SIP 183 Session Progress Message
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http://tools.ietf.org/html/draft-ietf-sip-183-00
SIP 183 Session Progress Message
Status of this Memo
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026
.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that other
groups may also distribute working documents as Internet-Drafts.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet- Drafts as reference mate-
rial or to cite them other than as "work in progress."
The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/lid-abstracts.txt
.
The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html
.
Abstract
This document describes a proposed extension to SIP [1
]. This exten-
sion adds the 183 Session Progress response and a new header to indi-
cate why a SDP message body is included in a 18x message.
The introduction of the 183 informational response message would
allow a called user agent to indicate to the calling user agent
whether or not the calling user agent should apply local alerting for
the session. The existing 180 Ringing message would indicate that
the calling user agent has the option of providing local alerting
(and generally should). The 183 Session Progress message would indi-
cate that the calling user agent should not provide local alerting.
In additionally, the calling user agent may be called on to establish
a media session to be used by the called user agent to indicate the
status of the session setup request as part of the indicated media
stream. The indication of whether or not to play early media to the
calling user would be controlled with a new Session header included
in the 183 message.
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Internet Draft SIP 183 Session Progress Message October 1999
1
There are instances, most notably dealing with SIP to PSTN interwork-
ing, that necessitate that the SIP called User Agent (UA) be able to
suppress local alerting by the SIP calling UA and to set up a prelim-
inary media session from the called UA to the calling UA. This would
allow the called UA to play back media prior to the full SIP session
being set up. This media would be used to report on the status of
the session setup request. It could also be used to play music while
the session setup is attempted. This would be useful for find-me
like services that involve attempting multiple locations for a single
setup request.
The only method in the current SIP specification that allows the
called UA to playback media is to set up a full SIP session. In PSTN
interworking situations (and likely in end-to-end SIP sessions) this
will cause a billing relationship to be established between networks
for the session. This causes a problem when the reason for setting
up the media session is to indicate a failure in the session setup.
This document proposes an extension to the Session Initiation Proto-
col (SIP) that introduces this capability.
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2
In the PSTN today there are times when a media (voice) path is set up
from the called party to the calling party in order to play a treat-
ment (a special tone or announcement). The treatment can range from
alerting (ring back) to busy tones to announcements explaining why
the call could not be set up. The participants in this call are not
charged for the remote treatment portion of the call.
This one way voice path is generally set up as part of the processing
of the SS#7 ISUP ACM message.
The following call flow illustrates call setup using SS7 ISUP in a
PSTN network.
Originating Terminating
Network Network
IAM---------------------->
<----------------------ACM
<=========================
One way voice path
* <----------------------ANM
<========================>
Two way voice path
REL---------------------->
<----------------------RLC
* If the originating network is a Local Exchange Carrier and the termi-
nating network is an Interexchange Carrier then the LEC will start
charging for the call at this point in the call.
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The following call flow illustrates the setup of a call that does not
result in a completed call but does involve a media path being set
up. In this case, the terminating network may be playing a busy sig-
nal or playing an announcement. The following are examples of
announcements that might be played in this scenario:
- The number you have dialed is no longer valid.
- The wireless subscriber you are calling is not currently reachable.
Originating Terminating
Network Network
IAM---------------------->
<----------------------ACM
<=========================
One way voice path
REL---------------------->
<----------------------RLC
2.1
The following are a subset of the requirements for interworking
between a PSTN network and a SIP network.
When the SIP network is in the middle of two PSTN networks, it must
support the following:
- The ingress gateway into the SIP network shall have the ability to
determine, based on SIP signaling messages, when to send an ISUP ACM
message and when to send an ISUP ANM message.
- The SIP network shall have the ability to support fast setup. This
occurs when the terminating network does not send an ACM prior to
sending an ANM.
- The SIP network shall support the ability to cut through a voice
path from the terminating PSTN network to the originating PSTN net-
work without the interim SIP network incurring charges from the orig-
inating network.
The SIP network shall support the ability to place calls to a PSTN
network without the egress gateway knowing what type of device the
call was originated from. Thus, the egress gateway shall not need to
behave differently when the call originates from a PSTN network then
when the call originates from a native IP SIP device.
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The following is an illustration of the two scenarios that must be
supported:
+-------------+ +---------+ +-------------+
| Originating | +-----+ | SIP | +-----+ | Terminating |
| PSTN |-->| IGW |-->| Network |-->| EGW |--->| PSTN |
| Network | +-----+ | | +-----+ | Network |
+-------------+ +---------+ +-------------+
IGW = Ingress Gateway
EGW = Egress Gateway
+---------+ +-------------+
| SIP | +-----+ | Terminating |
| IP |-->| EGW |--->| PSTN |
| Device | +-----+ | Network |
+---------+ +-------------+
2.2
The following sections show the results of investigating various
options for addressing the above requirements using existing SIP pro-
tocol capabilities. In each case, it is shown why the option either
cannot address the requirements or has short comings that can be bet-
ter addressed using the 183 Session Progress message.
2.2.1
The first option investigated involved mapping the 100 Trying message
to the ACM message.
The following call flow illustrates this option.
Originating Ingress Egress Terminating
Network SIP GW SIP GW Network
------------>------------>----------->
IAM INVITE IAM
<-----------<------------<------------
ACM 100 Trying ACM
<=========== <============
One way voice One way voice
The call flow breaks at this point for two reasons. First, at this
point in the call flow a one way voice path is needed so that the
terminating network can provide session setup status as part of the
voice path. The 100 Trying does not cause a voice path cut-through
between the ingress and egress gateways. This potentially could be
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Internet Draft SIP 183 Session Progress Message October 1999
addressed by allowing the 100 Trying to carry SDP information to be
used for carrying the preliminary session media. This option is
explored in the context of the 180 Ringing message in section 3.3
.
The use of the 100 Trying also fails because a SIP Proxy Server sit-
ting in the signaling path between the ingress gateway and the egress
gateway might have generated the 100 Trying message. This would
result in the ACM message being sent prior to the egress gateway
receiving an ACM from the terminating network.
2.2.2
The second option investigated was to use a 180 Ringing message to
trigger the ACM message at the ingress gateway.
This option is illustrated in the following call flow:
Originating Ingress Egress Terminating
Network SIP GW SIP GW Network
------------>------------>----------->
IAM INVITE IAM
<-----------<------------<------------
ACM 180 Ringing ACM
<=========== <============
One way voice One way voice
This option breaks at this point because a media voice path cannot be
cut through at this point for the terminating PSTN network to report
on the session progress. This is due to the fact that the egress
gateway has not yet communicated its RTP information to the ingress
gateway. The next two options attempt to address this issue.
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Internet Draft SIP 183 Session Progress Message October 1999
2.2.3
The next option investigated involves using the presence of SDP in
the 180 Ringing message to indicate that session progress will be
communicated by the called user agent using the media stream. In
this case, absence of the SDP message body would indicate that local
alerting should take place.
The following call flow illustrates this option:
Originating Ingress Egress Terminating
Network SIP GW SIP GW Network
------------>------------>----------->
IAM INVITE IAM
<-----------<------------<------------
ACM 180 Ringing ACM
One way SDP
<============
One way voice path
<-----------<------------<------------
ANM 200 OK ANM
Two way SDP
------------>
ACK
<====================================>
Two Way Voice Path
<-----------<------------<------------
REL BYE REL
----------->
RLC
------------>
200 OK
------------>
RLC
Although this option looks promising on first review, it does not
give the called user agent the ability to include SDP in the message
and rely on the calling user agent (the ingress gateway in this sce-
nario) to provide local alerting. As illustrated in [2
] there are
other reasons that SDP might be included in a 180 Ringing message.
Thus the user requiring a coupling of SIP and QOS signaling, which
requires inclusion of SDP in the 18x message, could not also request
local alerting.
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Internet Draft SIP 183 Session Progress Message October 1999
2.2.4
The final option investigated involves setting up a full media ses-
sion in the SIP network prior to receiving the ANM from the terminat-
ing PSTN network. This involves mapping the 200 OK to the ACM mes-
sage at the ingress gateway and having the egress gateway send a re-
INVITE upon receipt of the ANM. The ingress gateway would use the
re-INVITE to trigger the ANM message.
This option is illustrated in the following call flow:
Originating Ingress Egress Terminating
Network SIP GW SIP GW Network
------------>------------>----------->
IAM INVITE IAM
<-----------<------------<------------
ACM 200 OK ACM
One way SDP
------------>
ACK
<============
One way voice path
<-----------<------------<------------
ANM INVITE ANM
------------>
200 OK
<------------
ACK
<====================================>
Two Way Voice Path
<-----------<------------<------------
REL BYE REL
----------->
RLC
------------>
200 OK
------------>
RLC
Although this will work in the above scenario, it introduces additional
messaging overhead. In addition, as illustrated in the following fast
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Internet Draft SIP 183 Session Progress Message October 1999
answer call flow, it is at best awkward and may result in clipping off
of the beginning of the voice call.
Originating Ingress Egress Terminating
Network SIP GW SIP GW Network
------------>------------>----------->
IAM INVITE IAM
<-----------<------------<------------
ACM 200 OK ANM
One way SDP
<-----------<------------
ANM INVITE
Two way SDP
------------>
ACK
<-----------
200 OK
----------->
ACK
<====================================>
Two Way Voice Path
<-----------<------------<------------
REL BYE REL
----------->
RLC
------------>
200 OK
------------>
RLC
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2.3
The following session signaling flows show the proposed solution
using the 183 Session Progress Message to map to the ISUP ACM message
and how the 183 Session Progress message is used for when the call
originates from a SIP IP Device.
2.3.1
The following session signaling flow shows the use of the 183 Session
Progress message for a session setup in a SIP based network when the
session will be between two PSTN networks.
Originating Ingress Egress Terminating
Network SIP GW SIP GW Network
------------>------------>----------->
IAM INVITE IAM
<-----------<------------<------------
ACM 183 Session ACM
Progress
One way SDP
<============
One way voice path
<-----------<------------<------------
ANM 200 OK ANM
Two way SDP
------------>
ACK
<====================================>
Two Way Voice Path
<-----------<------------<------------
REL BYE REL
----------->
RLC
------------>
200 OK
------------>
RLC
2.3.2
The following session signaling flow shows the method for handling of
the fast answer scenario. Note that in this case the 183 Session
Progress message is not used, as the ANM is mapped directly to a 200
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Internet Draft SIP 183 Session Progress Message October 1999
OK. This meets the requirement that the SIP gateways must be able to
differentiate between ACM and ANM messages.
Originating Ingress Egress Terminating
Network SIP GW SIP GW Network
------------>------------>----------->
IAM INVITE IAM
<-----------<------------<------------
ANM 200 OK ANM
Two way SDP
------------>
ACK
<====================================>
Two Way Voice Path
<-----------<------------<------------
REL BYE REL
----------->
RLC
------------>
200 OK
------------>
RLC
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2.3.3
The following session signaling flow shows the use of the 183 Session
Progress message for a session setup in a SIP based network when the
session originates in the SIP network and terminates to a PSTN net-
work.
Calling Egress Terminating
UA SIP GW Network
------------>------------>
INVITE IAM
<-----------
100 Trying
<-----------<------------
183 Session ACM
Progress
One way SDP
<========================
One way voice path
<-----------<------------
200 OK ANM
------------>
ACK
<========================
Two Way Voice Path
<-----------<------------
BYE REL
------------>
RLC
------------>
200 OK
3
The previous sections illustrated how the 183 response can be used to
communicate the need for the calling user agent to receive media
prior to receiving a final response from the called user agent.
There are other reasons for including SDP message bodies in an 18x
message.
One possible use is to use the SDP to establish security and/or QoS
relationship between the called and calling user agents, as
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Internet Draft SIP 183 Session Progress Message October 1999
documented in [2
].
As a result, it is necessary to communicate to the calling user agent
why the SDP was included in the 18x message. The following section
describes the Session header, which will be used for this purpose.
3.1
The Session header will be used to communicate to the calling user
agent the reason for a SDP message body being included in an 18x mes-
sage. The valid values for the Session header are Media, QoS and
Security. Multiple of these values can be indicated.
A value of Media indicates that the SDP should be used for establish-
ing of an early media session. The early media session will gener-
ally be used to communicate the status of the session but could also
be used for other reasons. For instance, it could be used to play
music while the calling user is being alerted.
A value of QoS indicates that the SDP should be used for establishing
of a QoS relationship between the calling and called user agents.
This could involve the user agents requesting QoS resources using
RSVP or some other signaling mechanism.
A value of Security indicates that the SDP should be used for estab-
lishing of a security relationship between the calling and called
user agents. This could be an IPSEC based relationship, for example.
4
The format of provisional responses with media session descriptions
is identical to that of 200-class responses to INVITE requests,
except as noted in section 7
, "Reliability"; the body will contain a
session description (usually SDP; see ref [5
]).
4.1
Under most circumstances, provisional responses used to initiate tem-
porary media will contain SDP that is a subset of the media descrip-
tion presented in the INVITE message (as in normal 200 responses).
If the original INVITE message contains no media description, the
server will generate SDP representing the capabilities it requires
for media transmission and include it in the provisional response.
The client will include a final SDP in its acknowledgement of receipt
(see section 4
, "Reliability" ).
In both cases, the media streams will be established after the mes-
sage confirming receipt of the provisional response has been sent
(from the clients perspective) or received (from the servers per-
spective).
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The designation of media capabilities in a provisional response has
no implications on the capabilities of any subsequent temporary con-
nections or the final connection. Each media stream is negotiated
relative to the session description in the original INVITE request
(or lack thereof).
4.2
After a temporary media stream has been established, its parameters
can be changed by sending further provisional responses that also
contain session descriptions. Upon receipt of such a response, the
client MUST immediately cease transmission of media relating to the
old temporary stream. As before, the new temporary media stream is
established after acknowledgement of the provisional response.
Provisional responses which contain no session description SHOULD NOT
have an effect on any currently established temporary media stream.
4.3
Sending of temporary media MUST be discontinued upon the sending
(from the servers perspective) or the receipt (from the clients
perspective) of any INVITE final response.
Sending a provisional response that contains a session description
with all media stream port numbers set to zero can also discontinue a
temporary media stream.
4.4
To allow for transmission of temporary media which does not corre-
spond to the four provisional status codes defined in the SIP RFC
(ref [1
]), this protocol extension defines one additional response
code of "183 Session Progress."
The 183 Session Progress response can be used for any arbitrary in-
band communication of call status. It SHOULD NOT, however, be used to
convey ringing, forwarding, or call queueing situations.
When applicable, the response text SHOULD include a text representa-
tion of the information conveyed by the media stream. In the case of
a recorded announcement, this text SHOULD be the text of the
announcement. For a tone, this text SHOULD be either the name of the
tone as defined in E.182 (ref [6
]) (e.g. "Payphone Recognition Tone")
or a description of the condition the tone is attempting to report
(e.g. "The Called Party is a Payphone").
4.5
Clients which understand this extension SHOULD also understand the
extension described in "Reliability of Provisional Responses in SIP"
(ref [3
]) and indicate that they require reliable transmission of
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Internet Draft SIP 183 Session Progress Message October 1999
provisional responses in Require: and Proxy-Require: headers.
4.6
If no acceptable media type is available in the clients INVITE
request session description, the server MAY return a "406 Not Accept-
able" message; the alternative is to forgo the transmission of provi-
sional media. While it is perhaps a more appropriate error code, "606
Not Acceptable" is not suggested, owing to its properties of termi-
nating any ongoing searches.
If the client finds the session description proposed by the server in
a provisional response unacceptable, its acknowledgement SHOULD con-
tain a session description with all media stream port numbers set to
zero. A server which receives such a message MAY respond with a "406
Not Acceptable" message; the alternative is to forgo the transmission
of provisional media.
4.7
There are situations when the called user agent (the UAS) requires
the support of the mechanisms defined in this document and does not
know through the INVITE message whether the calling user agent (the
UAC) supports the extension. There is currently no method for the
UAS to communicate to the UAC the required capabilities required to
properly setup the session.
This is a general problem with the SIP protocol that should be fixed
in an upcoming version. The mechanisms established for the SIP pro-
tocol will be used to indicate a need for this extension.
5
The remainder of the document describes the proposed extensions to
the SIP specification. The section number indicates the section of
the SIP specification that requires modification. Thus section 5
.M.N
would include proposed modifications to section M.N of the SIP speci-
fication.
Absence of a section indicates that no modifications are proposed for
that section.
5.5.1.1
The following is the updated Figure 5:
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Internet Draft SIP 183 Session Progress Message October 1999
Informational = "100" ; Trying
| "180" ; Ringing
| "181" ; Call is Being Forwarded
| "182" ; Queued
| "183" ; Session Progress
Success = "200" ; OK
Figure 5: Informational and Success codes
5.6
The following needs to be added to Table 5
where enc. e-e ACK BYE CAN INV OPT REG
--------------------------------------------------------------
Session 18x e - - - - - -
5.6.43
The session header is used in 18x response messages to communicate to
the UAC the purpose of the session description message body included
in response message.
The valid values are Media, QoS and Security.
A value of Media indicates that the SDP should be used for establish-
ing of an early media session. The early media session will gener-
ally be used to communicate the status of the session but could also
be used for other reasons. For instance, it could be used to play
music while the calling user is being alerted.
A value of QoS indicates that the SDP should be used for establishing
of a QoS relationship between the calling and called user agents.
This could involve the user agents requesting QoS resources using
RSVP or some other signaling mechanism.
A value of Security indicates that the SDP should be used for estab-
lishing of a security relationship between the calling and called
user agents. This could be an IPSEC based relationship, for example.
See [2
] for details on the handling of the QoS and Security values.
Session = "Session" ":" 1#( "Media" | "QoS" | "Security" )
5.7
5.7.1
5.7.1.2
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The following text is proposed to be added to the description of the
180 Ringing message:
The calling UA should initiate local alerting (for instance, the
playing of a ringing tone or other alerting mechanism) so as to indi-
cate the progress of the session setup.
5.7.1.5
The called UA may have the need to communicate session progress with-
out the indication that the called user is being alerted.
The calling UA should not apply local alerting upon receipt of the
183 Session Progress Response.
The 183 Session Progress may have been sent to communicate the status
of the session setup attempt as part of a media stream. The called
user agent will indicate this by including the Session header with a
value of media.
In this case, the calling UA shall establish a media session accord-
ing to the contents of the session description contained in the 183
message. The calling UA should not apply local alerting that would
interfere with the media session information supplied by the called
UA.
The 183 message SHOULD include enough session description information
to allow for a media session between the called UA and the calling
UA.
Although not strictly required for a one way voice path to be setup
between the egress gateway and the ingress gateway, the SDP in the
183 has the following benefits:
1. The list of audio (or video) codecs is reduced, so the calling
gateway need only expect a smaller set.
2. The 183 can contain security preconditions in the SDP (if they
were in the SDP in the INVITE), so that the calling gateway can per-
form appropriate authentication/encryption for each media stream from
each egress gateway.
3. If any kind of pre-call announcement requires two-way media (per-
haps some kind of speech recognition for credit card numbers, or even
DTMF too), the SDP in the 183 is needed.
5.8
5.8.1
The following is proposed rewording of paragraph 2 in Section 8.1
of
the SIP specification:
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Internet Draft SIP 183 Session Progress Message October 1999
For response messages, the request method and the response status
code determine the type and interpretation of any message body. All
responses MAY include a body. Message bodies for 1xx responses con-
tain advisory information about the progress of the request. In
addition, message bodies for 1xx responses can contain session
descriptions. 2xx responses ...
5.10
5.10.1.2
The following is proposed text for inclusion in section 10.1.2
of the
SIP specification:
183 responses SHALL always be forwarded.
5.11
5.11.6
When the called UA receives and INVITE message that results in the
need to report on the status of the media setup through a media
stream, the called UA has the option to send a 183 message with a
session description to the calling UA.
5.11.7
When the calling UA receives a 183 response that contains a session
description and an indication that the session description is for
early media, it SHALL setup the associated media session and present
any media received from the called UA to the user.
5.13
The security considerations for the 183 Session Progress message are
the same as for SIP in general.
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5.16
5.16.9
The following call flow illustrates the case where a call is origi-
nating from a PSTN network, transiting a SIP network and being deliv-
ered to a second PSTN network. In this case, the 183 message is used
to trigger the ACM message and results in a one way media session
being setup through the SIP network.
Originating Ingress Egress Terminating
Network SIP GW SIP GW Network
------------>------------>----------->
IAM INVITE IAM
<------------
100 Trying
<-----------<------------<------------
ACM 183 Session ACM
Progress
SESSION: Media
One way SDP
<=====================================
One way voice path
<-----------<------------<------------
ANM 200 OK ANM
------------>
ACK
<====================================>
Two Way Voice Path
<-----------<------------<------------
REL BYE REL
----------->
RLC
------------>
200 OK
------------>
RLC
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5.16.10
Calling Egress Terminating
UA SIP GW Network
------------>------------>
INVITE IAM
<-----------
100 Trying
<-----------<------------
183 Session ACM
Progress
Session: Media
One way SDP
<========================
One way voice path
<-----------<------------
200 OK ANM
------------>
ACK
<========================
Two Way Voice Path
<-----------<------------
BYE REL
------------>
RLC
------------>
200 OK
5.A Minimal Implementation
5.A.1 Client
The following is a suggested addition to Appendix A.1
of the SIP
specification:
PSTN Interworking: If a client wishes to interwork properly with PSTN
networks then it MUST support the 183 Session Progress message.
6
Only the relevant headers have been included in the following exam-
ples. Notably, the mandatory parameters Call-ID and CSeq are not
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shown.
6.1
Client to server:
INVITE sip:+12145551212@bell.com SIP/2.0
RAck: 0
To: sip:+12145551212@bell.com
From: sip:+15125559876@domain.com
Require: org.ietf.sip.reliable-100
Proxy-Require: org.ietf.sip.reliable-100
Content-Type: application/sdp
v=0
o=- 929142225 929142225 IN IP4 vgw.domain.com
c=IN IP4 vgw.domain.com
M=audio 49152 RTP/AVP 0 1
a=rtpmap:0 PCMU/8000
a=rtpmap:1 1016/8000
Server to client:
SIP/2.0 180 Ringing
RSeq: 1
To: sip:+12145551212@bell.com
From: sip:+15125559876@domain.com
Session: Media
Content-Type: application/sdp
v=0
o=- 929142942 929142942 IN IP4 media.bell.com
c=IN IP4 media.bell.com
M=audio 49180 RTP/AVP 0
a=rtpmap:0 PCMU/8000
Client to server:
INVITE sip:+12145551212@bell.com SIP/2.0
RAck: 1
To: sip:+12145551212@bell.com
From: sip:+15125559876@domain.com
Require: org.ietf.sip.reliable-100
Proxy-Require: org.ietf.sip.reliable-100
[Remote ringing tone is played]
Donovan, et al. draft-ietf-sip-183-00.txt
Page 21
Internet Draft SIP 183 Session Progress Message October 1999
Server to client:
SIP/2.0 182 Call is queued; est. wait is 5 minutes
RSeq: 2
To: sip:+12145551212@bell.com
From: sip:+15125559876@domain.com
Session: Media
Content-Type: application/sdp
v=0
o=- 929143057 929143057 IN IP4 media.bell.com
c=IN IP4 media.bell.com
M=audio 49180 RTP/AVP 1
a=rtpmap:1 1016/8000
[Ring tone is discontinued]
Client to server:
INVITE sip:+12145551212@bell.com SIP/2.0
RAck: 2
To: sip:+12145551212@bell.com
From: sip:+15125559876@domain.com
Require: org.ietf.sip.reliable-100
Proxy-Require: org.ietf.sip.reliable-100
["Your call is queued" announcement is played, followed by hold
music]
Server to client:
SIP/2.0 200 OK
To: sip:+12145551212@bell.com
From: sip:+15125559876@domain.com
Content-Type: application/sdp
v=0
o=- 929143373 929143373 IN IP4 vgw.bell.com
c=IN IP4 mg.bell.com
M=audio 49199 RTP/AVP 1
a=rtpmap:1 1016/8000
[Hold music is discontinued]
Donovan, et al. draft-ietf-sip-183-00.txt
Page 22
Internet Draft SIP 183 Session Progress Message October 1999
Client to server:
ACK sip:+12145551212@bell.com SIP/2.0
To: sip:+12145551212@bell.com
From: sip:+15125559876@domain.com
[Final media stream is established]
6.2
Client to server:
INVITE sip:+12145551212@bell.com SIP/2.0
RAck: 0
To: sip:+12145551212@bell.com
From: sip:+15125559876@domain.com
Require: org.ietf.sip.reliable-100
Proxy-Require: org.ietf.sip.reliable-100
Content-Type: application/sdp
v=0
o=- 929142225 929142225 IN IP4 vgw.domain.com
c=IN IP4 vgw.domain.com
M=audio 49152 RTP/AVP 0 1
a=rtpmap:0 PCMU/8000
a=rtpmap:1 1016/8000
Server to client:
SIP/2.0 180 Call is being forwarded
RSeq: 1
To: sip:+12145551212@bell.com
From: sip:+15125559876@domain.com
Session: Media
Content-Type: application/sdp
v=0
o=- 929142942 929142942 IN IP4 media.bell.com
c=IN IP4 media.bell.com
M=audio 49180 RTP/AVP 0
a=rtpmap:0 PCMU/8000
Donovan, et al. draft-ietf-sip-183-00.txt
Page 23
Internet Draft SIP 183 Session Progress Message October 1999
Client to server:
INVITE sip:+12145551212@bell.com SIP/2.0
RAck: 1
To: sip:+12145551212@bell.com
From: sip:+15125559876@domain.com
Require: org.ietf.sip.reliable-100
Proxy-Require: org.ietf.sip.reliable-100
[Announcement plays: "Your call is being forwarded to a phone
outside the companys premises. Please wait."]
Server to client:
SIP/2.0 180 Ringing
RSeq: 2
To: sip:+12145551212@bell.com
From: sip:+15125559876@domain.com
Session: Media
Content-Type: application/sdp
v=0
o=- 929143373 929143373 IN IP4 media.bell.com
c=IN IP4 media.bell.com
M=audio 0 RTP/AVP 0
[Media stream is discontinued. Local ring-tone is generated by
the client towards the PSTN user.]
Client to server:
INVITE sip:+12145551212@bell.com SIP/2.0
RAck: 2
To: sip:+12145551212@bell.com
From: sip:+15125559876@domain.com
Require: org.ietf.sip.reliable-100
Proxy-Require: org.ietf.sip.reliable-100
Donovan, et al. draft-ietf-sip-183-00.txt
Page 24
Internet Draft SIP 183 Session Progress Message October 1999
Server to client:
SIP/2.0 200 OK
To: sip:+12145551212@bell.com
From: sip:+15125559876@domain.com
Content-Type: application/sdp
v=0
o=- 929143373 929143373 IN IP4 vgw.bell.com
c=IN IP4 mg.bell.com
M=audio 49199 RTP/AVP 1
a=rtpmap:1 1016/8000
Client to server:
ACK sip:+12145551212@bell.com SIP/2.0
To: sip:+12145551212@bell.com
From: sip:+15125559876@domain.com
[Final media stream is established]
7
[1
] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, SIP: Ses-
sion Initiation Protocol", RFC 2543
, March 1999.
[2
] J. Rosenberg, H. Schulzrinne, S. Donovan, "Establishing QoS and
Security Preconditions for SDP Sessions", draft-rosenberg-mmusic-
sipqos-00.txt
, To be published, Work in Progress.
[3
] J. Rosenberg, H. Schulzrinne, "Reliability of Provisional Responses
in SIP", draft-ietf-mmusic-sip-100rel-01.txt
, May 21, 1999, Work in
Progress.
[4
] H. Schulzrinne, "RTP Profile for Audio and Video Conferences with
Minimal Control ", RFC 1890
, January, 1996.
[5
] M. Handley/V. Jacobson, "SDP: Session Description Protocol", RFC
2327
, April 1998.
[6
] "Application of Tones and Recorded Announcements in Telephone Ser-
vices", ITU-T Recommendation E.182; 1993.
Donovan, et al. draft-ietf-sip-183-00.txt
Page 25
Internet Draft SIP 183 Session Progress Message October 1999
8
Steve Donovan
MCI Worldcom
1493/678
901 International Parkway
Richardson, Texas 75081
Email: steven.r.donovan@wcom.com
Matthew Cannon
BroadBandOffice Inc.
2070 Chain Bridge Road Suite G-99
Vienna VA, 22182
USA
Email: mcannon@broadbandoffice.net
Henning Schulzrinne
Dept. of Computer Science
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
Email: schulzrinne@cs.columbia.edu
Jonathan Rosenberg
Lucent Technologies, Bell Laboratories
Rm. 4C-526
101 Crawfords Corner Road
Holmdel, NJ 07733
USA
Email: jdrosen@bell-labs.com
Adam Roach
Ericsson Inc.
Mailstop L-04
851 International Pkwy.
Richardson, TX 75081
USA
Phone: +1 972-583-7594
Fax: +1 972-669-0154
E-Mail: adam.roach@ericsson.com
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