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|
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Copyright © The IETF Trust (2008).
The "music on hold" feature is one of the most desired features of telephone systems in the business environment. "Music on hold" is where, when one party to a call has the call "on hold", that party's telephone provides an audio stream (often music) to be heard by the other party. Architectural features of SIP make it difficult to implement music-on-hold in a way that is fully compliant with the standards. The implementation of music-on-hold described in this document is fully effective and standards-compliant, but is simpler than the methods previously documented.
1.
Introduction
2.
Technique
2.1.
Placing a Call on Hold and Providing an External Media Stream
2.2.
Taking a Call off Hold and Terminating the External Media Stream
2.3.
Example Message Flow
2.4.
Re-INVITE and UPDATE from the Remote UA
2.5.
INVITE with Replaces
2.6.
Re-INVITE and UPDATE from the Music-On-Hold Source
2.7.
Payload Type Numbers
3.
Advantages
4.
Security Considerations
5.
Acknowledgments
6.
Revision History
6.1.
Changes from draft-worley-service-example-00 to draft-worley-service-example-01
6.2.
Changes from draft-worley-service-example-01 to draft-worley-service-example-02
7.
References
7.1.
Normative References
7.2.
Informative References
§
Author's Address
§
Intellectual Property and Copyright Statements
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Within SIP[1] (Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, “SIP: Session Initiation Protocol,” June 2002.)-based systems, it is desirable to be able to provide features that are similar to those provided by traditional telephony systems. A frequently requested feature is "music on hold": The music-on-hold feature is where, when one party to a call has the call "on hold", that party's telephone provides an audio stream (often music) to be heard by the other party.
Architectural features of SIP make it difficult to implement music-on-hold in a way that is fully compliant with the standards. The purpose of this document is to describe a method that is reasonably simple yet fully effective and standards-compliant.
| TOC |
The essence of the technique is that when the executing UA (the user's UA) performs a re-INVITE of the remote UA to establish the hold state, it provides no SDP[3] (Handley, M., Jacobson, V., and C. Perkins, “SDP: Session Description Protocol,” July 2006.) offer[2] (Rosenberg, J. and H. Schulzrinne, “An Offer/Answer Model with the Session Description Protocol (SDP),” June 2002.)[11] (Sawada, T. and P. Kyzivat, “SIP (Session Initiation Protocol) Usage of the Offer/Answer Model,” October 2007.), thus compelling the remote UA to provide an SDP offer. The executing UA then extracts the offer SDP from the remote UA's 2xx response, and uses that as the offer SDP in a new INVITE to the external media source. The external media source is thus directed to provide media directly to the remote UA. The media source's answer SDP is returned to the remote UA in the ACK to the re-INVITE.
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| TOC |
| TOC |
This section shows a message flow which is an example of this technique. The scenario is: Alice establishes a call with Bob. Bob then places the call on hold, with music-on-hold provided from an external source. Bob then takes the call off hold.
Note that this is just one possible message flow that illustrates this technique; numerous variations on these operations are allowed by the applicable standards.
Alice Bob Music Source Alice establishes the call: | | | | INVITE F1 | | |--------------->| | | 180 Ringing F2 | | |<---------------| | | 200 OK F3 | | |<---------------| | | ACK F4 | | |--------------->| | | RTP | | |<==============>| | | | | Bob places Alice on hold, compelling Alice's UA to provide SDP: | | | | INVITE F5 | | | (no SDP) | | |<---------------| | | 200 OK F6 | | | (SDP offer) | | |--------------->| | | | | Bob's UA initiates music-on-hold: | | | | | INVITE F7 | | | (SDP offer, | | | rev. hold) | | |------------->| | | 200 OK F8 | | | (SDP answer, | | | hold) | | |<-------------| | | ACK F9 | | |------------->| | | | Bob's UA provides an SDP answer containing the address/port of the Music Source: | | | | ACK F10 | | | (SDP answer, | | | hold | | |<---------------| | | no RTP | | | | | | Music-on-hold RTP | |<==============================| | | | The music on hold is active. Bob takes Alice off hold: | | | | INVITE F11 | | | (SDP offer) | | |<---------------| | | 200 OK F12 | | | (SDP answer) | | |--------------->| | | ACK F13 | | |<---------------| | | | BYE F14 | | |------------->| | | 200 F15 | | |<-------------| | RTP | | |<==============>| | | | | The normal media session between Alice and Bob is resumed.
Message Details
/* Alice calls Bob. */
F1 INVITE Alice -> Bob
INVITE sips:bob@biloxi.example.com SIP/2.0
Via: SIP/2.0/TLS atlanta.example.com:5061
;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sips:alice@atlanta.example.com>;tag=1234567
To: Bob <sips:bob@biloxi.example.com>
Call-ID: 12345600@atlanta.example.com
CSeq: 1 INVITE
Contact: <sips:a8342043f@atlanta.example.com;gr>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces, gruu
Content-Type: application/sdp
Content-Length: [omitted]
v=0
o=alice 2890844526 2890844526 IN IP4 atlanta.example.com
s=
c=IN IP4 atlanta.example.com
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F2 180 Ringing Bob -> Alice
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS atlanta.example.com:5061
;branch=z9hG4bK74bf9
;received=192.0.2.103
From: Alice <sips:alice@atlanta.example.com>;tag=1234567
To: Bob <sips:bob@biloxi.example.com>;tag=23431
Call-ID: 12345600@atlanta.example.com
CSeq: 1 INVITE
Contact: <sips:bob@biloxi.example.com>
Content-Length: 0
F3 200 OK Bob -> Alice
SIP/2.0 200 OK
Via: SIP/2.0/TLS atlanta.example.com:5061
;branch=z9hG4bK74bf9
;received=192.0.2.103
From: Alice <sips:alice@atlanta.example.com>;tag=1234567
To: Bob <sips:bob@biloxi.example.com>;tag=23431
Call-ID: 12345600@atlanta.example.com
CSeq: 1 INVITE
Contact: <sips:bob@biloxi.example.com>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: [omitted]
v=0
o=bob 2890844527 2890844527 IN IP4 biloxi.example.com
s=
c=IN IP4 biloxi.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 ACK Alice -> Bob
ACK sips:bob@biloxi.example.com SIP/2.0
Via: SIP/2.0/TLS atlanta.example.com:5061
;branch=z9hG4bK74bfd
Max-Forwards: 70
From: Alice <sips:alice@atlanta.example.com>;tag=1234567
To: Bob <sips:bob@biloxi.example.com>;tag=23431
Call-ID: 12345600@atlanta.example.com
CSeq: 1 ACK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Length: 0
/* Bob places Alice on hold. */
/* The re-INVITE contains no SDP, thus compelling Alice's UA
to provide an offer. */
F5 INVITE Bob -> Alice
INVITE sips:a8342043f@atlanta.example.com;gr SIP/2.0
Via: SIP/2.0/TLS biloxi.example.com:5061
;branch=z9hG4bK874bk
To: Alice <sips:alice@atlanta.example.com>;tag=1234567
From: Bob <sips:bob@biloxi.example.com>;tag=23431
Call-ID: 12345600@atlanta.example.com
CSeq: 712 INVITE
Contact: <sips:bob@biloxi.example.com>;+sip.rendering="no"
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Length: 0
/* Alice's UA provides an SDP offer.
Since it does not know that it is being put on hold,
the offer is the same as the original offer and describes
bidirectional media. */
F6 200 OK Alice -> Bob
SIP/2.0 200 OK
Via: SIP/2.0/TLS biloxi.example.com:5061
;branch=z9hG4bK874bk
;received=192.0.2.105
To: Alice <sips:alice@atlanta.example.com>;tag=1234567
From: Bob <sips:bob@biloxi.example.com>;tag=23431
Call-ID: 12345600@atlanta.example.com
CSeq: 712 INVITE
Contact: <sips:a8342043f@atlanta.example.com;gr>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces, gruu
Content-Type: application/sdp
Content-Length: [omitted]
v=0
o=alice 2890844526 2890844526 IN IP4 atlanta.example.com
s=
c=IN IP4 atlanta.example.com
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=active
/* Bob's UA initiates music-on-hold. */
/* This INVITE contains Alice's offer, but with the media
direction set to "reverse hold", receive-only. */
F7 INVITE Bob -> Music Source
INVITE sips:music@source.example.com SIP/2.0
Via: SIP/2.0/TLS biloxi.example.com:5061
;branch=z9hG4bKnashds9
Max-Forwards: 70
From: Bob <sips:bob@biloxi.example.com>;tag=02134
To: Music Source <sips:music@source.example.com>
Call-ID: 4802029847@biloxi.example.com
CSeq: 1 INVITE
Contact: <sips:bob@biloxi.example.com>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces, gruu
Content-Type: application/sdp
Content-Length: [omitted]
v=0
o=bob 2890844534 2890844534 IN IP4 atlanta.example.com
s=
c=IN IP4 atlanta.example.com
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=recvonly
F8 200 OK Music Source -> Bob
SIP/2.0 200 OK
Via: SIP/2.0/TLS biloxi.example.com:5061
;branch=z9hG4bKnashds9
;received=192.0.2.105
From: Bob <sips:bob@biloxi.example.com>;tag=02134
To: Music Source <sips:music@source.example.com>;tag=56323
Call-ID: 4802029847@biloxi.example.com
Contact: <sips:music@source.example.com>
CSeq: 1 INVITE
Content-Length: [omitted]
v=0
o=MusicSource 2890844576 2890844576 IN IP4 source.example.com
s=
c=IN IP4 source.example.com
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendonly
F9 ACK Bob -> Music Source
ACK sips:music@source.example.com SIP/2.0
Via: SIP/2.0/TLS source.example.com:5061
;branch=z9hG4bK74bT6
From: Bob <sips:bob@biloxi.example.com>;tag=02134
To: Music Source <sips:music@source.example.com>;tag=56323
Max-Forwards: 70
Call-ID: 4802029847@biloxi.example.com
CSeq: 1 ACK
Content-Length: 0
/* Bob's UA now sends the ACK that completes the re-INVITE
to Alice and completes the SDP offer/answer.
The ACK contains the SDP received from the Music Source,
and thus contains the address/port from which the Music Source
will send media. */
F10 ACK Bob -> Alice
ACK sips:a8342043f@atlanta.example.com;gr SIP/2.0
Via: SIP/2.0/TLS biloxi.example.com:5061
;branch=z9hG4bKq874b
To: Alice <sips:alice@atlanta.example.com>;tag=1234567
From: Bob <sips:bob@biloxi.example.com>;tag=23431
Call-ID: 12345600@atlanta.example.com
CSeq: 712 ACK
Contact: <sips:bob@biloxi.example.com>;+sip.rendering="no"
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Length: [omitted]
v=0
o=bob 2890844527 2890844528 IN IP4 biloxi.example.com
s=
c=IN IP4 source.example.com
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendonly
/* Bob picks up the call by sending a re-INVITE to Alice. */
F11 INVITE Bob -> Alice
INVITE sips:a8342043f@atlanta.example.com;gr SIP/2.0
Via: SIP/2.0/TLS biloxi.example.com:5061
;branch=z9hG4bK874bk
To: Alice <sips:alice@atlanta.example.com>;tag=1234567
From: Bob <sips:bob@biloxi.example.com>;tag=23431
Call-ID: 12345600@atlanta.example.com
CSeq: 713 INVITE
Contact: <sips:bob@biloxi.example.com>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: [omitted]
v=0
o=bob 2890844527 2890844529 IN IP4 biloxi.example.com
s=
c=IN IP4 biloxi.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F12 200 OK Alice -> Bob
SIP/2.0 200 OK
Via: SIP/2.0/TLS biloxi.example.com:5061
;branch=z9hG4bK874bk
;received=192.0.2.105
To: Alice <sips:alice@atlanta.example.com>;tag=1234567
From: Bob <sips:bob@biloxi.example.com>;tag=23431
Call-ID: 12345600@atlanta.example.com
CSeq: 713 INVITE
Contact: <sips:a8342043f@atlanta.example.com;gr>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces, gruu
Content-Type: application/sdp
Content-Length: [omitted]
v=0
o=alice 2890844526 2890844527 IN IP4 atlanta.example.com
s=
c=IN IP4 atlanta.example.com
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F13 ACK Bob -> Alice
ACK sips:a8342043f@atlanta.example.com;gr SIP/2.0
Via: SIP/2.0/TLS biloxi.example.com:5061
;branch=z9hG4bKq874b
To: Alice <sips:alice@atlanta.example.com>;tag=1234567
From: Bob <sips:bob@biloxi.example.com>;tag=23431
Call-ID: 12345600@atlanta.example.com
CSeq: 713 ACK
Contact: <sips:bob@biloxi.example.com>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Length: 0
F14 BYE Bob -> Music Source
BYE sips:music@source.example.com SIP/2.0
Via: SIP/2.0/TLS biloxi.example.com:5061
;branch=z9hG4bK74rf
Max-Forwards: 70
From: Bob <sips:bob@biloxi.example.com>;tag=02134
To: Music Source <sips:music@source.example.com>;tag=56323
Call-ID: 4802029847@biloxi.example.com
CSeq: 2 BYE
Contact: <sips:bob@biloxi.example.com>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces, gruu
Content-Length: [omitted]
F15 200 OK Music Source -> Alice
SIP/2.0 200 OK
Via: SIP/2.0/TLS atlanta.example.com:5061
;branch=z9hG4bK74rf
;received=192.0.2.103
From: Bob <sips:bob@biloxi.example.com>;tag=02134
To: Music Source <sips:music@source.example.com>;tag=56323
Call-ID: 4802029847@biloxi.example.com
CSeq: 2 BYE
Content-Length: 0
/* Normal media session between Alice and Bob is resumed */
| TOC |
While the call is on-hold, the remote UA can send a request to modify the SDP or the feature parameters of its Contact header. This can be done with either an INVITE or UPDATE method, both of which have much the same effect in regard to MOH.
A common reason for a re-INVITE will be when the remote UA desires to put the dialog on hold on its end. And because of the need to support this case, an implementation must process INVITEs and UPDATEs during the on-hold state as described below.
The executing UA handles these requests by echoing requests and responses: an incoming request from the remote UA causes the executing UA to send a similar request to the MOH source and an incoming response from the MOH source causes the executing UA to send a similar response to the remote UA. In all cases, SDP offers or answers that are received are added as bodies to the stimulated request or response to the other UA.
The passed-through SDP will usually need its o= line modified. The directionality attributes may need to be restricted. In regard to payload type numbers, since the mapping has already been established within the MOH dialog, a=rtpmap lines need not be added.
| TOC |
The executing UA must be prepared to receive INVITE requests with a Replaces headers that replaces the original dialog, and similarly it must be prepared to receive REFER requests within the dialog. The SDP within the new dialog is negotiated by being passed through to the MOH source within a new dialog with the MOH source. The SDP offer or answer can be passed to the MOH source with only modification to the o= line and directionality attributes.
In some cases, the previous dialog with the MOH source can be reused, but only if the executing UA presents the first offer within the new dialog, as otherwise there is no way to force the RTP payload types that have been used previously in the MOH dialog to be mapped to the correct codecs in the new dialog.
| TOC |
It is possible for the MOH source to send an INVITE or UPDATE request, and the executing UA can support doing so in similar manner as requests from the remote UA. However, if the MOH source is within the same administrative domain as the executing UA, the executing UA may have knowledge that the MOH source will not (or need not) make such requests, and so can respond to any such request with a failure response, avoiding the need to pass the request through.
However, in an environment in which ICE[8] (Rosenberg, J., “Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols,” October 2007.) is supported, the MOH source may need to send requests as part of ICE negotiation[9] (Elwell, J., “Subject: [Sipping] RE: I-D Action:draft-worley-service-example-00.txt,” November 2007.) with the remote UA. Hence, in environments that support ICE, the executing UA must be able to pass through requests from the MOH source as well as requests from the remote UA.
Again, as SDP is passed through, its o= line will need to be modified. In some cases, the directionality attributes will need to be restricted.
| TOC |
In this technique, the MOH source generates an SDP answer that the executing UA presents to the remote UA as an answer within the original dialog. In basic functionality, this presents no problem, because [2] (Rosenberg, J. and H. Schulzrinne, “An Offer/Answer Model with the Session Description Protocol (SDP),” June 2002.) (section 6.1, at the very end) specifies that the payload type numbers used in either direction of RTP are the ones specified in the SDP sent by the recipient of the RTP, which in this case is the remote UA, which composed the offer.
But strict compliance to [2] (Rosenberg, J. and H. Schulzrinne, “An Offer/Answer Model with the Session Description Protocol (SDP),” June 2002.) (section 8.3.2) requires that payload type numbers used in the SDP answer may duplicate the payload type numbers used in any offers and answers previously used in the dialog only if the payload type numbers represent the same media format (codec) as they did previously. However, the MOH source has no knowledge of the payload type numbers previously used in the original dialog, and it may accidentally specify a media format for a previously used payload type number in its answer (or in a subsequently generated INVITE or UPDATE). This would cause no problem with media decoding, as it cannot send any format that was not in the remote UA's offer, but it would violate [2] (Rosenberg, J. and H. Schulzrinne, “An Offer/Answer Model with the Session Description Protocol (SDP),” June 2002.).
We can prevent this problem by utilizing the requirement itself to control the behavior of the MOH source: When the executing UA is composing the INVITE to the MOH source, it compiles a list of all the (dynamically-assigned) payload type numbers which have been used in the original dialog but which are not mapped to a media format in the offer SDP. (The executing UA must be maintaining a list of all previously used payload type numbers anyway, in order to comply with [2] (Rosenberg, J. and H. Schulzrinne, “An Offer/Answer Model with the Session Description Protocol (SDP),” June 2002.).) Then, for each of these payload type numbers, it inserts session-level or media-level (as appropriate) a=rtpmap lines specifying the payload type number and the media format that it has been used for. Because of the reuse rule, the MOH source cannot propose those payload type numbers for any other media format.
However, once the payload type numbers have been defined within the MOH dialog, the executing UA need not insert additional a=rtpmap lines in later SDP that is passed through.
| TOC |
This technique for providing music-on-hold has advantages over other methods now in use:
| TOC |
Some UAs filter incoming media based on the address of origin in order to avoid SPIT. The technique described in this document ensures that any UA that should render MOH media will be informed of the source address of the media via the SDP that it receives. This should allow such UAs to filter without interfering with MOH operation.
| TOC |
The original version of this proposal was derived from [5] (Johnston, A., Sparks, R., Cunningham, C., Donovan, S., and K. Summers, “Session Initiation Protocol Service Examples,” October 2006.) and the similar implementation of MOH in the Snom UA. Significant improvements to the sequence of operations, allowing improvements to the SDP handling, were suggested by Venkatesh[7] (Venkatesh, “Subject: Re: [Sipping] I-D ACTION:draft-ietf-sipping-service-examples-11.txt,” October 2006.).
John Elwell[9] (Elwell, J., “Subject: [Sipping] RE: I-D Action:draft-worley-service-example-00.txt,” November 2007.) pointed out the need for the executing UA to pass through re-INVITEs/UPDATEs in order to allow ICE negotiation.
Paul Kyzivat[10] (Kyzivat, P., “Subject: Re: [Sipping] I-D ACTION:draft-ietf-sipping-service-examples-11.txt,” October 2006.) pointed out the difficulties regarding re-use of payload type numbers.
| TOC |
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Removed the original "Example Message Flow" and promoted the "Alternative Example Message Flow" to replace it because of a number of flaws that were described in the discussion of -00 on the SIPPING mailing list.
Described the use of the sip.rendering feature parameter to indicate on-hold status.
| TOC |
Added discussion of passing though re-INVITEs and UPDATEs.
Added discussion of payload type numbers.
Added Acknowledgments section.
| TOC |
| TOC |
| [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] | Rosenberg, J. and H. Schulzrinne, “An Offer/Answer Model with the Session Description Protocol (SDP),” RFC 3264, June 2002. |
| [3] | Handley, M., Jacobson, V., and C. Perkins, “SDP: Session Description Protocol,” RFC 4566, July 2006. |
| TOC |
| [4] | Johnston, A., Sparks, R., Cunningham, C., Donovan, S., and K. Summers, “Session Initiation Protocol Service Examples,” I-D draft-ietf-sipping-service-examples-13, July 2007. |
| [5] | Johnston, A., Sparks, R., Cunningham, C., Donovan, S., and K. Summers, “Session Initiation Protocol Service Examples,” I-D draft-ietf-sipping-service-examples-11, October 2006. |
| [6] | Rosenberg, J., Schulzrinne, H., and R. Mahy, “An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol (SIP),” RFC 4235, November 2005. |
| [7] | Venkatesh, “Subject: Re: [Sipping] I-D ACTION:draft-ietf-sipping-service-examples-11.txt,” IETF Sipping mailing list msg12180, October 2006. |
| [8] | Rosenberg, J., “Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols,” I-D draft-ietf-mmusic-ice-19, October 2007. |
| [9] | Elwell, J., “Subject: [Sipping] RE: I-D Action:draft-worley-service-example-00.txt,” IETF Sipping mailing list msg14678, November 2007. |
| [10] | Kyzivat, P., “Subject: Re: [Sipping] I-D ACTION:draft-ietf-sipping-service-examples-11.txt,” IETF Sipping mailing list msg12181, October 2006. |
| [11] | Sawada, T. and P. Kyzivat, “SIP (Session Initiation Protocol) Usage of the Offer/Answer Model,” I-D draft-ietf-sipping-sip-offeranswer-04, October 2007. |
| TOC |
| Dale R. Worley | |
| Nortel Networks Corp. | |
| 600 Technology Park Dr. | |
| Billerica, MA 01821 | |
| US | |
| Email: | dworley@nortel.com |
| URI: | http://www.nortel.com |
| TOC |
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