Internet DRAFT - draft-barnes-sipcore-rfc4244bis-callflows
draft-barnes-sipcore-rfc4244bis-callflows
Network Working Group M. Barnes
Internet-Draft Polycom
Intended status: Informational F. Audet
Expires: September 2, 2012 Skype
S. Schubert
NTT
J. van Elburg
Detecon International Gmbh
C. Holmberg
Ericsson
Mar 2012
Session Initiation Protocol (SIP) History-Info Header Call Flow Examples
draft-barnes-sipcore-rfc4244bis-callflows-03.txt
Abstract
This document describes use cases and documents call flows which
require the History-Info header field to capture the Request-URIs as
a Session Initiation Protocol (SIP) Request is retargeted. The use
cases are described along with the corresponding call flow diagrams
and messaging details.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
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time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on September 2, 2012.
Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
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(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
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Table of Contents
1. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Conventions and Terminology . . . . . . . . . . . . . . . . . 3
3. Detailed call flows . . . . . . . . . . . . . . . . . . . . . 3
3.1. Automatic Call Distribution . . . . . . . . . . . . . . . 3
3.2. Determining the Alias used. . . . . . . . . . . . . . . . 8
3.3. PBX Voicemail Example . . . . . . . . . . . . . . . . . . 10
3.4. Consumer Voicemail Example . . . . . . . . . . . . . . . . 15
3.5. GRUU . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
3.6. Limited Use Address . . . . . . . . . . . . . . . . . . . 22
3.7. Service Invocation . . . . . . . . . . . . . . . . . . . . 24
3.8. Toll Free Number . . . . . . . . . . . . . . . . . . . . . 25
4. Security Considerations . . . . . . . . . . . . . . . . . . . 27
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 27
5.1. Acknowledgements . . . . . . . . . . . . . . . . . . . . . 27
6. Informative References . . . . . . . . . . . . . . . . . . . . 28
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 29
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1. Overview
Many services that use SIP require the ability to determine why and
how the call arrived at a specific application. The use cases
provided in this document illustrate the use of the History-Info
header [I-D.ietf-sipcore-rfc4244bis] for example applications and
common scenarios. The optional "rc" and "mp" header field parameters
defined in [I-D.ietf-sipcore-rfc4244bis] are required for several of
the use cases. Descriptions of the example use cases, call flow
diagrams and messaging details are provided.
2. Conventions and Terminology
The term "retarget" is used as defined in
[I-D.ietf-sipcore-rfc4244bis]. The terms "location service",
"redirect", "redirect" and "AOR" are used consistent with the
terminology in [RFC3261].
3. Detailed call flows
The scenarios in this section provide sample use cases for the
History-Info header for informational purposes only. They are not
intended to be normative. In many cases, only the relevant messaging
details are included in the body of the call flow.
3.1. Automatic Call Distribution
This scenario highlights an example of an Automatic Call Distribution
service, where the agents are divided into groups based upon the type
of customers they handle. In this example, the Gold customers are
given higher priority than Silver customers, so a Gold call would get
serviced even if all the agents servicing the Gold group were busy,
by retargeting the request to the Silver Group for delivery to an
agent. Upon receipt of the call at the agent assigned to handle the
incoming call, based upon the History-Info header in the message, the
application at the agent can provide an indication that this is a
Gold call by extracting the hi-entry associated with the incoming
request which is determined by locating the hi-entry whose index is
reflected in the first hi-entry with an hi-target of "mp". In the
example this would be the hi-entry referenced by the value of the
last "mp" header field parameter -i.e., the hi-entry containing an
index of "1". An application can also determine how many groups from
which the call may have overflowed before reaching the agent, etc.
and present the information to the agent so that the call can be
handled appropriately by the agent - i.e., "I'm so sorry for the
delay, blah, blah, blah..."
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For scenarios whereby calls might overflow from the Silver to the
Gold, clearly the alternate group identification, internal routing,
or actual agent that handles the call should not be sent to UA1.
Thus, for this scenario, one would expect that the Proxy would not
support the sending of the History-Info in the response, even if
requested by Alice.
As with the other examples, this is not a complete prescription of
how one would do this type of service but an example of a subset of
processing that might be associated with such a service. In
addition, this example is not addressing any aspects of Agent
availability resulting in the call being sent to an agent in another
group, which might also be done via a SIP interface.
Alice example.com Gold Silver Agent
| | | | |
| INVITE F1 | | | |
|------------->| | | |
| | | | |
| | INVITE F2 | | |
| |------------->| | |
| | | | |
| | 302 Moved Temporarily F3 | |
| |<-------------| | |
| | | | |
| | INVITE F4 | | |
| |--------------------------->| |
| | | | |
| | | | INVITE F5 |
| | | |----------->|
| | | | |
| | | | 200 OK F6 |
| | | |<-----------|
| | | | |
| | 200 OK F7 | |
| |<---------------------------| |
| | | | |
| 200 OK F8 | | | |
|<-------------| | | |
| | | | |
| ACK | | | |
|------------->| ACK |
| |---------------------------------------->|
F1 INVITE Alice -> Example.com
INVITE sip:Gold@example.com SIP/2.0
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Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=1235
To: John <sip:Gold@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:Gold@example.com>;index=1
Contact: Alice <sip:alice@192.0.2.3>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F2 INVITE Example.com -> Gold.Example.com
INVITE sip:john@192.0.2.1 SIP/2.0
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=1235
To: John <sip:Gold@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:Gold@example.com>;index=1
History-Info: <sip:Gold@gold.example.com>;rc=1;index=1.1
Contact: Alice <sip:alice@192.0.2.3>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F3 302 Moved Temporarily Gold.Example.com -> Example.com
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=1235
To: John <sip:Gold@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:Gold@example.com>;index=1
History-Info: <sip:Gold@gold.example.com>;rc=1;index=1.1
Contact: <sip:Silver@example.com>;mp=1
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
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F4 INVITE Example.com -> Silver.Example.com
INVITE sip:Silver@example.com SIP/2.0
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=1235
To: John <sip:Gold@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:Gold@example.com>;index=1
History-Info: <sip:Gold@gold.example.com?Reason=SIP%3Bcause%3D302>;\
rc=1;index=1.1
History-Info: <sip:Silver@example.com>;index=1.2;mp=1
History-Info: <sip:Silver@silver.example.com>;index=1.2.1;rc=1.2
Contact: Alice <sip:alice@192.0.2.3>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F5 INVITE Silver.Example.com -> Agent
INVITE sip:Silver@192.0.2.7 SIP/2.0
Via: SIP/2.0/TCP silver.example.com:5060;branch=z9hG4bKerxs
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=1235
To: John <sip:Gold@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:Gold@example.com>;index=1
History-Info: <sip:Gold@gold.example.com?Reason=SIP%3Bcause%3D302>;\
index=1.1
History-Info: <sip:Silver@example.com>;index=1.2;mp=1
History-Info: <sip:Silver@silver.example.com>;index=1.2.1;rc=1.2
History-Info: <sip:Silver@192.0.2.7>;index=1.2.1.1;rc=1.2.1
Contact: Alice <sip:alice@192.0.2.3>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F6 200 OK Agent -> Silver.Example.com
SIP/2.0 200 OK
Via: SIP/2.0/TCP silver.example.com:5060;branch=z9hG4bKerxs
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Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=1235
To: John <sip:Gold@example.com>;tag=2325
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:Gold@example.com>;index=1
History-Info: <sip:Gold@gold.example.com?Reason=SIP%3Bcause%3D302>;\
index=1.1
History-Info: <sip:Silver@example.com>;index=1.2;mp=1
History-Info: <sip:Silver@silver.example.com>;index=1.2.1;rc=1.2
History-Info: <sip:Silver@192.0.2.7>;index=1.2.1.1;rc=1.2.1
Contact: Silver <sip:Silver@192.0.2.7>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F7 200 OK Silver.Example.com -> Example.com
SIP/2.0 200 OK
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=1235
To: John <sip:Gold@example.com>;tag=2325
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:Gold@example.com>;index=1
History-Info: <sip:Gold@gold.example.com?Reason=SIP%3Bcause%3D302>;\
index=1.1
History-Info: <sip:Silver@example.com>;index=1.2;mp=1
History-Info: <sip:Silver@silver.example.com>;index=1.2.1;rc=1.2
History-Info: <sip:Silver@192.0.2.7>;index=1.2.1.1;rc=1.2.1
Contact: Silver <sip:Silver@192.0.2.7>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F8 200 OK Example.com -> Alice
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=1235
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To: John <sip:Gold@example.com>;tag=2325
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:Gold@example.com>;index=1
History-Info: <sip:Gold@gold.example.com?Reason=SIP%3Bcause%3D302>;\
index=1.1
History-Info: <sip:Silver@example.com>;index=1.2;mp=1
History-Info: <sip:Silver@silver.example.com>;index=1.2.1;rc=1.2
History-Info: <sip:Silver@192.0.2.7>;index=1.2.1.1;rc=1.2.1
Contact: Silver <sip:Silver@192.0.2.7>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
The last hi-entry with the "mp" header field parameter contains a
"mp" header field parameter value of 1 which points to the original-
target which allows the operator to identify that the call was from
the "Gold" customer.
3.2. Determining the Alias used.
SIP user agents are associated with an address-of-record (AOR). It
is possible for a single UA to actually have multiple AORs associated
with it. One common usage for this is aliases. For example, a user
might have an AOR of sip:john@example.com but also have the AORs
sip:john.smith@example.com and sip:jsmith@example.com. Rather than
registering against each of these AORs individually, the user would
register against just one of them, and the home proxy would
automatically accept incoming calls for any of the aliases, treating
them identically and ultimately forwarding them towards the UA. This
is common practice in the Internet Multimedia Subsystem (IMS), where
it is called implicit registration and each alias is called a public
identity.
It is a common requirement for a UAS, on receipt of a call, to know
which of its aliases was used to reach it. This knowledge can be
used to choose ringtones to play, determine call treatment, and so
on. For example, a user might give out one alias to friends and
family only, resulting in a special ring that alerts the user to the
importance of the call.
The following call-flow and example messages show how History-Info
can be used to find out the alias used to reach the callee. The
alias for the call is determined by hi-entry with the index that
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matches the value of the last hi-entry with a "rc" header field
parameter in the Request received.
Alice Example.com John
| | REGISTER F1 |
| |<--------------------|
| | 200 OK F2 |
| |-------------------->|
| INVITE F3 | |
|-------------------->| |
| | INVITE F4 |
| |-------------------->|
* Rest of flow not shown *
F1 REGISTER John -> Example.com
REGISTER sip:example.com SIP/2.0
Via: SIP/2.0/UDP 192.0.2.1;branch=z9hG4bKnashds7
Max-Forwards: 70
From: John <sip:john@example.com>;tag=a73kszlfl
To: John <sip:john@example.com>
Call-ID: 1j9FpLxk3uxtm8tn@192.0.2.1
CSeq: 1 REGISTER
Contact: <sip:john@192.0.2.1>
Content-Length: 0
F2 200 OK Example.com -> John
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.0.2.1;branch=z9hG4bKnashds7
From: John <sip:john@example.com>;tag=a73kszlfl
To: John <sip:john@example.com>
Call-ID: 1j9FpLxk3uxtm8tn@192.0.2.1
CSeq: 1 REGISTER
Contact: <sip:john@192.0.2.1>;expires=3600
Content-Length: 0
F3 INVITE Alice -> Example.com
INVITE sip:john.smith@example.com SIP/2.0
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=a73kszlfl
To: John <sip:john.smith@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:john.smith@example.com>;index=1
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Contact: Alice <sip:alice@192.0.2.3>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F4 INVITE Example.com -> John
INVITE sip:john@192.0.2.1 SIP/2.0
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=a73kszlfl
To: John <sip:john.smith@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
Record-Route: <sip:proxy.example.com;lr>
History-Info: <sip:john.smith@example.com>;index=1
History-Info: <sip:john@192.0.2.1>;index=1.1;rc=1
Contact: Alice <sip:alice@192.0.2.3>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
Figure 1: Alias Example
The last hi-entry with the "rc" header field parameter references the
source of retargeting pointing at the alias AoR, which in the example
is "john.smith@example.com".
3.3. PBX Voicemail Example
A typical use case for voicemail is one whereby the original called
party is not reachable and the call arrives at a voicemail system.
In some cases multiple alternate destinations may be tried without
success. The voicemail system typically requires the original called
party information to determine the appropriate mailbox so an
appropriate greeting can be provided and the appropriate party
notified of the message.
In this example, Alice calls Bob, whose SIP client is forwarded to
Carol. Carol does not answer the call, thus it is forwarded to a VM
(voicemail) server (VMS). In order to determine the appropriate
mailbox to use for this call, the VMS needs the original target for
the request. The original target is determined by finding the first
hi-entry tagged with "rc" and using the hi-entry referenced by the
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index of "rc" header field parameter as the target for determining
the appropriate mailbox. This hi-entry is used to populate the
"target" URI parameter as defined in [RFC4458]. The reason
associated with the first hi-entry tagged with "rc" (i.e., 302) could
be used to provide a customized voicemail greeting and is used to
populate the "cause" URI parameter as defined in [RFC4458]. Note
that some VMSs may also (or instead) use the information available in
the History-Info headers for custom handling of the VM in terms of
how and why the call arrived at the VMS.
Furthermore it is the proxy forwarding the call to VMS that
determines the target of the voicemail, it is the proxy that sets the
target of voicemail which is also the entity that utilizes RFC4244bis
to find the target which is usually based on local policy installed
by the user or an administrator.
Alice example.com Bob Carol VM
| INVITE F1 | | | |
|------------->| | | |
| | INVITE F2 | | |
| |------------->| | |
| | | | |
| 100 Trying | | | |
|<-------------| 302 Moved Temporarily F3 | |
| |<-------------| | |
| | | | |
| | INVITE F4 | | |
| |--------------------------->| |
| | | | |
| | 180 Ringing F5 | |
| |<---------------------------| |
| | | | |
| 180 Ringing | | | |
|<-------------| | | |
| | | | |
| | (timeout) | |
| | | | |
| | INVITE F6 | | |
| |-------------------------------------->|
| | | | |
| | 200 OK F7 |
| |<--------------------------------------|
| 200 OK | | | |
|<-------------| | | |
| | | | |
| ACK | | | |
|------------->| ACK |
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| |-------------------------------------->|
F1 INVITE Alice -> Example.com
INVITE sip:bob@example.com
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: Bob <sip:bob@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
Contact: Alice <sip:alice@192.0.2.3>
Content-Length: <appropriate value>
[SDP Not Shown]
F2 INVITE Example.com -> Bob
INVITE sip:bob@192.0.2.5 SIP/2.0
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: Bob <sip:bob@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:bob@example.com>;index=1
History-Info: <sip:bob@192.0.2.5>;index=1.1;rc=1
Contact: Alice <sip:alice@192.0.2.3>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F3 302 Moved Temporarily Bob -> Example.com
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: Bob <sip:bob@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:bob@example.com>;index=1
History-Info: <sip:bob@192.0.2.5>; index=1.1;rc=1
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Contact: <sip:carol@example.com>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F4 INVITE Example.com -> Carol
INVITE sip:carol@192.0.2.4 SIP/2.0
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: Bob <sip:bob@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:bob@example.com>;index=1
History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
index=1.1;rc=1
History-Info: <sip:carol@example.com>;index=1.2;mp=1
History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
Contact: Alice <sip:alice@192.0.2.3>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F5 180 Ringing Carol -> Example.com
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: Bob <sip:bob@example.com>;tag=setss3x
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:bob@example.com>;index=1
History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
index=1.1;rc=1
History-Info: <sip:carol@example.com>;index=1.2;mp=1
History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
Contact: <sip:carol@example.com>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
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F6 INVITE Example.com -> VM
INVITE sip:vm@192.0.2.6;target=sip:bob%40example.com;cause=408
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: Bob <sip:bob@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:bob@example.com>;index=1
History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
index=1.1;rc=1
History-Info: <sip:carol@example.com>;index=1.2;mp=1
History-Info: <sip:carol@192.0.2.4?Reason=SIP%3Bcause%3D408>;\
index=1.2.1;rc=1.2
History-Info: <sip:vm@example.com;\
target=sip:bob%40example.com;cause=408>;\
index=1.3;mp=1.2
History-Info: <sip:vm@192.0.2.6;\
target=sip:bob%40example.com;cause=408>;\
index=1.3.1;rc=1.3
Contact: Alice <sip:alice@192.0.2.3>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F7 200 OK VM -> Example.com
SIP/2.0 200 OK
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: Bob <sip:bob@example.com>;tag=3dweggs
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:bob@example.com>;index=1
History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
index=1.1;rc=1
History-Info: <sip:carol@example.com>;index=1.2;mp=1
History-Info: <sip:carol@192.0.2.4?Reason=SIP%3Bcause%3D408>;\
index=1.2.1;rc=1.2
History-Info: <sip:vm@example.com;\
target=sip:bob%40example.com;cause=408>;\
index=1.3;mp=1.2
History-Info: <sip:vm@192.0.2.6;\
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target=sip:bob%40example.com;cause=408>;\
index=1.3.1;rc=1.3
Contact: <sip:carol@example.com>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
The VMS can look at the last hi-entry and find the target of the
mailbox by looking at the URI entry in the "target" URI parameter in
the hi-entry.
3.4. Consumer Voicemail Example
In the case of a consumer, when the call is retargeted, it is usually
to another administrative domain. The voicemail system in these
environment typically requires the last called party information to
determine the appropriate mailbox so an appropriate greeting can be
provided and the appropriate party notified of the message.
In this example, Alice calls the Bob but Bob has temporarily
forwarded his phone to Carol because she is his wife. Carol does not
answer the call, thus it is forwarded to a VM (voicemail) server
(VMS). In order to determine the appropriate mailbox to use for this
call, the VMS needs the appropriate target for the request. The last
target is determined by finding the hi-entry referenced by the index
of last hi-entry tagged with "rc" for determining the appropriate
mailbox. This hi-entry is used to populate the "target" URI
parameter as defined in [RFC4458]. Note that some VMSs may also (or
instead) use the information available in the History-Info headers
for custom handling of the VM in terms of how and why the called
arrived at the VMS.
Alice example.com Bob Carol VM
| INVITE F1 | | | |
|------------->| | | |
| | INVITE F2 | | |
| |------------->| | |
| | | | |
| 100 Trying | | | |
|<-------------| 302 Moved Temporarily F3 | |
| |<-------------| | |
| | | | |
| | INVITE F4 | | |
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| |--------------------------->| |
| | | | |
| | 180 Ringing F5 | |
| |<---------------------------| |
| | | | |
| 180 Ringing | | | |
|<-------------| | | |
| | | | |
| | (timeout) | |
| | | | |
| | INVITE F6 | | |
| |-------------------------------------->|
| | | | |
| | 200 OK F7 |
| |<--------------------------------------|
| 200 OK | | | |
|<-------------| | | |
| | | | |
| ACK | | | |
|------------->| ACK |
| |-------------------------------------->|
F1 INVITE Alice -> Example.com
INVITE sip:bob@example.com
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: Bob <sip:bob@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
Contact: Alice <sip:alice@192.0.2.3>
Content-Length: <appropriate value>
[SDP Not Shown]
F2 INVITE Example.com -> Bob
INVITE sip:bob@192.0.2.5 SIP/2.0
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: Bob <sip:bob@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:bob@example.com>;index=1
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History-Info: <sip:bob@192.0.2.5>;index=1.1;rc=1
Contact: Alice <sip:alice@192.0.2.3>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F3 302 Moved Temporarily Bob -> Example.com
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: Bob <sip:bob@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:bob@example.com>;index=1
History-Info: <sip:bob@192.0.2.5>; index=1.1;rc=1
Contact: <sip:carol@example.com>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F4 INVITE Example.com -> Carol
INVITE sip:carol@192.0.2.4 SIP/2.0
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: Bob <sip:bob@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:bob@example.com>;index=1
History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
index=1.1;rc=1
History-Info: <sip:carol@example.com>;index=1.2;mp=1
History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
Contact: Alice <sip:alice@192.0.2.3>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F5 180 Ringing Carol -> Example.com
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SIP/2.0 180 Ringing
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: Bob <sip:bob@example.com>;tag=setss3x
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:bob@example.com>;index=1
History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
index=1.1;rc=1
History-Info: <sip:carol@example.com>;index=1.2;mp=1
History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
Contact: <sip:carol@example.com>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F6 INVITE Example.com -> VM
INVITE sip:vm@192.0.2.6;target=sip:carol%40example.com
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: Bob <sip:bob@example.com>
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:bob@example.com>;index=1
History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
index=1.1;rc
History-Info: <sip:carol@example.com?Reason=SIP%3Bcause%3D408>;\
index=1.2;mp=1
History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
History-Info: <sip:vm@example.com;target=sip:carol%40example.com>;\
index=1.3;mp=1.2
History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com>;\
index=1.3.1
Contact: Alice <sip:alice@192.0.2.3>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F7 200 OK VM -> Example.com
SIP/2.0 200 OK
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Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: Bob <sip:bob@example.com>;tag=3dweggs
Supported: histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:bob@example.com>;index=1
History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
index=1.1;rc
History-Info: <sip:carol@example.com?Reason=SIP%3Bcause%3D408>;\
index=1.2;mp=1
History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
History-Info: <sip:vm@example.com;target=sip:carol%40example.com>;\
index=1.3;mp=1.2
History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com>;\
index=1.3.1
Contact: <sip:carol@example.com>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
The VMS can look at the last hi-entry and find the target of the
mailbox by looking for the "target" URI parameter in the hi-entry.
3.5. GRUU
A variation on the problem in Section 3.2 occurs with Globally
Routable User Agent URI (GRUU) [RFC5627]. A GRUU is a URI assigned
to a UA instance which has many of the same properties as the AOR,
but causes requests to be routed only to that specific instance. It
is desirable for a UA to know whether it was reached because a
correspondent sent a request to its GRUU or to its AOR. This can be
used to drive differing authorization policies on whether the request
should be accepted or rejected, for example. However, like the AOR
itself, the GRUU is lost in translation at the home proxy. Thus, the
UAS cannot know whether it was contacted via the GRUU or its AOR.
Following call-flow and example messages show how History-Info can be
used to find out the GRUU used to reach the callee.
While a GRUU is comprised of an AoR with a URI parameter as defined
in [RFC5627] , the GRUU construct itself is not an AoR. Thus, the
retargeting of a request based on a GRUU does not result in the
addition of an "rc" header field parameter to the hi-entry containing
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the GRUU. The lack of an "rc" header field parameter in the hi-
entries can be a hint that the source of retargeting is a GRUU.
However, to ensure this is the case, the UAS needs to search for a
"gr" parameter in the hi-entry prior to the last hi-entry. If there
is a GRUU, the URI will always be prior to the last hi-entry as GRUU
does not allow multiple instance to be mapped to a contact address.
Alice Example.com John
| | REGISTER F1 |
| |<--------------------|
| | 200 OK F2 |
| |-------------------->|
| INVITE F3 | |
|-------------------->| |
| | INVITE F4 |
| |-------------------->|
* Rest of flow not shown *
F1 REGISTER John -> Example.com
REGISTER sip:example.com SIP/2.0
Via: SIP/2.0/UDP 192.0.2.1;branch=z9hG4bKnashds7
Max-Forwards: 70
From: John <sip:John@example.com>;tag=a73kszlfl
Supported: gruu
To: John <sip:john@example.com>
Call-ID: 1j9FpLxk3uxtm8tn@192.0.2.1
CSeq: 1 REGISTER
Contact: <sip:john@192.0.2.1>
;+sip.instance="<urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>"
Content-Length: 0
[SDP Not Shown]
F2 200 OK Example.com -> John
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.0.2.1;branch=z9hG4bKnashds7
From: John <sip:john@example.com>;tag=a73kszlfl
To: John <sip:john@example.com> ;tag=b88sn
Call-ID: 1j9FpLxk3uxtm8tn@192.0.2.1
CSeq: 1 REGISTER
Contact: <sip:john@192.0.2.1>
;pub-gruu="sip:john@example.com
;gr=urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6"
;temp-gruu=
"sip:tgruu.7hs==jd7vnzga5w7fajsc7-ajd6fabz0f8g5@example.com;gr"
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;+sip.instance="<urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>"
;expires=3600
Content-Length: 0
[SDP Not Shown]
Assuming Alice has a knowledge of a gruu either through
prior communication or through other means such as presence
places a call to John's gruu.
F3 INVITE Alice -> Example.com
INVITE sip:john@example.com
;gr=urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6 SIP/2.0
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: <sip:john@example.com
;gr=urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>
Supported: gruu, histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info: <sip:john@example.com
;gr=urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>;index=1
Contact: Alice <sip:alice@192.0.2.3>
Content-Length: <appropriate value>
[SDP Not Shown]
F4 INVITE Example.com -> John
INVITE sip:john@192.0.2.1 SIP/2.0
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: John <sip:john@example.com>
Supported: gruu, histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
Record-Route: <sip:proxy.example.com;lr>
History-Info: <sip:john@example.com
;gr=urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>;index=1
History-Info: <sip:john@192.0.2.1>;index=1.1;rc=1
Contact: Alice <sip:alice@192.0.2.3>
Content-Type: application/sdp
Content-Length: <appropriate value>
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[SDP Not Shown]
Figure 2: GRUU Example
By analyzing the entry referenced by the entry with the last "rc",
one can realize that the URI used to reach the device was GRUU by
finding the "gr" parameter.
3.6. Limited Use Address
A limited use address is a SIP URI that is minted on-demand, and
passed out to a small number (usually one) remote correspondent.
Incoming calls targeted to that limited use address are accepted as
long as the UA still desires communications from the remote target.
Should they no longer wish to be bothered by that remote
correspondent, the URI is invalidated so that future requests
targeted to it are rejected.
Limited use addresses are used in battling voice spam [RFC5039]. The
easiest way to provide them would be for a UA to be able to take its
AOR, and "mint" a limited use address by appending additional
parameters to the URI. It could then give out the URI to a
particular correspondent, and remember that URI locally. When an
incoming call arrives, the UAS would examine the parameter in the URI
and determine whether or not the call should be accepted.
Alternatively, the UA could push authorization rules into the
network, so that it need not even see incoming requests that are to
be rejected.
This approach, especially when executed on the UA, requires that
parameters attached to the AOR, but not used by the home proxy in
processing the request, will survive the translation at the home
proxy and be presented to the UA. This will not be the case with the
logic in RFC 3261, since the Request-URI is replaced by the
registered contact, and any such parameters are lost.
Using the history-info John's UA can easily see if the call was
addressed to its AoR, GRUU or a temp-gruu and treat the call
accordingly by looking for a "gr" tag in the hi-entry prior to the
last hi-entry.
Alice Example.com John
| | REGISTER F1 |
| |<--------------------|
| | 200 OK F2 |
| |-------------------->|
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| INVITE F3 | |
|-------------------->| |
| | INVITE F4 |
| |-------------------->|
* Rest of flow not shown *
F1 REGISTER John -> Example.com
REGISTER sip:example.com SIP/2.0
Via: SIP/2.0/UDP 192.0.2.1;branch=z9hG4bKnashds7
Max-Forwards: 70
From: John <sip:John@example.com>;tag=a73kszlfl
Supported: gruu
To: John <sip:john@example.com>
Call-ID: 1j9FpLxk3uxtm8tn@192.0.2.1
CSeq: 1 REGISTER
Contact: <sip:john@192.0.2.1>
;+sip.instance="<urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>"
Content-Length: 0
F2 200 OK Example.com -> John
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.0.2.1;branch=z9hG4bKnashds7
From: John <sip:john@example.com>;tag=a73kszlfl
To: John <sip:john@example.com> ;tag=b88sn
Call-ID: 1j9FpLxk3uxtm8tn@192.0.2.1
CSeq: 1 REGISTER
Contact: <sip:john@192.0.2.1>
;pub-gruu="sip:john@example.com
;gr=urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6"
;temp-gruu=
"sip:tgruu.7hs==jd7vnzga5w7fajsc7-ajd6fabz0f8g5@example.com;gr"
;+sip.instance="<urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>"
;expires=3600
Content-Length: 0
Assuming Alice has a knowledge of a temp-gruu, she places a
call to the temp-gruu.
F3 INVITE Alice -> Example.com
INVITE sip:tgruu.7hs==jd7vnzga5w7fajsc7-ajd6fabz0f8g5@example.com
;gr SIP/2.0
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: <sip:sip:tgruu.7hs==jd7vnzga5w7fajsc7-ajd6fabz0f8g5@example.com
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;gr>
Supported: gruu, histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
History-Info:
<sip:tgruu.7hs==jd7vnzga5w7fajsc7-ajd6fabz0f8g5@example.com;gr>
;index=1
Contact: Alice <sip:alice@192.0.2.3>
Content-Length: <appropriate value>
F4 INVITE Example.com -> John
INVITE sip:john@192.0.2.1 SIP/2.0
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bK12s4
Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
From: Alice <sip:alice@example.com>;tag=kkaz-
To: John <sip:john@example.com>
Supported: gruu, histinfo
Call-Id: 12345600@example.com
CSeq: 1 INVITE
Record-Route: <sip:proxy.example.com;lr>
History-Info:
<sip:tgruu.7hs==jd7vnzga5w7fajsc7-ajd6fabz0f8g5@example.com;gr>
;index=1
History-Info: <sip:john@192.0.2.1>;index=1.1;rc=1
Contact: Alice <sip:alice@192.0.2.3>
Content-Type: application/sdp
Content-Length: <appropriate value>
Figure 3: Limited Use Address Example
By analyzing the entry referenced by the entry with the last "rc",
one can realize that the URI used to reach the device was GRUU by
finding the "gr" parameter.
3.7. Service Invocation
Several SIP specifications have been developed which make use of
complex URIs to address services within the network rather than
subscribers. The URIs are complex because they contain numerous
parameters that control the behavior of the service. Examples of
this include the specification which first introduced the concept,
[RFC3087], control of network announcements and IVR with SIP URI
[RFC4240], and control of voicemail access with SIP URI [RFC4458].
A common problem with all of these mechanisms is that once a proxy
has decided to rewrite the Request-URI to point to the service, it
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cannot be sure that the Request-URI will not be destroyed by a
downstream proxy which decides to forward the request in some way,
and does so by rewriting the Request-URI.
Section on voicemail (Section 3.3) shows how History-Info can be used
to invocate a service.
3.8. Toll Free Number
Toll free numbers, also known as 800 or 8xx numbers in the United
States, are telephone numbers that are free for users to call.
In the telephone network, toll free numbers are just aliases to
actual numbers which are used for routing of the call. In order to
process the call in the PSTN, a switch will perform a query (using a
protocol called TCAP), which will return either a phone number or the
identity of a carrier which can handle the call.
There has been recent work on allowing such PSTN translation services
to be accessed by SIP proxy servers through IP querying mechanisms.
ENUM, for example [RFC3761] has already been proposed as a mechanism
for performing Local Number Portability (LNP) queries [RFC4769], and
recently been proposed for performing calling name queries
[I-D.ietf-enum-cnam]. Using it for 8xx number translations is a
logical next-step.
Once such a translation has been performed, the call needs to be
routed towards the target of the request. Normally, this would
happen by selecting a PSTN gateway which is a good route towards the
translated number. However, one can imagine all-IP systems where the
8xx numbers are SIP endpoints on an IP network, in which case the
translation of the 8xx number would actually be a SIP URI and not a
phone number. Assuming for the moment it is a PSTN connected entity,
the call would be routed towards a PSTN gateway. Proper treatment of
the call in the PSTN (and in particular, correct reconciliation of
billing records) requires that the call be marked with both the
original 8xx number AND the target number for the call. However, in
our example here, since the translation was performed by a SIP proxy
upstream from the gateway, the original 8xx number would have been
lost, and the call will not interwork properly with the PSTN.
Furthermore, even if the translation of the 8xx number was a SIP URI,
the enterprise or user who utilize the 8xx service would like to know
whether the call came in via 8xx number in order to treat the call
differently (for example to play a special announcement..) but if the
original R-URI is lost through translation, there is no way to tell
if the call came in via 8xx number.
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Similar problems arise with other "special" numbers and services used
in the PSTN, such as operator services, pay/premium numbers (9xx
numbers in the U.S), and short service codes such as 311.
To find the service number, the UAS can extract the hi-entry whose
index matches the value of the first hi-entry with an "mp" tag.
Technically the call can be forwarded to these "special" numbers from
non "special" numbers, however that is uncommon based on the way
these services authorize translations.
Alice Toll Free Service Atlanta.com John
| | | |
| INVITE F1 | | |
|--------------->| INVITE F2 | |
| |------------->| |
| | | INVITE F3 |
| | |------------------>|
* Rest of flow not shown *
F1: INVITE 192.0.2.1 -> Toll Free Service
INVITE sip:+18005551002@example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK74bf
From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
To: sip:+18005551002@example.com;user=phone
Call-ID: c3x842276298220188511
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:alice@192.0.2.1>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F2: INVITE Toll Free Service -> Atlanta.com
INVITE sip:+15555551002@atlanta.com SIP/2.0
Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik8
Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK74bf
From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
To: sip:+18005551002@example.com;user=phone
Call-ID: c3x842276298220188511
CSeq: 1 INVITE
Max-Forwards: 70
Supported: histinfo
History-Info: <sip:+18005551002@example.com;user=phone>;index=1,
<sip:+15555551002@atlanta.com>;index=1.1;mp=1
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Contact: <sip:alice@192.0.2.1>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
F3: INVITE Atlanta.com -> John
INVITE sip:john@198.51.100.2 SIP/2.0
Via: SIP/2.0/TCP 198.51.100.1:5060;branch=z9hG4bKpxk7g
Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80
Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK74bf
From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
To: sip:+18005551002@example.com;user=phone
Call-ID: c3x842276298220188511
CSeq: 1 INVITE
Max-Forwards: 70
Supported: histinfo
History-Info: <sip:+18005551002@example.com;user=phone>;index=1,
<sip:+15555551002@atlanta.com>;index=1.1;mp=1,
<sip:john@atlanta.com>;index=1.1.1;rc=1.1
<sip:john@198.51.100.2>;index=1.1.2;rc=1.1
Contact: <sip:alice@192.0.2.1>
Content-Type: application/sdp
Content-Length: <appropriate value>
[SDP Not Shown]
Figure 4: Service Number Example
4. Security Considerations
The security considerations for the History-Info header field are
specified in [I-D.ietf-sipcore-rfc4244bis].
5. IANA Considerations
This document has no IANA considerations.
5.1. Acknowledgements
Jonathan Rosenberg et al produced the document that provided
additional use cases precipitating the requirement for the new
"target" parameter in the History-Info header field and the new SIP/
SIPS URI parameter. Hadriel Kaplan provided some comments.
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6. Informative References
[RFC3261] 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.
[RFC3326] Schulzrinne, H., Oran, D., and G. Camarillo, "The Reason
Header Field for the Session Initiation Protocol (SIP)",
RFC 3326, December 2002.
[RFC3323] Peterson, J., "A Privacy Mechanism for the Session
Initiation Protocol (SIP)", RFC 3323, November 2002.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246, August 2008.
[RFC4244] Barnes, M., "An Extension to the Session Initiation
Protocol (SIP) for Request History Information", RFC 4244,
November 2005.
[RFC5627] Rosenberg, J., "Obtaining and Using Globally Routable User
Agent URIs (GRUUs) in the Session Initiation Protocol
(SIP)", RFC 5627, October 2009.
[RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session
Initiation Protocol (SIP)", RFC 5630, October 2009.
[RFC3087] Campbell, B. and R. Sparks, "Control of Service Context
using SIP Request-URI", RFC 3087, April 2001.
[RFC4240] Burger, E., Van Dyke, J., and A. Spitzer, "Basic Network
Media Services with SIP", RFC 4240, December 2005.
[RFC5039] Rosenberg, J. and C. Jennings, "The Session Initiation
Protocol (SIP) and Spam", RFC 5039, January 2008.
[RFC4458] Jennings, C., Audet, F., and J. Elwell, "Session
Initiation Protocol (SIP) URIs for Applications such as
Voicemail and Interactive Voice Response (IVR)", RFC 4458,
April 2006.
[RFC3761] Faltstrom, P. and M. Mealling, "The E.164 to Uniform
Resource Identifiers (URI) Dynamic Delegation Discovery
System (DDDS) Application (ENUM)", RFC 3761, April 2004.
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[RFC4769] Livingood, J. and R. Shockey, "IANA Registration for an
Enumservice Containing Public Switched Telephone Network
(PSTN) Signaling Information", RFC 4769, November 2006.
[RFC3969] Camarillo, G., "The Internet Assigned Number Authority
(IANA) Uniform Resource Identifier (URI) Parameter
Registry for the Session Initiation Protocol (SIP)",
BCP 99, RFC 3969, December 2004.
[I-D.ietf-enum-cnam]
Shockey, R., "IANA Registration for an Enumservice Calling
Name Delivery (CNAM) Information and IANA Registration for
URI type 'pstndata'", draft-ietf-enum-cnam-08 (work in
progress), September 2008.
[I-D.ietf-sipcore-rfc4244bis]
Holmberg, C., Audet, F., Barnes, M., Elburg, H., and S.
Schubert, "An Extension to the Session Initiation Protocol
(SIP) for Request History Information",
draft-ietf-sipcore-rfc4244bis-06 (work in progress),
October 2011.
Authors' Addresses
Mary Barnes
Polycom
TX
US
Email: mary.ietf.barnes@gmail.com
Francois Audet
Skype
Email: francois.audet@skype.net
Shida Schubert
NTT
Tokyo
Japan
Email: shida@ntt-at.com
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Hans Erik van Elburg
Detecon International Gmbh
Oberkasseler str. 2
Bonn,
Germany
Email: ietf.hanserik@gmail.com
Christer Holmberg
Ericsson
Hirsalantie 11, Jorvas
Finland
Email: christer.holmberg@ericsson.com
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