|
C------D



   Figure 2: Connectivity Graph



   Note, however, that there is a possibility of loops. From here, if D
   calls B, and brings that stream into the conference, a loop is
   created. This loop can be detected using the mechanisms described in
   the RTP specification [2]. However, we expect these conditions to be
   extremely rare. Presumably, D knows B is in the conference already,
   and so would not likely call B and invite them in.

   A serious problem with the more complex topologies is that the



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   departure of a participant might cause a partition of the conference
   into several sub-conferences which cannot easily be healed.

2.2 Users Joining

   In this model, there is not any explicit conference "identifier" that
   can be used to join. This conference model, by its nature, is built
   around ad-hoc conferences. However, it is still possible for a user
   to join in the following way.

   Lets say a new user, E, simply calls B, unaware even, that B is in a
   conference (E might actually be aware, but the SIP messaging is no
   different). B's softphone, recognizing that B is already in a
   conference, asks B if E should be brought into the conference right
   away. If B clicks "yes", the call to E is answered. The media stream
   sent to E contains media from B, along with the media B is already
   receiving from A.

   If B had instead clicked no, E can easily be added to the conference
   later. No SIP signaling at all is needed to do this. B simply starts
   sending the mixed media to E.

2.3 Scalability

   A drawback of this model is its scalability. Viewing the conference
   from a graph perspective, if the number of edges touching a vertex
   (its degree) equals N, the user corresponding to that vertex has to
   perform up to N separate media stream encodings. We say "up to", as
   it depends on the number of paricipants who are talking at once. If
   only one participant is talking, the non-talking "mixer" endpoints
   don't need to do any additional encoding. If everyone is talking, it
   is N encodes. Since encoding is generally a complex process, a
   typical workstation these days can handle two or three simultaneous
   encodes using a low rate codec like G.723.1. The problem can be
   mitigated somewhat by distributing the mixing responsibilities
   (making the graph deep rather than wide). However, this requires a
   conscious effort of the participants regarding who is to make the
   call to add a new user. This is unlikely to happen in practice.

   Another limitation to scalability is bandwidth. If the degree of a
   vertex is N, the user needs enough bandwidth to send and receive up
   to N streams, for a total of 2N. On a 56K modem, using a G.723.1
   codec, this limits the degree to two (remember RTP overheads). This
   limitation exists even if only one user is talking. In this case, a
   mixing host receives the encoded packet stream, and needs to send a
   copy to each participant it is connected to.

   For these reasons, this conferencing model is ideal for three-way



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   conferences (i.e., degrees of two), but doesn't scale up much higher.

2.4 Location of Service Logic

   This model does not require any extension to SIP in order to work. It
   does require knowledge of this mechanism within the UA performing the
   mixing. Non-mixing participants do not need to know anything special.

2.5 Discovering Participant Identities

   The identities of other participants in the conference is NOT known
   through SIP. Rather, it is learned through RTP. UAs with degrees
   greater than one are RTP mixers. As such, they take the RTCP SDES of
   the streams they mix, and aggregrate them into the RTCP stream sent
   out. Since RTCP messages are sent infrequently, there may be a delay
   between when a user joins, and when their presence is known to the
   other participants.

3 Large-Scale Multicast Conferences

   Large-scale multicast conferences were the original motivation for
   both the Session Description Protocol (SDP) [3] and SIP. In a large-
   scale multicast conference, one or more multicast addresses are
   allocated to the conference (more than one may be needed if layered
   encodings are in use). Each participant joins that multicast groups,
   and sends their media to those groups. Signaling is not sent to the
   multicast groups. The sole purpose of the signaling is to inform
   participants of which multicast groups to join.

   Large-scale multicast conferences are usually pre-arranged, with
   specific start and stop times (which is why this information exists
   in SDP). Protocols such as the Session Announcement Protocol (SAP)
   [4] are used to announce these conferences. However, multicast
   conferences do not need to be pre-arranged, so long as a mechanism
   exists to dynamically obtain a multicast address. SAP itself was
   originally used for this purpose; this has been supplanted by the
   malloc architecture [5], still under development.

   So, if there are N participants, there will be point-to-point SIP
   relationships with pairs of participants. Each participant sends a
   single media stream to the group, and receives up to N-1 streams at
   any time. Note that the number of streams that a user will receive
   depends on who is actually sending at any given time. If the stream
   is audio, and silence suppression is utilized, the number of streams
   a user will receive at any given time is equal to the number of users
   talking at any given time. Even for very large conferences, this is
   usually just a small number of users.




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3.1 Inviting Users to Join

   Inviting users to join is simple. Any user may invite any other user
   to join. The SIP INVITE request contains SDP that indicates multicast
   addresses for each media line. The SDP in the 200 OK response may
   actually be empty. From Section B.3 of RFC2543:


        For multicast, receive and send multicast addresses are the
        same and all parties use the same port numbers to receive
        media data. If the session description provided by the
        caller is acceptable to the callee, the callee can choose
        not to include a session description or MAY echo the
        description in the response.

   The called party then joins the multicast groups indicated in the
   SDP, using multicast protocols such as IGMP [6]. Note that it is not
   even necessary for users to send each other BYE messages when the
   conference is over, especially for large-scale, pre-arranged
   conferences that have explicit end times indicated in SDP.


        OPEN ISSUE: Do we need to specify a SIP mechanism for
        indicating that no BYE is needed?

   SDP aside, a participant can simply leave the conference at any time
   by leaving the multicast groups. No SIP signaling is needed to
   accomplish this.

3.2 Users Joining

   Users can join a conference of this type without being invited. All
   they need is the multicast addresses, ports, and codecs being used.
   These can be obtained through any number of means, including SAP. SDP
   conference descriptions can even be obtained from web pages, for
   example.

   Once the addresses are obtained, the user simply joins the
   appropriate multicast groups. Note that absolutely no SIP signaling
   is required in this case.

3.3 Scalability

   The scalability of conferences of this type is can be excellent,
   especially for audio conferences. However, it is scalable under the
   assumption that multicast itself can scale to very large groups.
   Indeed, in local networks, protocols like DVMRP [7] and PIM-DM have
   tremendous scalability for conferences with very large numbers of



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   members (the so called dense modes). Given the existence of scalable
   multicast, the primary bottleneck to scalability of this conference
   type is the periodicity of RTCP reporting. Work has been done on
   improving the problematic cases [8] so that conferences with well
   over a million members are possible.

   Scaling is a bit harder for video conferences. Unlike voice, where
   silence suppression allows for no data to be sent during periods of
   inactivity, the same is not the case for video. This makes it hard to
   scale without flooding users with lots of video packets.

   Security is also hard for multicast conferences. Group key
   management, especially when users leave the group, is very complex.

   Unfortunately, multicast has not been widely deployed across
   backbones (some do, like Internet2, but they are the exception rather
   than the rule). The MBone has collapsed, for all intents and
   purposes. Very few ISPs support multicast. As a result, wide area
   conferences are not really viable using multicast. However, these
   conferences are very suitable for LAN or enterprise conferences,
   where multicast is often deployed.

3.4 Location of Service Logic

   This conferencing model does not require any SIP extensions. It does
   require that SIP UAs are prepared to receive SIP invitations with
   multicast addresses in the SDP. These UAs need to be prepared to
   mirror the SDP in the response. They should also be prepared to never
   receive a BYE for the conference.

3.5 Discovering Participant Identities

   The identity of the participants in the session is learned entirely
   through RTCP. Each user a group multicasts RTCP packets with their
   name, email address, and so on. Note, however, that in large
   conferences, there may be significant amounts of time between a
   participant joining, and sending of their first RTCP SDES packet
   (this is for receivers only; senders will become known much faster).

4 Dial-In Conference Servers

   Dial-In conference servers closely mirror dial-in conference bridges
   in the traditional PSTN.

   A dial-in conference server acts as a normal SIP UA. Users call it,
   and the server maintains point to point SIP relationships with each
   user that calls in. The server takes the media from the users who
   dial into the same conference, mixes them, and sends out the



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   appropriate mixed stream to each participant separately.






                          +-----+
                          |     |
                          | A   |
                          |     |
                          +-----+
                            | .
                            | .
                            | .
                            | .
                            | .
                        +---------+
        +-----+         |         |         +-----+
        |     |---------|  Conf.  |---------|     |
        | D   |         |  Server |         | B   |
        |     |.........|         |.........|     |
        +-----+         |         |         +-----+
                        +---------+
                          |  .
                          |  .
                          |  .
                          |  .
                         +-----+
                         |     |
                         |  C  |
                         |     |
                         +-----+



   Figure 3: Dial-In Conference Servers



   The model is depicted in Figure 3. Note that each UA (A,B,C,D) has a
   point to point SIP and RTP relationship with the conference server.
   Each call has a different Call-ID. Each user sends their own media to
   the server. The media delivered to user A by the server is the media
   mixed from users B,C and D. The media delivered to user B by the
   server is the media mixed from users A, C and D. The media delivered
   to user C by the server is the media mixed from users A, B and D. The
   media delivered to user D is the media mixed from users A, B and C
   (this is also known as a mix-minus configuration).



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   The conference is identified by the request URI of the calls from
   each participant. This provides numerous advantages from a services
   and routing point of view [9]. For example, one conference on the
   server might be known as sip:conference34@servers.com. All users who
   call sip:conference34@servers.com are mixed together.

   Dial-In conference servers are usually associated with pre-arranged
   conferences. However, the same model applies to ad-hoc conferences.
   An ad-hoc conference server creates the conference state when the
   first user joins, and destroys it when the last one leaves. The SIP
   and RTP interfaces are identical to the pre-arranged case.

   Since conferencing servers are nothing more than SIP UASes, they can
   use any of the procedures SIP allows a UAS to use. This includes
   authentication. So, for example, a specific conference may have a
   password associated with it. Users who join are challenged (with a
   401) using digest authentication. The realm, in this case, would
   identify the conference. The INVITE that comes back would have an
   Authorization header that includes the response to the challenge -
   the name of the user trying to join the conference, and the
   conference password, hashed as defined in [10].

   Conferences can also limit the number of participants. When a new
   user tries to join, but the conference is full, the conference server
   can just reject the request with a "500 Conference Full" response.

4.1 Inviting Users to Join

   Inviting users to join is done using the SIP REFER message [11]. If
   user A wishes to ask user B to join, A would send B a REFER that
   looks like:


   REFER sip:B@example.com SIP/2.0
   From: sip:A@example.com
   To: sip:B@example.com
   Refer-To: sip:conference34@servers.com



   This would cause B to send an INVITE message to the conference
   server:


   INVITE sip:conference34@servers.com
   From: sip:B@example.com
   To: sip:conference34@servers.com
   Referred-By: sip:A@example.com



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   Since the request URI identifies the conference, this will cause B to
   get added to conference 34.

   An additional mechanism for inviting a user to join is to send REFER
   from A to the conference server, with a Refer-To containing the
   address of B. This REFER would look like:


   REFER sip:conference34@servers.com SIP/2.0
   From: sip:A@example.com
   To: sip:B@example.com
   Refer-To: sip:B@example.com



   This approach has the advantage that it doesn't require REFER support
   from B, only from the conference server.


        OPEN ISSUE: A problem with the mechanisms for adding a user
        is that they assume that the UA for user A (the one who
        adds another user to the conference) knows that it is
        indeed talking to a conference server. If the mechanisms in
        this section were applied to a UA which was not a
        conference server, the result would be the creation of
        additional call legs, but not a conference. This means that
        we require some mechanism for identifying that a URL is a
        conference URL.

4.2 Users Joining

   It is easy for users to join the conference. The participant that
   wishes to join simply sends an INVITE to the conference server, with
   the conference ID in the request URI. The conference ID (which is a
   SIP URL), can be learned by any number of means, including having it
   on a web page, receiving it in an email, etc.

   For example, if B wishes to join sip:conference34@servers.com, B
   would send the following request:


   INVITE sip:conference34@servers.com
   From: sip:B@example.com
   To: sip:conference34@servers.com



4.3 Scalability



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   The scalability of this model is limited by the bandwidth and
   processing power of the conference server. If there are N
   participants in a conference, M of which are sending media streams,
   the server will need to manage N signaling relationships, perform M
   RTP stream decodes, and N RTP stream encodes (assuming M > 0). The
   encoding is the primary processing bottleneck, and the sending of the
   N media streams is the primary bandwidth bottleneck. However,
   conference servers can be built using heavy duty hardware, and have
   high bandwith access.

   Furthermore, since we are using the request URI to name the
   conferences, we can use standard SIP techniques for distributing
   conferences across servers [9].

4.4 Location of Service Logic

   The SIP UA of the conference participants does not require any
   special processing. The RTP implementation in those clients, however,
   should support RTCP and be prepared to receive contributing sources.

   All of the new logic for providing this service resides in the
   conferencing server. No SIP extensions are needed, simply logic that
   resides above the SIP stack to manage the conferencing service.

4.5 Discovering Participant Identities

   The identities of other participants in the conference are NOT known
   through SIP. Rather, it is learned through RTP. THe conference server
   is an RTP mixer. As such, it takes the RTCP SDES of the streams it
   mixes, and aggregrates them into the RTCP stream sent out. This will
   allow participants to gradually (over a few seconds), learn the
   identities of the other participants.

   As an implementation choice, the conference server can generate the
   RTCP SDES of its participants, rather than using those provided by
   the participants. The reason for this is authenticity. A conference
   server can use SIP authentication mechanisms to identify the
   participants in the conference. This may allow it to validate the
   RTCP SDES provided by the participants. A conference server could
   remove any false information, and regenerate the SDES using the
   correct user identity as validated through SIP.

5 Ad-hoc Centralized Conferences

   In an ad-hoc centralized conference, two users A and B start with a
   normal SIP call. At some point later, they decide to add a third
   party. Instead of using end system mixing, they would prefer to use a
   conference server, as defined in Section 4.



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   The call flow for starting this kind of conference is shown in Figure
   4. Initially, A calls B (1-3). At some point, B decides to add a
   user, C, to the call, and begins the transition to a conference
   server. The first step in this process is the discovery of a
   conference server that supports ad-hoc conferences. This can be done
   through static configuration, or through any of a number of standard
   service discovery protocols, such as the Service Location Protocol
   [12].

   Once the server is discovered, a conference ID is chosen. This ID
   must be globally unique. The conference ID is then prepended to the
   server, and a SIP URL for the ad-hoc conference is formed. For
   example, if the server "a.servers.com" is used, and the unique ID is
   "a7hytaskp09878a", the SIP URL for this conference is
   sip:a7hytaskp09878a@a.servers.com.

   B then sends an INVITE to this URL (4). This creates the initial
   conference state in the server. The conference server accepts the
   call (5) and B sends an ACK (6). B then sends a REFER to A (7),
   referring them to sip:a7hytaskp09878a@a.servers.com. A accepts the
   referral (8) and this triggers an INVITE to this address (9). This
   causes A to be added to the conference. The conference server accepts
   the INVITE (10), and an ACK is generated (11). Once the NOTIFY
   request (indicating successful completion of the referred call) is
   sent from A to B (12), A responds with a 200 OK. Since B is now
   assured that A is connected through the conference server, B hangs up
   to A with a BYE (14).


        OPEN ISSUE: Its not clear that this is the best flow. An
        alternative flow is for B to REFER the conference server to
        A, using a call replacement mechanism. This is probably
        more correct, since this is not so much a transfer as a
        call leg replacement.

   Finally, B can add C to the call. This is identical to the procedures
   described in Section 4 for adding userst to the conference. First, B
   generates a REFER (16) to C. The Refer-To header contains the
   conference URL, sip:a7hytaskp09878a@a.servers.com. C responds to the
   referral with a 200 OK (17). C then INVITEs itself to the conference
   (18-20). C then generates a NOTIFY informing B that the REFER has
   completed (21).

   It is also possible to transition from a end system mixed conference
   (even one with a complex connection topology), to a centralized
   conference server. Consider a end-system mixed conference with the
   topology of Figure 2. User A wishes to transition to a centralized



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   conference server in order to add another participant. The transition
   is shown in Figures 5 and 6.



   First, user A discovers a conference server, and creates a new
   conference by sending an INVITE to it (1-3). A then REFERs the two
   end systems it is connected to (B and C), to the server (4-5 and 6-7
   respectively). This causes B to INVITE itself to the conference
   server (8-10), and C to do the same (11-13). Since C had gotten a
   REFER from B, it "passes it on" to D by sending a REFER to it (14-
   15). This causes D to join the conference server by sending it an
   INVITE (16-18).

   Once the REFER triggered INVITEs complete, notifications start to get
   sent. Since B completed first, it will be the first to send a NOTIFY
   to A (19) followed by C (21). At this point, A can terminate its legs
   to B and C (23-24 and 25-26 respectively). Since D completed its
   REFER triggered INVITE next, it generates a NOTIFY to C (27). This
   causes C to terminate its leg with D (29). The call has now
   transitioned to a centralized server.


        OPEN ISSUE: There is no way for A to know that the entire
        conference has transitioned. Also, as above, its not clear
        that a REFER from the conference server wouldn't be better.

   Once the conference has been formed, further operation is identical
   to the dial-in conferencing model of Section 4. The only difference
   in the conferences is that the conference identifier is dynamic in
   this case, and static in Section 4. This makes users asynchronously
   joining nearly impossible.

5.1 Inviting Users to Join

   Once the ad-hoc conference has been created on the server, inviting
   users proceeds as defined in Section 4.1.

5.2 Users Joining

   Once the ad-hoc conference has been created on the server, joining
   proceeds as defined in Section 4.2.

5.3 Scalability

   The scalability of this conference model is identical to that of
   dial-in conference servers, as described in Section 4.3.




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         A               B            Conference         C
                                      Server
         |(1) INVITE     |               |               |
         |-------------->|               |               |
         |(2) 200 OK     |               |               |
         |<--------------|               |               |
         |(3) ACK        |               |               |
         |-------------->|               |               |
         |               |(4) INVITE     |               |
         |               |-------------->|               |
         |               |(5) 200 OK     |               |
         |               |<--------------|               |
         |               |(6) ACK        |               |
         |(7) REFER      |-------------->|               |
         |<--------------|               |               |
         |(8) 200 OK     |               |               |
         |-------------->|               |               |
         |(9) INVITE     |               |               |
         |------------------------------>|               |
         |(10) 200 OK    |               |               |
         |<------------------------------|               |
         |(11) ACK       |               |               |
         |------------------------------>|               |
         |(12) NOTIFY    |               |               |
         |-------------->|               |               |
         |(13) 200 OK    |               |               |
         |<--------------|               |               |
         |(14) BYE       |               |               |
         |<--------------|               |               |
         |(15) 200 OK    |               |               |
         |-------------->|(16) REFER     |               |
         |               |------------------------------>|
         |               |(17) 200 OK    |               |
         |               |<------------------------------|
         |               |               |(18) INVITE    |
         |               |               |<--------------|
         |               |               |(19) 200 OK    |
         |               |               |-------------->|
         |               |               |(20) ACK       |
         |               |               |<--------------|
         |               |(21) NOTIFY    |               |
         |               |<------------------------------|
         |               |(22) 200 OK    |               |
         |               |------------------------------>|
         |               |               |               |



   Figure 4: Transitioning to ad-hoc

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      |(1) INVITE    |              |              |              |
      |---------------------------------------------------------->|
      |(2) 200 OK    |              |              |              |
      |<----------------------------------------------------------|
      |(3) ACK       |              |              |              |
      |---------------------------------------------------------->|
      |(4) REFER     |              |              |              |
      |------------->|              |              |              |
      |(5) 200 OK    |              |              |              |
      |<-------------|              |              |              |
      |(6) REFER     |              |              |              |
      |---------------------------->|              |              |
      |(7) 200 OK    |              |              |              |
      |<----------------------------|              |              |
      |              |(8) INVITE    |              |              |
      |              |------------------------------------------->|
      |              |(9) 200 OK    |              |              |
      |              |<-------------------------------------------|
      |              |(10) ACK      |              |              |
      |              |------------------------------------------->|
      |              |              |(11) INVITE   |              |
      |              |              |---------------------------->|
      |              |              |(12) 200 OK   |              |
      |              |              |<----------------------------|
      |              |              |(13) ACK      |              |
      |              |              |---------------------------->|
      |              |              |(14) REFER    |              |
      |              |              |------------->|              |
      |              |              |(15) 200 OK   |              |
      |              |              |<-------------|(16) INVITE   |
      |              |              |              |------------->|
      |              |              |              |(17) 200 OK   |
      |              |              |              |<-------------|
      |              |              |              |(18) ACK      |
      |              |              |              |------------->|
      |              |              |              |              |
      |              |              |              |              |

      A              B              C              D            Conf.
                                                                 Server



   Figure 5: Adhoc transition from end-system mixed:  part I




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5.4 Location of Service Logic

   The logic for handling the transition process must be located in at
   least one UA in the conference. All UAs that are mixers in a end
   system mixed conference must know to propagate the REFER requests
   they receive during the transition.

5.5 Discovering Participant Identities

   Once the ad-hoc conference is established, conference identities are
   determined through RTCP, as in the dial-in case.

6 Dial-Out Conferences

   Dial-out conferences are a simple variation on dial-in conferences.
   Instead of the users joining the conference by sending an INVITE to
   the server, the server chooses the users who are to be members of the
   conference, and then sends them the INVITE. Typically dial out
   conferences are pre-arranged, with specific start times and an
   initial group membership list. However, there are other means for the
   dial-out server to determine the list of participants, including user
   presence [13]. The model in no way limits the means by which the
   server determines the set of users.

   Once the users accept or reject the call from the dial out server,
   the behavior of this system is identical to the dial-in server case
   of Section 4. Thus, a dial-out conference server will generally need
   to support dial-in access for the same conference, if it wishes to
   allow joining after the conference begins.

   Note that, from the participants perspective, they will learn the
   conference identity (the URL) from the From field in the INVITE
   messages received from the server.


        OPEN ISSUE: Or is the Contact more appropriate?

6.1 Inviting Users to Join

   Once the conference is established, inviting users to join is
   identical to the scenario described in Section 4.1. Note that the URL
   to be used in the REFER is obtained from the From field of the INVITE
   received from the dial-out server.

6.2 Users Joining

   Once the conference is established, joining is identical to the
   scenario described in Section 4.2. Note that the URL to be used in



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      |(19) NOTIFY   |              |              |              |
      |<-------------|              |              |              |
      |(20) 200 OK   |              |              |              |
      |------------->|              |              |              |
      |(21) NOTIFY   |              |              |              |
      |<----------------------------|              |              |
      |(22) 200 OK   |              |              |              |
      |---------------------------->|              |              |
      |(23) BYE      |              |              |              |
      |------------->|              |              |              |
      |(24) 200 OK   |              |              |              |
      |<-------------|              |              |              |
      |(25) BYE      |              |              |              |
      |---------------------------->|              |              |
      |(26) 200 OK   |              |              |              |
      |<----------------------------|(27) NOTIFY   |              |
      |              |              |<-------------|              |
      |              |              |(28) 200 OK   |              |
      |              |              |------------->|              |
      |              |              |(29)  BYE     |              |
      |              |              |------------->|              |
      |              |              |(30) 200 OK   |              |
      |              |              |<-------------|              |
      |              |              |              |              |
      |              |              |              |              |
      |              |              |              |              |
      |              |              |              |              |
      |              |              |              |              |
      |              |              |              |              |
      |              |              |              |              |
      |              |              |              |              |
      |              |              |              |              |
      |              |              |              |              |
      |              |              |              |              |
      |              |              |              |              |
      |              |              |              |              |
      |              |              |              |              |

      A              B              C              D            Conf.
                                                                 Server



   Figure 6: Adhoc transition from end-system mixed:  part II

   the INVITE of new participants is obtained from the From field of the
   INVITE received from the dial-out server by the initial participants.

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   The scalability of this conference model is identical to that of
   dial-in conference servers, as described in Section 4.3.

6.4 Location of Service Logic

   The SIP UA of the conference participants does not require any
   special processing. The RTP implementation in those clients, however,
   should support RTCP and be prepared to receive contributing sources.

   All of the new logic for providing this service resides in the
   conferencing server. No SIP extensions are needed, simply logic that
   resides above the SIP stack to manage the conferencing service.

6.5 Discovering Participant Identities

   Once the conference is established, conference identities are
   determined through RTCP, as in the dial-in case.

7 Centralized Signaling, Distributed Media

   In this conferencing model, there is a centralized controller, as in
   the dial-in and dial-out cases. However, the centralized server
   handles signaling only. The media is still sent directly between
   participants, using either multicast or multi-unicast. Multi-unicast
   is when a user sends multiple packets (one for each recipient,
   addressed to that recipient). This is referred to as a "Decentralized
   Multipoint Conference" in H.323. Interestingly, this conference model
   is possible with baseline SIP.

   It works through third party call control [14]. The conference server
   uses re-INVITEs to each participant when a new one joins. The re-
   INVITEs add a media stream that gets sent to the new participant (and
   similarly in the reverse direction).

   Let us assume for the moment that a conference already exists with
   three participants. In this state, each participant is sending media
   directly to each other. This is because the SDP that the conference
   server has given to each participant contains three media lines, each
   of type audio, with connection addresses and ports corresponding to
   each of the three users.


   The call flow from here is shown in Figure 7. In the figure, the word
   after the INV or SIP response code refers to the connection
   adress(es) in the SDP in the message. +X means the addition of a
   stream with X as the recipient address.

   A new participant joins the conference. It does so by sending an



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   INVITE (1)to the server, with the conference ID in the request URI.
   The SDP in the INVITE contains a single media stream, with an IP
   address and port where it would like to receive media (D). The 200
   response from the conference server (2) contains a single media line
   with an IP address of 0.0.0.0 and a random port, indicating hold.

   The next step is for the server to obtain two more addresses where
   the new participant will be receiving media (it already has one from
   the original INVITE). To do this, it sends a re-INVITE to the new
   participant (4). This re-INVITE contains two additional media streams
   (for three total), all three of which are on hold. The 200 response
   to the re-INVITE (5) contains two additional IP addresses and ports
   where the user is willing to receive media.

   Now the server needs to inform the other parties that they should
   begin sending media to the new user. It first sends a re-INVITE to
   user C (7). This re-INVITE adds an additional media stream to the two
   already that C has been sending. This new media stream uses one of
   the three connection addresses and ports returned by D in message
   (5). Call this address/port D1. The other two are D2 and D3. The 200
   OK response from user C (8) contains the address and port where C is
   willing to receive a new, third media stream. Call this port C3. The
   server holds on to this port, as it will use it later on, sending it
   to D, so that D sends media there. At this point, however, C can
   begin sending media to D.

   This re-INVITE process happens for B and for A as well. In the re-
   INVITE to B (10), the server adds an additional media line (above the
   two already in use by C) using address/port D2. The response (11)
   contains a new address/port to send media to B. Call this port B3. In
   the re-INVITE to A (13), the server adds an additional media line
   using address/port D3. The response (14) contains a new address/port
   to send media to A. Call this port A3.

   Finally, the server sends a re-INVITE (15) to the new party. This
   re-INVITE takes all three streams off hold, and updates their
   connection addresses and ports with C3, B3, and A3, respectively. The
   200 OK response (16) returns the same ports and addresses returned in
   message (5) (as noted in [14], these addresses/ports MUST NOTchange).
   Now, D can send media to A,B and C.

   The result of these manipulations is, indeed, a full mesh of unicast
   RTP streams between all participants. Unlike the case of end system
   mixing, the stream sent by any participant to all of the others is
   identical. Each particpant needs to mix, but it mixes the media it
   receives, and plays that out the speakers. This is normal behavior
   for multiple streams of the same type. Note that the SIP relationship
   is still point-to-point. There are four calls at the end of Figure 7,



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    |             |              |               |(1) INV D      |
    |             |              |               |-------------->|
    |             |              |               |(2) 200 hold   |
    |             |              |               |<--------------|
    |             |              |               |(3) ACK        |
    |             |              |               |-------------->|
    |             |              |               |(4) INV 3held  |
    |             |              |               |<--------------|
    |             |              |               |(5) 200 3recv  |
    |             |              |               |-------------->|
    |             |              |               |(6) ACK        |
    |             |              |               |<--------------|
    |             |              | (7) INV +D1   |               |
    |             |              |<------------------------------|
    |             |              | (8) 200 +C3   |               |
    |             |              |------------------------------>|
    |             |              | (9) ACK       |               |
    |             |              |<------------------------------|
    |             |(10) INV +D2  |               |               |
    |             |<---------------------------------------------|
    |             |(11) 200 +B3  |               |               |
    |             |--------------------------------------------->|
    |             |(12) ACK      |               |               |
    |             |<---------------------------------------------|
    |(13) INV +D3 |              |               |               |
    |<-----------------------------------------------------------|
    |(14) 200 +A3 |              |               |               |
    |----------------------------------------------------------->|
    |(15) ACK     |              |               |               |
    |<-----------------------------------------------------------|
    |             |              |               |(16) INV A3,B3,C3
    |             |              |               |<--------------|
    |             |              |               |(17) 200       |
    |             |              |               |-------------->|
    |             |              |               |(18) ACK       |
    |             |              |               |<--------------|
    |             |              |               |               |
    |             |              |               |               |
    |             |              |               |               |
    |             |              |               |               |
    |             |              |               |               |
    |             |              |               |               |

    A             B              C               D             Server



   Figure 7: Centralized Signaling, Decentralized Media

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   one from each participant to the server, each with a different Call-
   ID.

   Note that hybrids are easily possible. Certain users can instead be
   mixed (sending audio to the conference server), while others are set
   to send audio to each other.

7.1 Inviting Users to Join

   Inviting users to join works identically to the dial-in conference
   bridge scenario 4.

7.2 Users Joining

   A user joins in the same way described in section 4.

7.3 Scalability

   The scalability of this conferencing model depends on many factors.
   From a media perspective, the conference server never even touches a
   single media stream. However, for N participants, each participant
   needs to be able to receive, decode, and mix N-1 media streams. For
   users accessing the server through dial-in modems, this will severely
   limit the sizes of these conferences. However, the processing burden
   is much less than that of the end system mixing model. This is
   because each end user needs to decode N-1 streams, but only encode 1.
   Decoding is much, much cheaper than encoding, so supporting many
   decodes is not necessarily a problem. This is especially the case
   when silence suppression is in use. In that case, streams are only
   sent by talking users. This means any given user only needs to decode
   (and receive) as many streams at a time as there are users talking.
   THis can vastly improve scalability of the conference.

   There is a signaling burden on the server, however. If there are N
   users in the conference, addition of a new user (the N+1th) requires
   N+3 INVITE transactions, each of which has three messages. Similarly,
   departure of a user requires N BYE transactions, each of which has 2
   messages. For large N, and highly dynamic conferences, this can
   represent a potential burden. However, we believe this bottleneck is
   much farther out than the processing and bandwidth bottlenecks at the
   end users.

   For these reasons, we believe this conference model is ideal in
   corporate enterprises, where bandwidth is more plentiful and PCs are
   generally faster.

7.4 Location of Service Logic




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   Nearly all of the logic for implementing this conferencing service
   lives in the server itself.

   The only requirement from the end users is that they support
   multiple, parallel media streams of the same type, and that they be
   prepared to mix those streams together. They must also support the
   third party control primitives [14], which don't require anything
   beyond baseline SIP, but are not likely supported unless explicit
   actions are taken to do so.

   It is this combination - no need for media processing in the server,
   combined with no need for specialized SIP processing in the end
   systems, that makes this model attractive.

7.5 Discovering Participant Identities

   Conference identities are discovered through RTCP. Each user will
   receive N-1 RTP streams, each of which has its own RTCP channel that
   carries the participant identification.

8 Summary of Models

   Table 1 shows a summary of the differences between the various
   models.


   Table 1: Summary of Models

   Name        signaling  media     inviting   joining      discovering  scale
   End-Mixing  tree       tree      normal     normal       RTCP         small
                                    invite     invite
   Multicast   pairs      m-cast    normal     multicast    RTCP         large
                                    invite     join
   Dial-Up     star       star      refer      normal       RTCP         medium
                                               invite
   Ad-Hoc      star       star      refer      normal       RTCP         medium
                                               invite
   Dial-Out    star       star      refer      normal       RTCP         medium
                                               invite
   Decentral   star       fullmesh  refer +    normal       RTCP         medium
                                    server     invite and
                                    messaging  server msg.



9 Security Considerations

   The use of a server that performs the mixing on behalf of other



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   users, which is the case for all but one of the conference models
   described here, introduces security risks. That entity must be
   trusted by the others to properly mix the media - not omitting a
   stream, for example. As such, it is recommended that participants in
   a conference authenticate the identity of the server. In the dial-in,
   dial-out, and decentralized conferences, this will require
   authentication of responses by participants.

   Mixing also eliminates the privacy possible with end-to-end media
   transport with mixing in the receivers. Such privacy is still
   possible in the large-scale multicast conferences, but requires
   shared keying material for the conference. Doing this for highly
   dynamic groups is still an open research problem.

10 Conclusion

   In this draft, we have shown how to use baseline SIP (assuming
   endpoints that support the mixing and/or third party call control
   feature sets) to construct several multiparty conferencing models.
   These include end system mixing, large-scale multicast conferences,
   dial-in conference servers, dial-out conferences, ad-hoc centralized
   conferences, and centralized signaling, distributed media
   conferences.

11 Acknowledgements

   We would like to thank Mary Barnes for her comments and input.

12 Authors Addresses


   Jonathan Rosenberg
   dynamicsoft
   72 Eagle Rock Avenue
   First Floor
   East Hanover, NJ 07936
   email: jdrosen@dynamicsoft.com

   Henning Schulzrinne
   Columbia University
   M/S 0401
   1214 Amsterdam Ave.
   New York, NY 10027-7003
   email: schulzrinne@cs.columbia.edu



13 Normative References



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14 Informative References

   [1] J. Rosenberg, H. Schulzrinne, et al.  , "SIP: Session initiation
   protocol," Internet Draft, Internet Engineering Task Force, Feb.
   2002.  Work in progress.

   [2] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: a
   transport protocol for real-time applications," RFC 1889, Internet
   Engineering Task Force, Jan. 1996.

   [3] M. Handley and V. Jacobson, "SDP: session description protocol,"
   RFC 2327, Internet Engineering Task Force, Apr. 1998.

   [4] M. Handley, C. Perkins, and E. Whelan, "Session announcement
   protocol," RFC 2974, Internet Engineering Task Force, Oct. 2000.

   [5] D. Thaler, M. Handley, and D. Estrin, "The internet multicast
   address allocation architecture," RFC 2908, Internet Engineering Task
   Force, Sept.  2000.

   [6] W. Fenner, "Internet group management protocol, version 2," RFC
   2236, Internet Engineering Task Force, Nov. 1997.

   [7] D. Waitzman, C. Partridge, and S. E. Deering, "Distance vector
   multicast routing protocol," RFC 1075, Internet Engineering Task
   Force, Nov. 1988.

   [8] J. Rosenberg and H. Schulzrinne, "Timer reconsideration for
   enhanced RTP scalability," in Proceedings of the Conference on
   Computer Communications (IEEE Infocom) , (San Francisco, California),
   March/April 1998.

   [9] J. Rosenberg, P. Mataga, and H. Schulzrinne, "An application
   server component architecture for SIP," Internet Draft, Internet
   Engineering Task Force, Mar. 2001.  Work in progress.

   [10] J. Franks, P. Hallam-Baker, J. Hostetler, S. Lawrence, P. Leach,
   A. Luotonen, and L. Stewart, "HTTP authentication: Basic and digest
   access authentication," RFC 2617, Internet Engineering Task Force,
   June 1999.

   [11] R. Sparks, "The SIP refer method," Internet Draft, Internet
   Engineering Task Force, June 2002.  Work in progress.

   [12] E. Guttman, C. Perkins, J. Veizades, and M. Day, "Service
   location protocol, version 2," RFC 2608, Internet Engineering Task
   Force, June 1999.




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   [13] J. Rosenberg, "Session initiation protocol (SIP) extensions for
   presence," Internet Draft, Internet Engineering Task Force, May 2002.
   Work in progress.

   [14] J. Rosenberg, J. Peterson, H. Schulzrinne, and G. Camarillo,
   "Third party call control in SIP," Internet Draft, Internet
   Engineering Task Force, Nov.  2001.  Work in progress.


   Full Copyright Statement

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