draft-donovan-mmusic-183
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Internet Engineering Task Force Steve Donovan
Internet Draft John Hearty
draft-donovan-mmusic-183-00.txt Matt Cannon
MCI Worldcom
Henning Schulzrinne
Columbia University
June, 1999 Jonathan Rosenberg
Expires: December, 1999 Bell Laboratories
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 of the Session Initia-
tion Protocol. This extension would add the 183 Session Progress
response 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
and should 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.
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1 Introduction
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 would be 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 net-
works 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.
2 PSTN Interworking Issues
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.
Donovan, et al. draft-donovan-mmusic-183-00.txt Page 2
Internet Draft SIP 183 Session Progress Message June 1999
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.
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
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2.1 PSTN to SIP Network Interworking Requirements
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.
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 |
+---------+ +-------------+
3.0 Options With Existing SIP Specification
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
Donovan, et al. draft-donovan-mmusic-183-00.txt Page 4
Internet Draft SIP 183 Session Progress Message June 1999
better addressed using the 183 Session Progress message.
3.1 100 Trying Mapped to ACM
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
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, causing the ACM
message to be sent prior to the egress gateway receiving an ACM from
the terminating network.
3.2 180 Ringing Mapped to ACM
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:
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Internet Draft SIP 183 Session Progress Message June 1999
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.
3.3 180 With SDP Mapped to ACM
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:
Donovan, et al. draft-donovan-mmusic-183-00.txt Page 6
Internet Draft SIP 183 Session Progress Message June 1999
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.
3.4 200 OK Mapped to ACM
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:
Donovan, et al. draft-donovan-mmusic-183-00.txt Page 7
Internet Draft SIP 183 Session Progress Message June 1999
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
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Although this will work in the above scenario, it introduces additional
messaging overhead. In addition, as illustrated in the following fast
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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Internet Draft SIP 183 Session Progress Message June 1999
4 Proposed 183 Session Progress
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.
4.1 PSTN to SIP to PSTN Session Using 183 Session Progress
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
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4.2 PSTN Fast Answer
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
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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4.3 SIP to PSTN Session Using 183 Session Progress
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
------------>------------>
200 OK RLC
5 Proposed Extensions to the SIP Specification
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.
5.7 Status Code Definitions
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Internet Draft SIP 183 Session Progress Message June 1999
5.7.1 Informational 1xx
5.7.1.2 180 Ringing
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.
5.7.1.5 183 Session Progress
The called UA has the need to communicate the status of the session
setup attempt as part of a media stream. The calling UA shall estab-
lish a media session according 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 SIP Message Body
5.8.1 Body Inclusion
The following is proposed rewording of paragraph 2 in Section 8.1 of
the SIP specification:
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 ...
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5.10 Behavior of SIP Clients and Servers
5.10.1.2 Responses
The following is proposed text for inclusion in section 10.1.2 of the
SIP specification:
183 responses SHALL always be forwarded.
5.11 Behavior of SIP User Agents
5.11.6. Callee Needs Early Media
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 Caller Receives 183 Response
When the calling UA receives a 183 response that contains a session
description it SHALL setup the associated media session and present
any media received from the called UA to the user.
5.13 Security Considerations
The security considerations for the 183 Session Progress message are
the same as for SIP in general.
5.16 Examples
5.16.9 PSTN to PSTN Session Setup (SIP in the middle)
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.
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Internet Draft SIP 183 Session Progress Message June 1999
Originating Ingress Egress Terminating
Network SIP GW SIP GW Network
------------>------------>----------->
IAM INVITE IAM
<------------
100 Trying
<-----------<------------<------------
ACM 183 Session ACM
Progress
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 SIP User Agent Session Setup to a PSTN Destination
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
------------>------------>
200 OK RLC
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
works then it MUST support the 183 Session Progress message.
6 References
[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.
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Internet Draft SIP 183 Session Progress Message June 1999
[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.
6 Authors' Addresses
Steve Donovan
MCI Worldcom
1493/678
901 International Parkway
Richardson, Texas 75081
Email: steven.r.donovan@mci.com
John Hearty
MCI Worldcom
9514/107
2400 North Glenville Drive
Richardson, TX 75082
Email: john.h.hearty@mci.com
Mathew Cannon
MCI Worldcom
9514/107
2400 North Glenville Drive
Richardson, TX 75082
Email: matt.cannon@wcom.com
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
Donovan, et al. draft-donovan-mmusic-183-00.txt Page 17