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Versions: (draft-rosenberg-sip-3pcc) 00 01 02 03 04 05 06 RFC 3725

SIPPING                                                     J. Rosenberg
Internet-Draft                                               dynamicsoft
Expires: December 29, 2003                                   J. Peterson
                                                                 Neustar
                                                          H. Schulzrinne
                                                     Columbia University
                                                            G. Camarillo
                                            Ericsson Advanced Signalling
                                                            Research Lab
                                                           June 30, 2003


   Best Current Practices for Third Party Call Control in the Session
                          Initiation Protocol
                       draft-ietf-sipping-3pcc-04

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
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   http://www.ietf.org/shadow.html.

   This Internet-Draft will expire on December 29, 2003.

Copyright Notice

   Copyright (C) The Internet Society (2003). All Rights Reserved.

Abstract

   Third party call control refers to the ability of one entity to
   create a call in which communications is actually between other
   parties. Third party call control is possible using the mechanisms
   specified within the Session Initiation Protocol (SIP). However,
   there are several possible approaches, each with different benefits



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   and drawbacks. This document discusses best current practices for the
   usage of SIP for third party call control.

Table of Contents

   1.   Introduction . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.   Terminology  . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.   Definitions  . . . . . . . . . . . . . . . . . . . . . . . .   5
   4.   3pcc Call Establishment  . . . . . . . . . . . . . . . . . .   6
   4.1  Flow I . . . . . . . . . . . . . . . . . . . . . . . . . . .   6
   4.2  Flow II  . . . . . . . . . . . . . . . . . . . . . . . . . .   7
   4.3  Flow III . . . . . . . . . . . . . . . . . . . . . . . . . .   9
   4.4  Flow IV  . . . . . . . . . . . . . . . . . . . . . . . . . .  11
   5.   Recommendations  . . . . . . . . . . . . . . . . . . . . . .  13
   6.   Error Handling . . . . . . . . . . . . . . . . . . . . . . .  14
   7.   Continued Processing . . . . . . . . . . . . . . . . . . . .  16
   8.   3pcc and Early Media . . . . . . . . . . . . . . . . . . . .  18
   9.   Third Party Call Control and SDP Preconditions . . . . . . .  21
   9.1  Controller Initiates . . . . . . . . . . . . . . . . . . . .  21
   9.2  Party A Initiates  . . . . . . . . . . . . . . . . . . . . .  23
   10.  Example Call Flows . . . . . . . . . . . . . . . . . . . . .  27
   10.1 Click to Dial  . . . . . . . . . . . . . . . . . . . . . . .  27
   10.2 Mid-Call Announcement Capability . . . . . . . . . . . . . .  28
   11.  Implementation Recommendations . . . . . . . . . . . . . . .  31
   12.  Security Considerations  . . . . . . . . . . . . . . . . . .  32
   12.1 Identity . . . . . . . . . . . . . . . . . . . . . . . . . .  32
   12.2 End-to-End Encryption and Integrity  . . . . . . . . . . . .  32
   13.  IANA Considerations  . . . . . . . . . . . . . . . . . . . .  34
   14.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . .  35
        Normative References . . . . . . . . . . . . . . . . . . . .  36
        Informative References . . . . . . . . . . . . . . . . . . .  37
        Authors' Addresses . . . . . . . . . . . . . . . . . . . . .  37
        Intellectual Property and Copyright Statements . . . . . . .  39


















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1. Introduction

   In the traditional telephony context, third party call control allows
   one entity (which we call the controller) to set up and manage a
   communications relationship between two or more other parties.  Third
   party call control (referred to as 3pcc) is often used for operator
   services (where an operator creates a call that connects two
   participants together), and conferencing.

   Similarly, many SIP services are possible through third party call
   control. These include the traditional ones on the PSTN, but also new
   ones such as click-to-dial. Click-to-dial allows a user to click on a
   web page when they wish to speak to a customer service
   representative. The web server then creates a call between the user
   and a customer service representative. The call can be between two
   phones, a phone and an IP host, or two IP hosts.

   Third party call control is possible using only the mechanisms
   specified within RFC 3261 [1]. Indeed, many different call flows are
   possible, each of which will work with SIP compliant user agents.
   However, there are benefits and drawbacks to each of these flows. The
   usage of third party call control also becomes more complex when
   aspects of the call utilize SIP extensions or optional features of
   SIP. In particular, the usage of RFC 3312 [2] (used for coupling of
   signaling to resource reservation) with third party call control is
   non-trivial. Similarly, the usage of early media (where session data
   is exchanged before the call is accepted) with third party call
   control is not trivial.

   This document serves as a best current practice for implementing
   third party call control without usage of any extensions specifically
   designed for that purpose. Section 4 presents the known call flows
   that can be used to achieve third party call control, and provides
   guidelines on their usage. Section 9 discusses the interactions of
   RFC 3312 [2] with third party call control. Section 8 discusses the
   interactions of early media with third party call control. Section 10
   provides example applications that make usage of the flows
   recommended here.













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2. 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 [3] and
   indicate requirement levels for compliant implementations.













































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3. Definitions

   The following terms are used throughout this document:

   3pcc: Third Party Call Control, which refers to the general ability
      to manipulate calls between other parties.

   Controller: A controller is a SIP User Agent that wishes to create a
      session between two other user agents.










































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4. 3pcc Call Establishment

   The primary primitive operation of third party call control is the
   establishment of a session between participants A and B.
   Establishment of this session is orchestrated by a third party,
   referred to as the controller.

   This section documents three call flows that the controller can
   utilize in order to provide this primitive operation.

4.1 Flow I


             A              Controller               B
             |(1) INVITE no SDP  |                   |
             |<------------------|                   |
             |(2) 200 offer1     |                   |
             |------------------>|                   |
             |                   |(3) INVITE offer1  |
             |                   |------------------>|
             |                   |(4) 200 OK answer1 |
             |                   |<------------------|
             |                   |(5) ACK            |
             |                   |------------------>|
             |(6) ACK answer1    |                   |
             |<------------------|                   |
             |(7) RTP            |                   |
             |.......................................|

                                Figure 1

   The call flow for Flow I is shown in Figure 1. The controller first
   sends an INVITE A (1). This INVITE has no session description. A's
   phone rings, and A answers. This results in a 200 OK (2) that
   contains an offer [4]. The controller needs to send its answer in the
   ACK, as mandated by [1]. To obtain the answer, it sends the offer it
   got from A (offer1) in an INVITE to B (3). B's phone rings. When B
   answers, the 200 OK (4) contains the answer to this offer, answer1.
   The controller sends an ACK to B (5), and then passes answer1 to A in
   an ACK sent to it (6). Because the offer was generated by A, and the
   answer generated by B, the actual media session is between A and B.
   Therefore, media flows between them (7).

   This flow is simple, requires no manipulation of the SDP by the
   controller, and works for any media types supported by both
   endpoints. However, it has a serious timeout problem. User B may not
   answer the call immediately. The result is that the controller cannot
   send the ACK to A right away. This causes A to retransmit the 200 OK



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   response periodically. As specified in RFC 3261 Section 13.3.1.4, the
   200 OK will be retransmitted for 64*T1 seconds. If an ACK does not
   arrive by then, the call is considered to have failed. This limits
   the applicability of this flow to scenarios where the controller
   knows that B will answer the INVITE immediately.

4.2 Flow II


             A              Controller               B
             |(1) INVITE bh sdp1 |                   |
             |<------------------|                   |
             |(2) 200 sdp2       |                   |
             |------------------>|                   |
             |                   |(3) INVITE sdp2    |
             |                   |------------------>|
             |(4) ACK            |                   |
             |<------------------|                   |
             |                   |(5) 200 OK sdp3    |
             |                   |<------------------|
             |                   |(6) ACK            |
             |                   |------------------>|
             |(7) INVITE sdp3    |                   |
             |<------------------|                   |
             |(8) 200 OK sdp2    |                   |
             |------------------>|                   |
             |(9) ACK            |                   |
             |<------------------|                   |
             |(10) RTP           |                   |
             |.......................................|

                                Figure 2

   An alternative flow, Flow II, is shown in Figure 2. The controller
   first sends an INVITE to user A (1). This is a standard INVITE,
   containing an offer (sdp1) with a single audio media line, one codec,
   a random port number (but not zero), and a connection address that is
   somewhere within the .invalid DNS top level domain (TLD) [8] (for
   example, rtp.invalid). The use of a connection address in the invalid
   domain creates an initial media stream that is ``black holed'', since
   no media (or RTCP packets [9] will flow from A. The INVITE causes A's
   phone to ring.

   When A answers (2), the 200 OK contains an answer, sdp2, with a valid
   address in the connection line.

      A user agent should respond to an offer containing a .invalid
      domain just as it would any other offer. The answer contains its



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      own address in the connection line. However, due to the presence
      of a purposefully invalid domain name, media would not be sent to
      the offerer.

   The controller sends an ACK (4). It then generates a second INVITE
   (3). This INVITE is addressed to user B, and it contains sdp2 as the
   offer to B. Note that the role of sdp2 has changed. In the 200 OK
   (message 2), it was an answer, but in the INVITE, it is an offer.
   Fortunatly, all valid answers are valid initial offers. This INVITE
   causes B's phone to ring. When it answers, it generates a 200 OK (5)
   with an answer, sdp3. The controller then generates an ACK (6). Next,
   it sends a re-INVITE to A (7) containing sdp3 as the offer. Once
   again, there has been a reversal of roles. sdp3 was an answer, and
   now it is an offer. Fortunately, an answer to an answer recast as an
   offer is, in turn, a valid offer. This re-INVITE generates a 200 OK
   (8) with sdp2, assuming that A doesn't decide to change any aspects
   of the session as a result of this re-INVITE. This 200 OK is ACKed
   (9), and then media can flow from A to B. Media from B to A could
   already start flowing once message 5 was sent.

   This flow has the advtange that all final responses are immediately
   ACKed. It therefore does not suffer from the timeout and message
   inefficiency problems of flow 1. However, it too has troubles. First
   off, it requires that the controller know the media types to be used
   for the call (since it must generate a ``blackhole'' SDP, which
   requires media lines). Secondly, the first INVITE to A (1) contains
   media with a rtp.invalid connection address to indicate that neither
   media nor RTCP packets should be sent. This mechanism for indicating
   that neither media or RTCP should be sent is different the one
   documented in RFC 3264 [4], which uses 0.0.0.0 for this purpose.
   However, according to RFC 3330 [11], 0.0.0.0 is a special case
   address which refers to hosts on this network. This is not the same
   as the desired semantic, which is much closer to the meaning of the
   .invalid TLD. However, it is unknown how off-the-shelf user agents
   will react to an SDP that contains a .invalid TLD in a connection
   line.

   However, the most serious flaw in this flow is the assumption that
   the 200 OK to the re-INVITE (message 8) contains the same SDP as in
   message 2. This may not be the case. If it is not, the controller
   needs to re-INVITE B with that SDP (say, sdp4), which may result in
   getting a different SDP, sdp5 , in the 200 OK from B. Then, the
   controller needs to re-INVITE A again, and so on. The result is an
   infinite loop of re-INVITEs. It is possible to break this cycle by
   having very smart UAs which can return the same SDP whenever
   possible, or really smart controllers that can analyze the SDP to
   determine if a re-INVITE is really needed. However, we wish to keep
   this mechanism simple, and avoid SDP awareness in the controller. As



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   a result, this flow is not really workable. It is therefore NOT
   RECOMMENDED.

4.3 Flow III


             A                 Controller                  B
             |(1) INVITE no SDP     |                      |
             |<---------------------|                      |
             |(2) 200 offer1        |                      |
             |--------------------->|                      |
             |(3) ACK answer1 (bh)  |                      |
             |<---------------------|                      |
             |                      |(4) INVITE no SDP     |
             |                      |--------------------->|
             |                      |(5) 200 OK offer2     |
             |                      |<---------------------|
             |(6) INVITE offer2'    |                      |
             |<---------------------|                      |
             |(7) 200 answer2'      |                      |
             |--------------------->|                      |
             |                      |(8) ACK answer2       |
             |                      |--------------------->|
             |(9) ACK               |                      |
             |<---------------------|                      |
             |(10) RTP              |                      |
             |.............................................|

                                Figure 3

   A third flow, Flow III, is shown in Figure 3

   First, the controller sends an INVITE (1) to user A without any SDP
   (which is good, since it means that the controller doesn't need to
   assume anything about the media composition of the session). A's
   phone rings. When A answers, a 200 OK is generated (2) containing its
   offer, offer1. The controller generates an immediate ACK containing
   an answer (3). This answer is a ``black hole'' SDP, with its
   connection address set to any domain in the .invalid TLD, such as
   media.invalid.

   The controller then sends an INVITE to B without SDP (4). This causes
   B's phone to ring. When they answer, a 200 OK is sent, containing
   their offer, offer2 (5). This SDP is used to create a re-INVITE back
   to A (6). That re-INVITE is based on offer2, but may need to be
   reorganized to match up media lines, or to trim media lines. For
   example, if offer1 contained an audio and a video line, in that
   order, but offer2 contained just an audio line, the controller would



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   need to add a video line to the offer (setting its port to zero) to
   create offer2'. Since this is a re-INVITE, it should complete quickly
   in the general case. Thats good, since user B is retransmitting their
   200 OK, waiting for an ACK. The SDP in the 200 OK (7) from A,
   answer2', may also need to be reorganized or trimmed before sending
   it an the ACK to B (8) as answer2. Finally, an ACK is sent to A (9),
   and then media can flow.

   This flow has many benefits. First, it will usually operate without
   any spurious retransmissions or timeouts (although this may still
   happen if a re-INVITE is not responded to quickly). Secondly, it does
   not require the controller to guess the media that will be used by
   the participants.

   There are some drawbacks. The controller does need to perform SDP
   manipulations. Specifically, it must take some SDP, and generate
   another SDP which has the same media composition, but has connection
   addresses within the .invalid TLD. This is needed for message 3.
   Secondly, it may need to reorder and trim on SDP X, so that its media
   lines match up with those in some other SDP, Y. Thirdly, the offer
   from B (offer2) may have no codecs or media streams in common with
   the offer from A (offer 1). The controller will need to detect this
   condition, and terminate the call. Fourth, it requires usage of the
   .invalid TLD within SDP, which may not be supported. Finally, the
   flow is far more complicated than the simple and elegant Flow I
   (Figure 1).

























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4.4 Flow IV


             A                 Controller                  B
             |(1) INVITE offer1     |                      |
             |no media              |                      |
             |<---------------------|                      |
             |(2) 200 answer1       |                      |
             |no media              |                      |
             |--------------------->|                      |
             |(3) ACK               |                      |
             |<---------------------|                      |
             |                      |(4) INVITE no SDP     |
             |                      |--------------------->|
             |                      |(5) 200 OK offer2     |
             |                      |<---------------------|
             |(6) INVITE offer2'    |                      |
             |<---------------------|                      |
             |(7) 200 answer2'      |                      |
             |--------------------->|                      |
             |                      |(8) ACK answer2       |
             |                      |--------------------->|
             |(9) ACK               |                      |
             |<---------------------|                      |
             |(10) RTP              |                      |
             |.............................................|

                                Figure 4

   Flow IV shows a variation on Flow III that reduces its complexity.
   The actual message flow is identical, but the SDP placement and
   construction differs. The initial INVITE (1) contains SDP with no
   media at all, meaning that there are no m lines. This is valid, and
   implies that the media makeup of the session will be established
   later through a re-INVITE [4]. Once the INVITE is received, user A is
   alerted. When they answer the call, the 200 OK (2) has an answer with
   no media either. This is acknowledged by the controller (3). The flow
   from this point onwards is identical to Flow III. However, the
   manipulations required to convert offer2 to offer2', and answer2' to
   answer2, are much simpler. Indeed, no media manipulations are needed
   at all. The only change that is needed is to modify the origin lines,
   so that the origin line in offer2' is valid based on the value in
   offer1 (validify requires that the version increments by one, and
   that the other parameters remain unchanged).

   There are some limitations associated with this flow. First, user A
   will be alerted without any media having been established yet. This
   means that user A will not be able to reject or accept the call based



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   on its media composition. Secondly, both A and B will end up
   answering the call (i.e., generating a 200 OK) before it is known
   whether their is compatible media. If there is no media in common,
   the call can be terminated later with a BYE. However, the users will
   have already been alerted, resulting in user annoyance and possibly
   resulting in billing events.













































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5. Recommendations

   Flow I (Figure 1) represents the simplest and the most efficient
   flow. This flow SHOULD be used by a controller if it knows with
   certainty that user B is actually an automata that will answer the
   call immediately. This is the case for devices such as media servers,
   conferencing servers, and messaging servers, for example. Since we
   expect a great deal of third party call control to be to automata,
   special caseing this scenario is reasonable.

   For calls to unknown entities, or to entities known to represent
   people, it is RECOMMENDED that Flow IV (Figure 4) be used for third
   party call control. Flow III MAY be used instead, but it provides no
   additional benefits over Flow IV. However, Flow II SHOULD NOT be
   used, because of the potential for infinite ping-ponging of
   re-INVITEs.

   Several of these flows use a ``black hole'' connection address, which
   is any domain in the .invalid domain. A UA MUST be prepared to
   receive offers or answers with connection lines containing addresses
   within this domain. A UA receiving such a domain proceeds with normal
   RFC 3264 offer/answer processing. However, it will not send media to
   the address. A UA MAY bypass DNS lookup of the connection line when
   it sees it's within the .invalid domain. Of course, a host that
   performs a lookup with receive a response indicating that the host
   name is invalid.

























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6. Error Handling

   There are numerous error cases which merit discussion.

   With all of the call flows in Section 4, one call is established to
   A, and then the controller attempts to establish a call to B.
   However, this call attempt may fail, for any number of reasons. User
   B might be busy (resulting in a 486 response to the INVITE), there
   may not be any media in common, the request may time out, and so on.
   If the call attempt to B should fail, it is RECOMMENDED that the
   controller send a BYE to A. This BYE SHOULD include a Reason header
   [5] which carries the status code from the error response. This will
   inform A of the precise reason for the failure. The information is
   important from a user interface perspective. For example, if A was
   calling from a black phone, and B generated a 486, the BYE will
   contain a Reason code of 486, and this could be used to generate a
   local busy signal so that A knows that B is busy.


             A                 Controller                  B
             |(1) INVITE offer1     |                      |
             |no media              |                      |
             |<---------------------|                      |
             |(2) 200 answer1       |                      |
             |no media              |                      |
             |--------------------->|                      |
             |(3) ACK               |                      |
             |<---------------------|                      |
             |                      |(4) INVITE no SDP     |
             |                      |--------------------->|
             |                      |(5) 180               |
             |                      |<---------------------|
             |(6) INVITE offer2     |                      |
             |--------------------->|                      |
             |(7) 491               |                      |
             |<---------------------|                      |
             |(8) ACK               |                      |
             |--------------------->|                      |

                                Figure 5

   Another error condition worth discussion is shown in Figure 5. After
   the controller establishes the dialog with A (messages 1-3) it
   attempts to contact B (message 4). Contacting B may take some time.
   During that interval, A could possibly attempt a re-INVITE, providing
   an updated offer. However, the controller cannot pass this offer on
   to B, since it has an INVITE transaction pending with it. As a
   result, the controller needs to reject the request. It is RECOMMENDED



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   that a 491 response be used. The situation here is similar to the
   glare condition described in [1], and thus the same error handling is
   sensible. However, A is likely to retry its request (as a result of
   the 491), and this may occur before the exchange with B is completed.
   In that case, the controller would respond with another 491.














































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7. Continued Processing

   Once the calls are established, both participants believe they are in
   a single point-to-point call. However, they are exchanging media
   directly with each other, rather than with the controller. The
   controller is involved in two dialogs, yet sees no media.

   Since the controller is still a central point for signaling, it now
   has complete control over the call. If it receives a BYE from one of
   the participants, it can create a new BYE and hang up with the other
   participant. This is shown in Figure 6.


             A              Controller               B
             |(1) BYE            |                   |
             |------------------>|                   |
             |(2) 200 OK         |                   |
             |<------------------|                   |
             |                   |(3) BYE            |
             |                   |------------------>|
             |                   |(4) 200 OK         |
             |                   |<------------------|

                                Figure 6

   Similarly, if it receives a re-INVITE from one of the participants,
   it can forward it to the other participant. Depending on which flow
   was used, this may require some manipulation on the SDP before
   passing it on.

   However, the controller need not "proxy" the SIP messages received
   from one of the parties. Since it is a B2BUA, it can invoke any
   signaling mechanism on each dialog, as it sees fit. For example, if
   the controller receives a BYE from A, it can generate a new INVITE to
   a third party, C, and connect B to that participant instead. A call
   flow for this is shown in Figure 7, assuming the case where C
   represents an end user, not an automata. Note that it is just Flow
   IV.













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             A           Controller            B                C
             |(1) BYE         |                |                |
             |--------------->|                |                |
             |(2) 200 OK      |                |                |
             |<---------------|                |                |
             |                |(3) INV no media|                |
             |                |-------------------------------->|
             |                |(4) 200 no media|                |
             |                |<--------------------------------|
             |                |(5) ACK         |                |
             |                |-------------------------------->|
             |                |(6) INV no SDP  |                |
             |                |--------------->|                |
             |                |(7) 200 offer3  |                |
             |                |<---------------|                |
             |                |(8) INV offer3' |                |
             |                |-------------------------------->|
             |                |(9) 200 answer3'|                |
             |                |<--------------------------------|
             |                |(10) ACK        |                |
             |                |-------------------------------->|
             |                |(11) ACK answer3|                |
             |                |--------------->|                |
             |                |                |(12) RTP        |
             |                |                |................|

                                Figure 7

   From here, new parties can be added, removed, transferred, and so on,
   as the controller sees fit.

   It is important to point out that the call need not have been
   established by the controller in order for the processing of this
   section to be used. Rather, the controller could have acted as a
   B2BUA during a call established by A towards B (or vice a versa).
















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8. 3pcc and Early Media

   Early media represents the condition where the session is established
   (as a result of the completion of an offer/answer exchange), yet the
   call itself has not been accepted. This is usually used to convey
   tones or announcements regarding progress of the call. Handling of
   early media in a third party call is straightforward.


             A                 Controller                  B
             |                      |                      |
             |(1) INVITE offer1     |                      |
             |no media              |                      |
             |<---------------------|                      |
             |                      |                      |
             |<ring>                |                      |
             |                      |                      |
             |<answer>              |                      |
             |                      |                      |
             |(2) 200 answer1       |                      |
             |no media              |                      |
             |--------------------->|                      |
             |(3) ACK               |                      |
             |<---------------------|                      |
             |                      |(4) INVITE no SDP     |
             |                      |--------------------->|
             |                      |                      |<ring>
             |                      |(5) 183 offer2        |
             |                      |<---------------------|
             |(6) INVITE offer2'    |                      |
             |<---------------------|                      |
             |(7) 200 answer2'      |                      |
             |--------------------->|                      |
             |(8) ACK               |                      |
             |<---------------------|                      |
             |                      |(9) PRACK answer2     |
             |                      |--------------------->|
             |                      |(10) 200 PRACK        |
             |                      |<---------------------|
             |(11) RTP              |                      |
             |.............................................|
             |                      |                      |<answer>
             |                      |(12) 200 OK           |
             |                      |<---------------------|
             |                      |(13) ACK              |
             |                      |--------------------->|





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                                Figure 8

   Figure 8 shows the case where user B generates early media before
   answering the call. The flow is almost identical to Flow IV from
   Figure 4. The only difference is that user B generates a reliable
   provisional response (5) [6] instead of a final response, and answer2
   is carried in a PRACK (8) instead of an ACK. When party B finally
   does accept the call (11), there is no change in the session state,
   and therefore, no signaling needs to be done with user A. The
   controller simply ACKs the 200 OK (12) to confirm the dialog.


             A                 Controller                  B
             |                      |                      |
             |(1) INVITE offer1     |                      |
             |no media              |                      |
             |<---------------------|                      |
             |                      |                      |
             |ring                  |                      |
             |                      |                      |
             |(2) 183 answer1       |                      |
             |no media              |                      |
             |--------------------->|                      |
             |(3) PRACK             |                      |
             |<---------------------|                      |
             |(4) 200 PRACK         |                      |
             |--------------------->|                      |
             |                      |(5) INVITE no SDP     |
             |                      |--------------------->|
             |                      |                      |ring
             |                      |                      |
             |                      |                      |answer
             |                      |                      |
             |                      |(6) 200 OK offer2     |
             |                      |<---------------------|
             |(7) UPDATE offer2'    |                      |
             |<---------------------|                      |
             |answer                |                      |
             |                      |                      |
             |(8) 200 answer2'      |                      |
             |--------------------->|                      |
             |                      |(9) ACK answer2       |
             |                      |--------------------->|
             |(10) RTP              |                      |
             |.............................................|
             |(11) 200 OK           |                      |
             |--------------------->|                      |
             |(12) ACK              |                      |



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             |<---------------------|                      |


                                Figure 9

   The case where user A generates early media is more complicated, and
   is shown in Figure 9. The flow is based on Flow IV. The controller
   sends an INVITE to user A (1), with an offer containing no media
   streams. User A generates a reliable provisional response (2)
   containing an answer with no media streams. The controller PRACKs
   this provisional response (3). Now, the controller sends an INVITE
   without SDP to user B (5). User B's phone rings, and they answer,
   resulting in a 200 OK (6) with an offer, offer2. The controller now
   needs to update the session parameters with user A. However, since
   the call has not been answered, it cannot use a re-INVITE. Rather, it
   uses a SIP UPDATE request (7) [7], passing the offer (after modifying
   it to get the origin field correct). User A generates its answer in
   the 200 OK to the UPDATE (8). This answer is passed to user B in the
   ACK (9). When user A finally answers (11), there is no change in
   session state, so the controller simply ACKs the 200 OK (12).

   Note that it is likely that there will be clipping of media in this
   call flow. User A is likely a PSTN gateway, and has generated a
   provisional response because of early media from the PSTN side. The
   PSTN will deliver this media even though the gateway does not have
   anywhere to send it, since the initial offer from the controller had
   no media streams. When user B answers, media can begin to flow.
   However, any media sent to the gateway from the PSTN up to that point
   will be lost.






















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9. Third Party Call Control and SDP Preconditions

   A SIP extension has been specified that allows for the coupling of
   signaling and resource reservation [2]. This specification relies on
   exchanges of session descriptions before completion of the call
   setup. These flows are initiated when certain SDP parameters are
   passed in the initial INVITE. As a result, the interaction of this
   mechanism with third party call control is not obvious, and worth
   detailing.

9.1 Controller Initiates

   In one usage scenario, the controller wishes to make use of
   preconditions in order to avoid the call failure scenarios documented
   in Section 4.4. Specifically, the controller can use preconditions in
   order to guarantee that neither party is alerted unless there is a
   common set of media and codecs. It can also provide both parties with
   information on the media composition of the call before they decide
   to accept it.
































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           User             Controller       Customer Service
             |                   |                   |
             |(1) INVITE no SDP  |                   |
             |require precon     |                   |
             |<------------------|                   |
             |(2) 183 offer1     |                   |
             |optional precon    |                   |
             |------------------>|                   |
             |                   |                   |
             |                   |(3) INVITE offer1  |
             |                   |------------------>|
             |                   |                   |<ring>
             |                   |                   |
             |                   |                   |<answer>
             |                   |(4) 200 OK answer1 |
             |                   |no precon          |
             |                   |<------------------|
             |                   |(5) ACK            |
             |                   |------------------>|
             |(6) PRACK answer1  |                   |
             |<------------------|                   |
             |<ring>             |                   |
             |                   |                   |
             |(7) 200 PRACK      |                   |
             |------------------>|                   |
             |<answer>           |                   |
             |                   |                   |
             |(8) 200 INVITE     |                   |
             |------------------>|                   |
             |(9) ACK            |                   |
             |<------------------|                   |


                               Figure 10

   The flow for this scenario is shown in Figure 10. In this example, we
   assume that user B is an automata or agent of some sort which will
   answer the call immediately. Therefore, the flow is based on Flow I.
   The controller sends an INVITE to user A containing no SDP, but with
   a Require header indicating that preconditions are required. This
   specific scenario (an INVITE without an offer, but with a Require
   header indicating preconditions) is not described in [2]. It is
   RECOMMENDED that the UAS respond with an offer in a 1xx including the
   media streams it wishes to use for the call, and for each, list all
   preconditions it supports as optional. Of course, the user is not
   alerted at this time. The controller takes this offer and passes it
   to user B (3). User B does not support preconditions, or does, but is
   not interested in them. Therefore, when it answers the call, the 200



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   OK contains an answer without any preconditions listed (4). This
   answer is passed to user A in the PRACK (6). At this point, user A
   knows that there are no preconditions actually in use for the call,
   and therefore, it can alert the user. When the call is answered, user
   A sends a 200 OK to the controller (8) and the call is complete.

   In the event that the offer generated by user A was not acceptable to
   user B (because of non-overlapping codecs or media, for example),
   user B would immediately reject the INVITE (message 3). The
   controller would then CANCEL the request to user A. In this
   situation, neither user A nor user B would have been alerted,
   achieving the desired effect. It is interesting to note that this
   property is achieved using preconditions even though it doesn't
   matter what specific types of preconditions are supported by user A.

   It is also entirely possible that user B does actually desire
   preconditions. In that case, it might generate a 1xx of its own with
   an answer containing preconditions. That answer would still be passed
   to user A, and both parties would proceed with whatever measures are
   necessary to meet the preconditions. Neither user would be alerted
   until the preconditions were met.

9.2 Party A Initiates

   In Section 9.1, the controller requested the use of preconditions to
   achieve a specific goal. It is also possible that the controller
   doesn't care (or perhaps doesn't even know) about preconditions, but
   one of the participants in the call does care. A call flow for this
   case is shown in Figure 11.


             A                 Controller                  B
             |(1) INVITE offer1     |                      |
             |no media              |                      |
             |<---------------------|                      |
             |(2) 183 answer1       |                      |
             |no media              |                      |
             |--------------------->|                      |
             |(3) PRACK             |                      |
             |<---------------------|                      |
             |(4) 200 OK            |                      |
             |--------------------->|                      |
             |                      |(5) INVITE no SDP     |
             |                      |--------------------->|
             |                      |(6) 183 offer2        |
             |                      |des=sendrecv          |
             |                      |conf=recv             |
             |                      |cur=none              |



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             |                      |<---------------------|
             |(7) UPDATE offer2'    |                      |
             |des=sendrecv          |                      |
             |conf=recv             |                      |
             |cur=none              |                      |
             |<---------------------|                      |
             |(8) 200 UPDATE        |                      |
             |answer2'              |                      |
             |des=sendrecv          |                      |
             |conf=recv             |                      |
             |cur=none              |                      |
             |--------------------->|                      |
             |                      |(9) PRACK answer2     |
             |                      |des=sendrecv          |
             |                      |conf=recv             |
             |                      |cur=none              |
             |                      |--------------------->|
             |                      |(10) 200 PRACK        |
             |                      |<---------------------|
             |(11) reservation      |                      |
             |-------------------------------------------->|
             |(12) reservation      |                      |
             |<--------------------------------------------|
             |(13) UPDATE offer3    |                      |
             |des=sendrecv          |                      |
             |conf=recv             |                      |
             |cur=recv              |                      |
             |--------------------->|                      |
             |                      |(14) UPDATE offer3'   |
             |                      |des=sendrecv          |
             |                      |conf=recv             |
             |                      |cur=recv              |
             |                      |--------------------->|
             |                      |(15) 200 UPDATE       |
             |                      |answer3'              |
             |                      |des=sendrecv          |
             |                      |conf=recv             |
             |                      |cur=send              |
             |                      |<---------------------|
             |(16) 200 UPDATE       |                      |
             |answer3               |                      |
             |des=sendrecv          |                      |
             |conf=recv             |                      |
             |cur=send              |                      |
             |<---------------------|                      |
             |                      |                      |<ring>
             |                      |(17) UPDATE offer4    |
             |                      |des=sendrecv          |



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             |                      |conf=recv             |
             |                      |cur=sendrecv          |
             |                      |<---------------------|
             |(18) UPDATE offer4'   |                      |
             |des=sendrecv          |                      |
             |conf=recv             |                      |
             |cur=sendrecv          |                      |
             |<---------------------|                      |
             |<ring>                |                      |
             |(19) 200 UPDATE       |                      |
             |answer4'              |                      |
             |des=sendrecv          |                      |
             |conf=recv             |                      |
             |cur=sendrecv          |                      |
             |--------------------->|                      |
             |                      |(20) 200 UPDATE       |
             |                      |answer4               |
             |                      |des=sendrecv          |
             |                      |conf=recv             |
             |                      |cur=sendrecv          |
             |                      |--------------------->|
             |(21) 180 INVITE       |                      |
             |--------------------->|                      |
             |                      |(22) 180 INVITE       |
             |                      |<---------------------|
             |<answer>              |                      |
             |(23) 200 INVITE       |                      |
             |--------------------->|                      |
             |(24) ACK              |                      |
             |<---------------------|                      |
             |                      |                      |<answer>
             |                      |(25) 200 INVITE       |
             |                      |<---------------------|
             |                      |(26) ACK              |
             |                      |--------------------->|

                               Figure 11

   The controller follows Flow IV; it has no specific requirements for
   support of the preconditions specification [2]. Therefore, it sends
   an INVITE (1) with SDP that contains no media lines. User A is
   interested in supporting preconditions, and does not want to ring its
   phone until resources are reserved. Since there are no media streams
   in the INVITE, it can't reserve resources for media streams, and
   therefore it can't ring the phone until they are conveyed in a
   subsequent offer and then reserved. Therefore, it generates a 183
   with the answer, and doesn't alert the user (2). The controller
   PRACKs this (3) and A responds to the PRACK (4).



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   At this point, the controller attempts to bring B into the call. It
   sends B an INVITE without SDP (5). B is interested in having
   preconditions for this call. Therefore, it generates its offer in a
   183 that contains the appropriate SDP attributes (6). The controller
   passes this offer to A in an UPDATE request (7). The controller uses
   UPDATE because the call has not been answered yet, and therefore, it
   cannot use a re-INVITE. User A sees that its peer is capable of
   supporting preconditions. Since it desires preconditions for the
   call, it generates an answer in the 200 OK (8) to the UPDATE. This
   answer, in turn, is passed to B in the PRACK for the provisional
   response (9). Now, both sides perform resource reservation. User A
   succeeds first, and passes an updated session description in an
   UPDATE request (13). The controller simply passes this to A (after
   the manipulation of the origin field, as required in Flow IV) in an
   UPDATE (14), and the answer (15) is passed back to A (16). The same
   flow happens, but from B to A, when B's reservation succeeds (17-20).
   Since the preconditions have been met, both sides ring (21 and 22),
   and then both answer (23 and 25), completing the call.

   What is important about this flow is that the controller doesn't know
   anything about preconditions. It merely passes the SDP back and forth
   as needed. The trick is the usage of UPDATE and PRACK to pass the SDP
   when needed. That determination is made entirely based on the offer/
   answer rules described in [6] and [7], and is independent of
   preconditions.


























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10. Example Call Flows

10.1 Click to Dial

   The first application of this capability we discuss is click to dial.
   In this service, a user is browsing the web page of an e-commerce
   site, and would like to speak to a customer service representative.
   They click on a link, and a call is placed to a customer service
   representative. When the representative picks up, the phone on the
   user's desk rings. When they pick up, the customer service
   representative is there, ready to talk to the user.


     Customer Service       Controller          Users Phone        Users Browser
             |                   |(1) HTTP POST      |                   |
             |                   |<--------------------------------------|
             |                   |(2) HTTP 200 OK    |                   |
             |                   |-------------------------------------->|
             |(3) INVITE offer1  |                   |                   |
             |no media           |                   |                   |
             |<------------------|                   |                   |
             |(4) 200 answer1    |                   |                   |
             |no media           |                   |                   |
             |------------------>|                   |                   |
             |(5) ACK            |                   |                   |
             |<------------------|                   |                   |
             |                   |(6) INVITE no SDP  |                   |
             |                   |------------------>|                   |
             |                   |(7) 200 OK offer2  |                   |
             |                   |<------------------|                   |
             |(8) INVITE offer2' |                   |                   |
             |<------------------|                   |                   |
             |(9) 200 answer2'   |                   |                   |
             |------------------>|                   |                   |
             |                   |(10) ACK answer2   |                   |
             |                   |------------------>|                   |
             |(11) ACK           |                   |                   |
             |<------------------|                   |                   |
             |(12) RTP           |                   |                   |
             |.......................................|                   |

                               Figure 12

   The call flow for this service is given in Figure 12. It is identical
   to that of Figure 4, with the exception that the service is triggered
   through an http GET request when the user clicks on the link.

   We note that this service can be provided through other mechanisms,



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   namely PINT [10]. However, there are numerous differences between the
   way in which the service is provided by pint, and the way in which it
   is provided here:

   o  The pint solution enables calls only between two PSTN endpoints.
      The solution described here allows calls between PSTN phones
      (through SIP enabled gateways) and native IP phones.

   o  When used for calls between two PSTN phones, the solution here may
      result in a portion of the call being routed over the Internet. In
      pint, the call is always routed only over the PSTN. This may
      result in better quality calls with the pint solution, depending
      on the codec in use and QoS capabilities of the network routing
      the Internet portion of the call.

   o  The PINT solution requires extensions to SIP (PINT is an extension
      to SIP), whereas the solution described here is done with baseline
      SIP.

   o  The PINT solution allows the controller (acting as a PINT client)
      to "step out" once the call is established. The solution described
      here requires the controller to maintain call state for the entire
      duration of the call.


10.2 Mid-Call Announcement Capability

   The third party call control mechanism described here can also be
   used to enable mid-call announcements. Consider a service for
   pre-paid calling cards. Once the pre-paid call is established, the
   system needs to set a timer to fire when they run out of minutes.
   When this timer fires, we would like the user to hear an announcement
   which tells them to enter a credit card to continue. Once they enter
   the credit card info, more money is added to the pre-paid card, and
   the user is reconnected to the destination party.

   We consider here the usage of third party call control just for
   playing the mid-call dialog to collect the credit card information.


       Pre-Paid User        Controller         Called Party        Media Server
             |                   |(1) INV SDP c=invld|                   |
             |                   |------------------>|                   |
             |                   |(2) 200 answer1    |                   |
             |                   |<------------------|                   |
             |                   |(3) ACK            |                   |
             |                   |------------------>|                   |
             |(4) INV no SDP     |                   |                   |



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             |<------------------|                   |                   |
             |(5) 200 offer2     |                   |                   |
             |------------------>|                   |                   |
             |                   |(6) INV offer2     |                   |
             |                   |-------------------------------------->|
             |                   |(7) 200 answer2    |                   |
             |                   |<--------------------------------------|
             |(8) ACK answer2    |                   |                   |
             |<------------------|                   |                   |
             |                   |(9) ACK            |                   |
             |                   |-------------------------------------->|
             |(10) RTP           |                   |                   |
             |...........................................................|
             |                   |(11) BYE           |                   |
             |                   |-------------------------------------->|
             |                   |(12) 200 OK        |                   |
             |                   |<--------------------------------------|
             |                   |(13) INV no SDP    |                   |
             |                   |------------------>|                   |
             |                   |(14) 200 offer3    |                   |
             |                   |<------------------|                   |
             |(15) INV offer3'   |                   |                   |
             |<------------------|                   |                   |
             |(16) 200 answer3'  |                   |                   |
             |------------------>|                   |                   |
             |                   |(17) ACK answer3'  |                   |
             |                   |------------------>|                   |
             |(18) ACK           |                   |                   |
             |<------------------|                   |                   |
             |(19) RTP           |                   |                   |
             |.......................................|                   |

                               Figure 13

   We assume the call is set up so that the controller is in the call as
   a B2BUA. When the timer fires, we wish to connect the caller to a
   media server. The flow for this is shown in Figure 13. When the timer
   expires, the controller places the called party with a connection
   address of rtp.invalid (1). This effectively ``disconnects'' the
   called party. The controller then sends an INVITE without SDP to the
   the pre-paid caller (4). The offer returned from the caller (5) is
   used in an INVITE to the media server which will be collecting digits
   (6). This is an instantiation of Flow I. This flow can only be used
   here because the media server is an automata, and will answer the
   INVITE immediately. If the controller was connecting the pre-paid
   user with another end user, Flow III would need to be used. The media
   server returns an immediate 200 OK (7) with an answer, which is
   passed to the caller in an ACK (8). The result is that the media



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   server and the pre-paid caller have their media streams connected.

   The media server plays an announcement, and prompts the user to enter
   a credit card number. After collecting the number, the card number is
   validated. The media server then passes the card number to the
   controller (using some means outside the scope of this
   specification), and then hangs up the call (11).

   After hanging up with the media server, the controller reconnects the
   user to the original called party. To do this, the controller sends
   an INVITE without SDP to the called party (13). The 200 OK (14)
   contains an offer, offer3. The controller modifies the SDP (as is
   done in Flow III), and passes the offer in an INVITE to the pre-paid
   user (15). The pre-paid user generates an answer in a 200 OK (16)
   which the controller passes to user B in the ACK (17). At this point,
   the caller and called party are reconnected.



































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11. Implementation Recommendations

   Most of the work involved in supporting third party call control is
   within the controller. A standard SIP UA should be controllable using
   the mechanisms described here. However, third party call control
   relies on a few features that might not be implemented. As such, we
   RECOMMEND that implementors of user agent servers to support the
   following:

   o  Offers and answers that contain a connection line with an address
      within the .invalid TLD.

   o  Re-invites that change the port to which media should be sent

   o  Re-invites that change the connection address

   o  Re-invites that add a media stream

   o  Re-invites that remove a media stream (setting its port to zero)

   o  Re-invites that add a codec amongst the set in a media stream

   o  SDP Connection address of zero

   o  Initial invites with a connection address of zero

   o  Initial invites with no SDP

   o  Initial invites with SDP but no media lines

   o  Re-invites with no SDP

   o  The UPDATE method [7]

   o  Reliability of provisional responses [6]

   o  Integration of resource management and SIP [2].














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12. Security Considerations

12.1 Identity

   The principal security consideration with third party call control is
   identity. When the controller initiates the call, what identity does
   it place in the From field of the INVITE? The controller could
   indicate that the call is from itself (From:
   sip:controller@example.com), but the call is really from some user,
   and is just facilitated by the controller. This impacts on how the
   call is authenticated by the end users.

   There are many cases, and the right one depends on the application of
   3pcc. In one common scenario, the controller is acting on behalf of
   one of the participants in the call. A typical example is
   click-to-dial, where the controller and the customer service
   representative are run by the same administrative domain. Indeed, for
   the purposes of identification, the controller can legitimately claim
   to be the customer service representative. In this scenario, it would
   be appropriate for the INVITE to the end user to contain a From field
   identifying the customer service rep, and authenticate the request
   using S/MIME signed by the key of the customer service rep (which is
   held by the controller)

   This requires the controller to actually have credentials with which
   it can authenticate itself as the customer support representative. In
   many other cases, the controller is representing one of the
   participants, but does not possess their credentials. Unfortunately,
   there are currently no standardized mechanisms that allow a user to
   delegate credentials to the controller in a way that limits their
   usage to specific third party call control operations. In the absence
   of such a mechanisms, the best that can be done is to use the display
   name in the From field to indicate the identity of the user on who's
   behalf the call is being made. It is RECOMMENDED that the display
   name be set to ``[controller] on behalf of [user]'', where user and
   controller are textual identities of the user and controller,
   respectively. In this case, the URI in the From field would identify
   the controller.

   In other situations, there is no real relationship between the
   controller and the participants in the call. In these situations,
   ideally the controller would have a means to assert that the call is
   from a particular identity (which could be one of the participants,
   or even a third party, depending on the application), and to validate
   that assertion with a signature using the key of the controller.

12.2 End-to-End Encryption and Integrity




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   With third party call control, the controller is actually one of the
   participants as far as the SIP dialog is concerened. Therefore,
   encryption and integrity of the SIP messages, as provided by S/MIME,
   will occur between participants and the controller, rather than
   directly between participants.

   However, end-to-end integrity, authenticity and confidentiality of
   the media sessions can be guaranteed through a controller. End-to-end
   media security is based on the exchange of keying material within SDP
   [12]. The proper operation of these mechanisms with third party call
   control depends on the controller behaving properly. So long as it is
   not attempting to explicitly disable these mechanisms, the protocols
   will properly operate end-to-end, resulting in a secure media session
   that even the controller cannot eavesdrop or modify. Since third
   party call control is based on a model of trust between the users and
   the controller, it is reasonable to assume it is operating in a
   well-behaved manner.


































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13. IANA Considerations

   There are no IANA considerations associated with this specification.
















































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14. Acknowledgements

   The authors would like to thank Paul Kyzivat, Rohan Mahy, Eric
   Rescorla, Allison Mankin and Sriram Parameswar for their comments.















































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Normative References

   [1]  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.

   [2]  Camarillo, G., Marshall, W. and J. Rosenberg, "Integration of
        Resource Management and Session Initiation Protocol (SIP)", RFC
        3312, October 2002.

   [3]  Bradner, S., "Key words for use in RFCs to Indicate Requirement
        Levels", BCP 14, RFC 2119, March 1997.

   [4]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
        Session Description Protocol (SDP)", RFC 3264, June 2002.

   [5]  Schulzrinne, H., Oran, D. and G. Camarillo, "The Reason Header
        Field for the Session Initiation Protocol (SIP)", RFC 3326,
        December 2002.

   [6]  jdrosen@dynamicsoft.com and schulzrinne@cs.columbia.edu,
        "Reliability of Provisional Responses in Session Initiation
        Protocol (SIP)", RFC 3262, June 2002.

   [7]  Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE
        Method", RFC 3311, October 2002.

   [8]  Eastlake, D. and A. Panitz, "Reserved Top Level DNS Names", BCP
        32, RFC 2606, June 1999.






















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Informative References

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

   [10]  Petrack, S. and L. Conroy, "The PINT Service Protocol:
         Extensions to SIP and SDP for IP Access to Telephone Call
         Services", RFC 2848, June 2000.

   [11]  IANA, "Special-Use IPv4 Addresses", RFC 3330, September 2002.

   [12]  Baugher, M. and D. Wing, "SDP Security Descriptions for Media
         Streams", draft-ietf-mmusic-sdescriptions-00 (work in
         progress), February 2003.


Authors' Addresses

   Jonathan Rosenberg
   dynamicsoft
   600 Lanidex Plaza
   Parsippany, NJ  07054
   US

   Phone: +1 973 952-5000
   EMail: jdrosen@dynamicsoft.com
   URI:   http://www.jdrosen.net


   Jon Peterson
   Neustar
   1800 Sutter Street
   Suite 570
   Concord, CA  94520
   US

   Phone: +1 925 363-8720
   EMail: jon.peterson@neustar.biz
   URI:   http://www.neustar.biz











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   Henning Schulzrinne
   Columbia University
   M/S 0401
   1214 Amsterdam Ave.
   New York, NY  10027
   US

   EMail: schulzrinne@cs.columbia.edu
   URI:   http://www.cs.columbia.edu/~hgs


   Gonzalo Camarillo
   Ericsson Advanced Signalling Research Lab
   FIN-02420 Jorvas
   Finland

   EMail: Gonzalo.Camarillo@ericsson.com


































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   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
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