SIP D.R. Worley
Internet-Draft Bluesocket
Expires: July 19, 2008 January 16, 2008
Session Initiation Protocol Service Example -- Music on Hold
draft-worley-service-example-01
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Copyright Notice
Copyright (C) The IETF Trust (2008).
Abstract
The "music on hold" feature is one of the most desired features of
telephone systems in the business environment. "Music on hold" is
when one party to a call has the call "on hold", the other party
receives a media stream (often either music or advertising).
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.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Technique . . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1. Placing a Call on Hold and Providing an External Media
Stream . . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.2. Taking a Call off Hold and Terminating the External
Media Stream . . . . . . . . . . . . . . . . . . . . . . . 5
2.3. Example Message Flow . . . . . . . . . . . . . . . . . . . 6
2.4. Managing o= Lines . . . . . . . . . . . . . . . . . . . . 14
3. Advantages . . . . . . . . . . . . . . . . . . . . . . . . . . 16
4. Security Considerations . . . . . . . . . . . . . . . . . . . 17
5. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 18
6. Revision History . . . . . . . . . . . . . . . . . . . . . . . 19
6.1. Changes from draft-worley-service-example-00 to
draft-worley-service-example-01 . . . . . . . . . . . . . 19
7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
7.1. Normative References . . . . . . . . . . . . . . . . . . . 20
7.2. Informative References . . . . . . . . . . . . . . . . . . 20
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 21
Intellectual Property and Copyright Statements . . . . . . . . . . 22
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1. Introduction
Within SIP[1]-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 when one party to a call has the call "on
hold", that party's telephone provides a media stream to be rendered
to 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.
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2. Technique
The essence of the technique is that when the user's UA (referred to
hereafter as the "executing UA") establishes the hold state, it uses
third party call control mechanisms to direct RTP from an audio
source service to the held party. The executing UA establishes a
dialog with the service to negotiate the audio stream, but does not
act as a media relay; media flows from the service to the held party.
In order to accomplish this within the offer/answer model, the
executing UA sends a re-INVITE to the remote UA to establish the hold
state, but in that INVITE it provides no SDP offer, thus compelling
the remote UA to provide an SDP offer. The executing UA then uses
that offer (modified as described in Section 2.4) 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.
2.1. Placing a Call on Hold and Providing an External Media Stream
1. The executing user instructs the executing UA to put the dialog
on-hold.
2. The executing UA sends a re-INVITE without SDP to the remote UA,
which forces the remote UA to provide an SDP offer in its 2xx
response. The Contact header of the re-INVITE SHOULD include the
'+sip.rendering="no"' field parameter so that the intent is clear
and so that any dialog events indicate that it is putting the
call on hold.[6] (The sip.rendering feature parameter is not
required by the MOH mechanics, but rather to facilitate reporting
the on-hold status in dialog events.)
3. The remote UA sends a 2xx to the re-INVITE, and includes an SDP
offer giving its own listening address/port. If the remote UA
understands the sip.rendering feature parameter, the offer may
indicate that it will not send media by specifying the media
directionalities as "recvonly" (the reverse of "on-hold") or
perhaps "inactive". But the remote UA may offer to send media.
4. The executing UA uses this offer to construct the offer SDP of an
initial INVITE that it sends to the configured music-on-hold
(MOH) source. The SDP in this request is, with the exceptions
below, copied from the SDP returned by the remote UA in the
previous step, particularly regarding the provided listening
address/port and payload type numbers. The changes to the SDP
offer are:
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* The o= line of the offer is modified according to Section 2.4.
* Any explicit or implicit media directionality of "sendrecv" is
replaced with "recvonly", and any media directionality of
"sendonly" is replaced with "inactive".
5. The MOH sends a 2xx response to the INVITE, which contains an SDP
answer that should include its media source address as its
listening address/port. This SDP must necessarily[2] specify
"sendonly" or "inactive" as the directionality for all media
streams. (Although this address/port should receive no RTP, the
answer SDP will reach the remote UA, informing it of the address/
port from which the MOH media will come, and presumably
preventing it from ignoring the MOH media as SPIT. This requires
the SDP answer to provide a real address/port in the c= line,
even though the MOH server does not receive RTP.)
6. The executing UA copies the MOH service SDP answer, modifying it
as listed below, and sends it as its SDP answer in the ACK for
the re-INVITE to the remote UA.
* The o= line of the answer is modified according to
Section 2.4.
7. Due to the sip.rendering feature parameter in the Contact of the
re-INVITE and the media directionality in the SDP answer
contained in the ACK, the on-hold state (at the executing end) of
the dialog is established. (The sip.rendering feature parameter
will be reported in the dialog events generated by both UAs,
allowing any monitoring application to determine that the call is
on hold.)
8. After this point, the MOH source generates RTP containing the
music-on-hold media, and sends it directly to the listening
address/port of the remote UA. The executing UA maintains two
dialogs (one to the remote UA, one to the MOH source), but does
not see or handle the MOH RTP.
2.2. Taking a Call off Hold and Terminating the External Media Stream
1. The executing user instructs the executing UA to take the dialog
off-hold.
2. The executing UA sends a re-INVITE to the remote UA with SDP that
requests to receive media. The Contact header of the re-INVITE
does not include the '+sip.rendering="no"' field parameter. (It
may contain a sip.rendering field parameter with value "yes" or
"unknown", or it may omit the field parameter.) Thus this INVITE
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removes the on-hold state (at the executing end) of the dialog.
3. When the remote UA sends a 2xx response to the re-INVITE, the
executing UA sends a BYE request in the dialog to the MOH source
to terminate the MOH media stream.
4. After this point, the MOH source does not generate RTP and
ordinary RTP flow is re-established in the original dialog.
2.3. Example Message Flow
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 server. 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 Server
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) | |
|--------------->| |
| | |
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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 SDP answer containing the address/port
of the Music Server:
| | |
| ACK (hold) F10 | |
| (SDP answer) | |
|<---------------| |
| 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 | |
|<==============>| |
| | |
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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
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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
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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 Server
INVITE sips:music@server.example.com SIP/2.0
Via: SIP/2.0/TLS biloxi.example.com:5061
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;branch=z9hG4bKnashds9
Max-Forwards: 70
From: Bob <sips:bob@biloxi.example.com>;tag=02134
To: Music Server <sips:music@server.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 Server -> 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 Server <sips:music@server.example.com>;tag=56323
Call-ID: 4802029847@biloxi.example.com
Contact: <sips:music@server.example.com>
CSeq: 1 INVITE
Content-Length: [omitted]
v=0
o=MusicServer 2890844576 2890844576 IN IP4 server.example.com
s=
c=IN IP4 server.example.com
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendonly
F9 ACK Bob -> Music Server
ACK sips:music@server.example.com SIP/2.0
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Via: SIP/2.0/TLS server.example.com:5061
;branch=z9hG4bK74bT6
From: Bob <sips:bob@biloxi.example.com>;tag=02134
To: Music Server <sips:music@server.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 Server,
and thus contains the address/port from which the Music Server
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 server.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
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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
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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 Server
BYE sips:music@server.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 Server <sips:music@server.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 Server -> 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 Server <sips:music@server.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 */
2.4. Managing o= Lines
The rulesSection 2.4 regarding the use of o= lines in successive SDP
offers/answers during a dialog are quit strict. In particular,
successive SDPs sent by one UA in the dialog must have identical o=
lines, other than that the version number field must be incremented
by 1. The single exception is that if two or more successive SDPs
are exactly the same, they may have the same version number.
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In order for the executing UA to conform to these rules, it must
modify the o= lines in any SDP that it passes from one dialog to the
other. In particular:
1. The SDP in the initial INVITE to to the MOH server must have its
o= line replaced with a newly constructed o= line that is valid
within the dialog with the MOH server -- See [3] for the details.
2. Within any later SDP copied from the original dialog to the MOH
dialog, the o= line must be replaced with a copy of the o= line
that the executing UA used in the previous SDP that it sent in
the MOH dialog, with the version number incremented by 1.
3. Within any SDP copied from the MOH dialog to the original dialog,
the o= line must be replaced with a copy of the o= line that the
executing UA used in the previous SDP that it sent in the
original dialog, with the version number incremented by 1.
4. When the executing UA takes the call off-hold, the o= line of the
SDP that it sends in the re-INVITE must have a version number one
greater than in the o= line of the last SDP the executing UA
copied from the MOH dialog.
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3. Advantages
This technique for providing music-on-hold has advantages over other
methods now in use:
1. The original dialog is not transferred to another UA, so the
"remote endpoint URI" displayed by the remote endpoint's user
interface and dialog event package[6] does not change during the
call.[4]
2. The music-on-hold media is sent directly from the music-on-hold
source to the remote UA, rather than being relayed through the
holding UA.
3. The remote UA sees, in the incoming SDP, the address/port that
the MOH server will send MOH media from, thus allowing it to
render the media, even if it is filtering incoming media based on
originating address as a SPIT preventative.
4. The technique requires relatively simple manipulation of SDP, and
in particular: (1) does not require a SIP agent to modify
unrelated SDP to be acceptable to be sent within an already
established sequence of SDP (a problem with [5]), and (2) does
not require converting an SDP answer into an SDP offer (which was
a problem with the -00 version of this document, as well as with
[5]).
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4. Security Considerations
Some UAs filter incoming media based on the address of origin in
order to avoid SPIT. This technique ensures that any UA that should
render MOH media will be informed of the source address via the SDP
that it receives. This should allow such UAs to filter without
interfering with MOH operation.
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5. Acknowledgments
The original version of this proposal was derived from [5] and the
similar implementation of MOH in the Snom UA. Significant
improvements to sequence of operations, allowing improvements to the
SDP handling, were suggested by Venkatesh[7].
John Elwell[8] pointed out the need for the executing UA to pass
through re-INVITEs/UPDATEs in order to allow ICE negotiation, which
suggested to me the need for pass-through to handle the remote UA
placing its end of the call on-hold.
This version benefitted from Scott Lawrence's careful reading and
comments.
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6. Revision History
6.1. Changes from draft-worley-service-example-00 to
draft-worley-service-example-01
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.
Added Acknowledgments section.
Added separate section on the management of o= lines.
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7. References
7.1. 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] 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.
7.2. Informative References
[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] Elwell, J., "Subject: [Sipping] RE: I-D
Action:draft-worley-service-example-00.txt", IETF Sipping
mailing list msg14678, November 2007.
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Author's Address
Dale R. Worley
Bluesocket Inc.
10 North Ave.
Burlington, MA 01803
US
Phone: +1 781 229 0533 x173
Email: dworley@pingtel.com
URI: http://www.pingtel.com
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