Network Working Group M. Barnes Internet-Draft Polycom Intended status: Informational F. Audet Expires: December 26, 2010 Skype S. Schubert NTT J. van Elburg Detecon International Gmbh C. Holmberg Ericsson June 24, 2010 Session Initiation Protocol (SIP) History-Info Header Call Flow Examples draft-barnes-sipcore-rfc4244bis-callflows-00.txt Abstract This document describes use cases and documents call flows which require the History-Info header 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 working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. 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 material or to cite them other than as "work in progress." This Internet-Draft will expire on December 26, 2010. Copyright Notice Copyright (c) 2010 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 Barnes, et al. Expires December 26, 2010 [Page 1] Internet-Draft History-Info Call Flows June 2010 (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. 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. . . . . . . . . . . . . . . . 5 3.3. PBX Voicemail Example . . . . . . . . . . . . . . . . . . 7 3.4. Call Center Voicemail Example . . . . . . . . . . . . . . 9 3.5. GRUU . . . . . . . . . . . . . . . . . . . . . . . . . . . 11 3.6. Limited Use Address . . . . . . . . . . . . . . . . . . . 14 3.7. Sub-Address . . . . . . . . . . . . . . . . . . . . . . . 16 3.8. Service Invocation . . . . . . . . . . . . . . . . . . . . 20 3.9. Toll Free Number . . . . . . . . . . . . . . . . . . . . . 20 4. Security Considerations . . . . . . . . . . . . . . . . . . . 23 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 23 5.1. Acknowledgements . . . . . . . . . . . . . . . . . . . . . 23 6. Informative References . . . . . . . . . . . . . . . . . . . . 23 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 24 Barnes, et al. Expires December 26, 2010 [Page 2] Internet-Draft History-Info Call Flows June 2010 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 Target parameter along with the "hit" SIP/SIPS URI 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 terms "retarget" and "forward" are used as defined in [I-D.ietf-sipcore-rfc4244bis]. The terms "location service", "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". 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..." For scenarios whereby calls might overflow from the Silver to the Barnes, et al. Expires December 26, 2010 [Page 3] Internet-Draft History-Info Call Flows June 2010 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 sip:Gold@example.com | | | |------------->| | | | | Supported: histinfo | | | | | | | INVITE sip:Gold@example.com | | |------------->| | | History-Info: ;index=1 History-Info: ;index=1.1 | | | | | | | 302 Moved Temporarily | | | |<-------------| | | History-Info: ;index=1 History-Info: ;\ index=1.1 Contact: | | | | | | INVITE sip:Silver@example.com | | |--------------------------->| | History-Info: ;index=1 History-Info: ;\ index=1.1 History-Info: ;index=2;mp=1 History-Info: ;index=2.1 | | | | | | | | INVITE sip:Silver@192.0.2.7 | | | |----------->| History-Info: ;index=1 History-Info: ;\ index=1.1 History-Info: ;index=2;mp=1 History-Info: ;index=2.1 History-Info: ;index=2.1.1;rc Barnes, et al. Expires December 26, 2010 [Page 4] Internet-Draft History-Info Call Flows June 2010 | | | | | | | | | 200 OK | | | | |<-----------| History-Info: ;index=1 History-Info: ;\ index=1.1 History-Info: ;index=2;mp=1 History-Info: ;index=2.1 History-Info: ;index=2.1.1;rc | | | | | | | 200 OK | | | |<---------------------------| | History-Info: ;index=1 History-Info: ;\ index=1.1 History-Info: ;index=2;mp=1 History-Info: ;index=2.1 History-Info: ;index=2.1.1;rc | | | | | 200 OK | | | | |<-------------| | | | | | | | | | ACK | | | | |------------->| ACK | | |---------------------------------------->| 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. Barnes, et al. Expires December 26, 2010 [Page 5] Internet-Draft History-Info Call Flows June 2010 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 the UAS extracting the hi-entry prior to the last hi-entry with the "rc" tag. 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 ;tag=a73kszlfl To: John Call-ID: 1j9FpLxk3uxtm8tn@192.0.2.1 CSeq: 1 REGISTER Contact: 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 ;tag=a73kszlfl To: John Call-ID: 1j9FpLxk3uxtm8tn@192.0.2.1 CSeq: 1 REGISTER Contact: ;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=232sxxeserg From: Alice To: John Supported: histinfo Call-Id: 12345600@example.com Barnes, et al. Expires December 26, 2010 [Page 6] Internet-Draft History-Info Call Flows June 2010 CSeq: 1 INVITE History-Info: ;index=1; Contact: Alice Content-Type: application/sdp Content-Length: [SDP Not Shown] F4 INVITE Example.com -> Bob INVITE sip:john@192.0.2.1 SIP/2.0 Via: SIP/2.0/TCP proxy.example.com:5060;branch=as2334se Via: SIP/2.0/TCP 192.0.2.3:5060;branch=232sxxeserg From: Alice To: John Supported: histinfo Call-Id: 12345600@example.com CSeq: 1 INVITE Record-Route: History-Info: ;index=1; History-Info: ;index=1.1;rc Contact: Alice Content-Type: application/sdp Content-Length: [SDP Not Shown] Figure 1: Alias Example 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 just prior as the target for determining the appropriate mailbox. This hi-entry is used to populate the "target" URI parameter as defined in [RFC4458]. Barnes, et al. Expires December 26, 2010 [Page 7] Internet-Draft History-Info Call Flows June 2010 The reason associated with the first 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 called arrived at the VMS. Alice example.com Bob Carol VM | INVITE sip:bob@example.com | | | |------------->| | | | | | INVITE sip:bob@192.0.2.3 | | | |------------->| | | History-Info: ;index=1 History-Info: ;index=1.1;rc | | | | | | 100 Trying | | | | |<-------------| 302 Moved Temporarily | | | |<-------------| | | History-Info: ;index=1 History-Info: ; index=1.1;rc Contact: | | | | | | | INVITE sip:Carol@192.0.2.4 | | | |--------------------------->| | History-Info: ;index=1 History-Info: ;\ index=1.1;rc History-Info: ;index=1.2;mp=1 History-Info: ;index=1.2.1;rc | | | | | | | 180 Ringing | | | |<---------------------------| | History-Info: ;index=1 History-Info: ;\ index=1.1;rc History-Info: ;index=1.2;mp=1 History-Info: ;index=1.2.1;rc | | | | | | 180 Ringing | | | | |<-------------| | | | | | | | | | . . . | | | | | | (timeout) | | | | | | | | | INVITE sip:vm@192.0.2.5;\ | | target=sip:bob@example.com>;\ | | cause=302 Barnes, et al. Expires December 26, 2010 [Page 8] Internet-Draft History-Info Call Flows June 2010 | |-------------------------------------->| History-Info: ;index=1 History-Info: ;\ index=1.1;rc History-Info: ;index=1.2;mp=1 History-Info: ;index=1.2.1;rc History-Info: ;\ target=sip:bob@example.com;cause=302>\ index=1.3;mp=1.2 History-Info: ;\ target=sip:bob@example.com;cause=302>\ index=1.3.1 | | | | | | | 200 OK | | |<--------------------------------------| History-Info: ;index=1 History-Info: ;\ index=1.1;rc History-Info: ;index=1.2;mp=1 History-Info: ;index=1.2.1;rc History-Info: ;\ target=sip:bob@example.com;cause=302>\ index=1.3;mp=1.2 History-Info: ; target=sip:bob@example.com;cause=302>\ index=1.3.1 | 200 OK | | | | |<-------------| | | | | | | | | | ACK | | | | |------------->| ACK | | |-------------------------------------->| 3.4. Call Center Voicemail Example In the case a call centers, where the original called number does not necessarily represent a specific individual, but rather is associated with a set of individuals, any of whom would handle an incoming call in much the same manner (e.g., insurance company representatives). 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 the direct number for her insurance agent. Bob has temporarily forwarded his phone to Carol because she Barnes, et al. Expires December 26, 2010 [Page 9] Internet-Draft History-Info Call Flows June 2010 is the co-worker assigned as his backup. 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 original target is determined by finding the last hi-entry tagged with "rc" and using the hi-entry just prior as the target 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 sip:bob@example.com | | | |------------->| | | | | | INVITE sip:bob@192.0.2.3 | | | |------------->| | | History-Info: ;index=1 History-Info: ;index=1.1;rc | | | | | | 100 Trying | | | | |<-------------| 302 Moved Temporarily | | | |<-------------| | | History-Info: ;index=1 History-Info: ; index=1.1;rc Contact: | | | | | | | INVITE sip:Carol@192.0.2.4 | | | |--------------------------->| | History-Info: ;index=1 History-Info: ;\ index=1.1;rc History-Info: ;index=1.2;mp=1 History-Info: ;index=1.2.1;rc | | | | | | | 180 Ringing | | | |<---------------------------| | History-Info: ;index=1 History-Info: ;\ index=1.1;rc History-Info: ;index=1.2;mp=1 History-Info: ;index=1.2.1;rc | | | | | | 180 Ringing | | | | |<-------------| | | | | | | | | | . . . | | | | Barnes, et al. Expires December 26, 2010 [Page 10] Internet-Draft History-Info Call Flows June 2010 | | (timeout) | | | | | | | | | INVITE sip:vm@192.0.2.5;\ | | target=sip:carol@example.com | |-------------------------------------->| History-Info: ;index=1 History-Info: ;\ index=1.1;rc History-Info: ;index=1.2;mp=1 History-Info: ;index=1.2.1;rc History-Info: ;\ index=1.3;mp=1.2 History-Info: ;\ index=1.3.1 | | | | | | | 200 OK | | |<--------------------------------------| History-Info: ;index=1 History-Info: ;\ index=1.1;rc History-Info: ;index=1.2;mp=1 History-Info: ;index=1.2.1;rc History-Info: ;\ index=1.3;mp=1.2 History-Info: ; target=sip:carol@example.com>;\ index=1.3.1 | 200 OK | | | | |<-------------| | | | | | | | | | ACK | | | | |------------->| ACK | | |-------------------------------------->| 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 Barnes, et al. Expires December 26, 2010 [Page 11] Internet-Draft History-Info Call Flows June 2010 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. GRUU is merely an AoR with a URI parameter that distinguishes the target instance, and as any URI parameters are preserved in history- info as Request-URI is trasnlated, UA can see if the request was addressed to a specific instance (gruu) by evaluating the presence of "gr" parameter in the hi-entry prior to the last hi-entry with the "rc" tag. 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 ;tag=a73kszlfl Supported: gruu To: John Call-ID: 1j9FpLxk3uxtm8tn@192.0.2.1 CSeq: 1 REGISTER Contact: ;+sip.instance="" 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 ;tag=a73kszlfl To: John ;tag=b88sn Call-ID: 1j9FpLxk3uxtm8tn@192.0.2.1 CSeq: 1 REGISTER Contact: ;pub-gruu="sip:john@example.com Barnes, et al. Expires December 26, 2010 [Page 12] Internet-Draft History-Info Call Flows June 2010 ;gr=urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6" ;temp-gruu= "sip:tgruu.7hs==jd7vnzga5w7fajsc7-ajd6fabz0f8g5@example.com;gr" ;+sip.instance="" ;expires=3600 Content-Length: 0 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=232sxxeserg From: Alice ;tag=kkaz- To: Supported: gruu, histinfo Call-Id: 12345600@example.com CSeq: 1 INVITE History-Info: ;index=1 Contact: Alice Content-Length: 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=as2334se Via: SIP/2.0/TCP 192.0.2.3:5060;branch=232sxxeserg From: Alice ;tag=kkaz- To: John Supported: gruu, histinfo Call-Id: 12345600@example.com CSeq: 1 INVITE Record-Route: History-Info: ;index=1 History-Info: ;index=1.1;rc Contact: Alice Content-Type: application/sdp Content-Length: Figure 2: GRUU Example Barnes, et al. Expires December 26, 2010 [Page 13] Internet-Draft History-Info Call Flows June 2010 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 at the hi-entry prior to the last hi-entry with the "rc" tag. 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 Barnes, et al. Expires December 26, 2010 [Page 14] Internet-Draft History-Info Call Flows June 2010 Via: SIP/2.0/UDP 192.0.2.1;branch=z9hG4bKnashds7 Max-Forwards: 70 From: John ;tag=a73kszlfl Supported: gruu To: John Call-ID: 1j9FpLxk3uxtm8tn@192.0.2.1 CSeq: 1 REGISTER Contact: ;+sip.instance="" 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 ;tag=a73kszlfl To: John ;tag=b88sn Call-ID: 1j9FpLxk3uxtm8tn@192.0.2.1 CSeq: 1 REGISTER Contact: ;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="" ;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=232sxxeserg From: Alice ;tag=kkaz- To: Supported: gruu, histinfo Call-Id: 12345600@example.com CSeq: 1 INVITE History-Info: ;index=1 Contact: Alice Content-Length: Barnes, et al. Expires December 26, 2010 [Page 15] Internet-Draft History-Info Call Flows June 2010 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=as2334se Via: SIP/2.0/TCP 192.0.2.3:5060;branch=232sxxeserg From: Alice ;tag=kkaz- To: John Supported: gruu, histinfo Call-Id: 12345600@example.com CSeq: 1 INVITE Record-Route: History-Info: ;index=1 History-Info: ;index=1.1;rc Contact: Alice Content-Type: application/sdp Content-Length: Figure 3: Limited Use Address Example 3.7. Sub-Address Sub-Addressing is very similar to limited use addresses. Sub- addresses are addresses within a subdomain that are multiplexed into a single address within a parent domain. The concept is best illustrated by example. Consider a VoIP service provided to consumers. A consumer obtains a single address from its provider, say sip:family@example.com. However, Joe is the patriarch of a family with four members, and would like to be able to have a separate identifier for each member of his family. One way to do that, without requiring Joe to purchase new addresses for each member from the provider, is for Joe to mint additional URI by adding a parameter to the AOR. For example, his wife Judy with have the URI sip:family@example.com;member=judy, and Joe himself would have the URI sip:family@example.com;member=joe. The SIP server provider would receive requests to these URI, and ignoring the unknown parameters (as required by [RFC3261]) route the request to the registered contact, which corresponds to a SIP server in Joes home. That server, in turn, can examine the URI parameters and determine which phone in the home to route the call to. This feature is not specific to VoIP, and has existing in Integrated Services Digital Networking (ISDN) for some time. It is particularly useful for small enterprises, in addition to families. It is also similar in spirit (though not mechanism) to the ubiquitous home routers used by consumers, which allow multiple computers in the home Barnes, et al. Expires December 26, 2010 [Page 16] Internet-Draft History-Info Call Flows June 2010 to "hide" behind the single IP address provided by the service provider, by using the TCP and UDP port as a sub-address. The sub-addressing feature is not currently feasible in SIP because of the fact that any SIP URI parameter used to convey the sub-address would be lost at the home proxy, due to the fact that the Request-URI is rewritten there. Call-flow and example messages below show the how History-Info can be used to deliver the sub-address. UAS or Proxy can determine the sub- address by looking at the hi-entry prior to the last hi-entry with the "rc" tag. Alice Example.com John's Home Judy John | | REGISTER F1 | | | | |<-------------| | | | | 200 OK F2 | | | | |------------->| | | | | | REGISTER F3 | | | | |<-------------| | | | | 200 OK F4 | | | | |------------->| | | | | | REGISTER F5 | | | |<----------------------------| | | | | 200 OK F6 | | | |---------------------------->| | INVITE F7 | | | | |----------->| | | | | | INVITE F8 | | | | |------------->| | | | | | INVITE F9 | | | | |------------->| | * Rest of flow not shown * F1 REGISTER John's Home Server -> 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 ;tag=a73kszlfl To: John Call-ID: 1j9FpLxk3uxtm8tn@192.0.2.1 CSeq: 1 REGISTER Contact: Content-Length: 0 F2 200 OK Example.com -> John's Home Server Barnes, et al. Expires December 26, 2010 [Page 17] Internet-Draft History-Info Call Flows June 2010 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.0.2.1;branch=z9hG4bKnashds7 From: John ;tag=a73kszlfl To: John ;tag=b88sn Call-ID: 1j9FpLxk3uxtm8tn@192.0.2.1 CSeq: 1 REGISTER Contact: ;expires=3600 Content-Length: 0 We assume that John's server acts as a proxy allowing each of the device in the house to register. F3 REGISTER Judy's phone -> John's Home Server REGISTER sip:198.51.100.1 SIP/2.0 Via: SIP/2.0/UDP 198.51.100.2;branch=z9hG4bKnasdds Max-Forwards: 70 From: Judy ;tag=a73kszlfl To: Judy Call-ID: 12345pLxk3uxtm8tn@198.51.100.2 CSeq: 1 REGISTER Contact: Content-Length: 0 F4 200 OK John's Home Server -> Judy's phone SIP/2.0 200 OK Via: SIP/2.0/UDP 198.51.100.2;branch=z9hG4bKnashds7 From: Judy ;tag=a73kszlfl To: Judy tag=b88sn Call-ID: 12345pLxk3uxtm8tn@198.51.100.2 CSeq: 1 REGISTER Contact: ;expires=3600 Content-Length: 0 F5 REGISTER John's phone -> John's Home Server REGISTER sip:198.52.100.1 SIP/2.0 Via: SIP/2.0/UDP 198.52.100.3;branch=z9hG4bKnasdds Max-Forwards: 70 From: Judy ;tag=a73kszlfl To: Judy Call-ID: 12346pLxk3uxtm8tn@198.51.100.3 CSeq: 1 REGISTER Contact: Content-Length: 0 F6 200 OK John's Home Server -> John's phone Barnes, et al. Expires December 26, 2010 [Page 18] Internet-Draft History-Info Call Flows June 2010 SIP/2.0 200 OK Via: SIP/2.0/UDP 198.51.100.3;branch=z9hG4bKnashds7 From: John ;tag=a73kszlfl To: John ;tag=b88sn Call-ID: 12346pLxk3uxtm8tn@198.51.100.3 CSeq: 1 REGISTER Contact: ;expires=3600 Content-Length: 0 F7 INVITE Alice -> Example.com INVITE sip:johnhome@example.com;member=judy SIP/2.0 Via: SIP/2.0/TCP 192.0.2.3:5060;branch=232sxxeserg From: Alice To: Judy Supported: histinfo Call-Id: 12345600@example.com CSeq: 1 INVITE History-Info: ;index=1; Contact: Alice Content-Type: application/sdp Content-Length: [SDP Not Shown] F8 INVITE Example.com -> John's Home INVITE sip:johnhome@192.0.2.1 SIP/2.0 Via: SIP/2.0/TCP proxy.example.com:5060;branch=as2334se Via: SIP/2.0/TCP 192.0.2.3:5060;branch=232sxxeserg From: Alice To: Judy Supported: histinfo Call-Id: 12345600@example.com CSeq: 1 INVITE Record-Route: History-Info: ;index=1; History-Info: ;index=1.1;rc Contact: Alice Content-Type: application/sdp Content-Length: [SDP Not Shown] John's Home server can see that the call was addressed to Judy by evaluating the entry prior to the last entry with the "rc" tag and forwards the call accordingly. Barnes, et al. Expires December 26, 2010 [Page 19] Internet-Draft History-Info Call Flows June 2010 F9 INVITE John's Home -> Judy INVITE sip:judy@198.51.100.2 SIP/2.0 Via: SIP/2.0/TCP 198.51.100.1:5060;branch=abc2334se Via: SIP/2.0/TCP proxy.example.com:5060;branch=as2334se Via: SIP/2.0/TCP 192.0.2.3:5060;branch=232sxxeserg From: Alice To: Judy Supported: histinfo Call-Id: 12345600@example.com CSeq: 1 INVITE Record-Route: History-Info: ;index=1; History-Info: ;index=1.1;rc History-Info: ;index=1.1.1;mp=1.1 History-Info: ;index=1.1.1.1;rc Contact: Alice Content-Type: application/sdp Content-Length: [SDP Not Shown] Figure 4: Sub-Address Example 3.8. 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 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. 3.9. 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. Barnes, et al. Expires December 26, 2010 [Page 20] Internet-Draft History-Info Call Flows June 2010 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. Similar problems arise with other "special" numbers and services used in the PSTN, such as operator services, pay 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 | | | | Barnes, et al. Expires December 26, 2010 [Page 21] Internet-Draft History-Info Call Flows June 2010 | INVITE F1 | | | |--------------->| INVITE F2 | | | |------------->| | | | | INVITE F3 | | | |------------------>| * Rest of flow not shown * F1: INVITE 192.0.2.1 -> proxy.example.com INVITE sip:+18005551002@example.com;user=phone SIP/2.0 Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9 From: Alice ;tag=9fxced76sl To: sip:+18005551002@example.com;user=phone Call-ID: c3x842276298220188511 CSeq: 1 INVITE Max-Forwards: 70 Supported: histinfo History-Info: ;index=1 Contact: Content-Type: application/sdp Content-Length: [SDP Not Shown] F2: INVITE proxy.example.com -> atlanta.com INVITE sip:+15555551002@atlanta.com SIP/2.0 Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-1 Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9 From: Alice ;tag=9fxced76sl To: sip:+18005551002@example.com;user=phone Call-ID: c3x842276298220188511 CSeq: 1 INVITE Max-Forwards: 70 Supported: histinfo History-Info: ;index=1, ;index=1.1;mp=1 Contact: Content-Type: application/sdp Content-Length: [SDP Not Shown] F3: INVITE atlanta.com -> Joe INVITE sip:joe@198.51.100.2 SIP/2.0 Via: SIP/2.0/TCP 198.51.100.1:5060;branch=z9hG4bK-pxk7g-3 Barnes, et al. Expires December 26, 2010 [Page 22] Internet-Draft History-Info Call Flows June 2010 Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-1 Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9 From: Alice ;tag=9fxced76sl To: sip:+18005551002@example.com;user=phone Call-ID: c3x842276298220188511 CSeq: 1 INVITE Max-Forwards: 70 Supported: histinfo History-Info: ;index=1, ;index=1.1;mp=1, ;index=1.1.1;mp=1.1, ;index=1.1.2;rc Contact: Content-Type: application/sdp Content-Length: [SDP Not Shown] Figure 5: Service Number Example 4. Security Considerations The security considerations for the History-Info header 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 and the new SIP/SIPS URI parameter. 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. [RFC5627] Rosenberg, J., "Obtaining and Using Globally Routable User Barnes, et al. Expires December 26, 2010 [Page 23] Internet-Draft History-Info Call Flows June 2010 Agent URIs (GRUUs) in the Session Initiation Protocol (SIP)", RFC 5627, 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. [RFC4769] Livingood, J. and R. Shockey, "IANA Registration for an Enumservice Containing Public Switched Telephone Network (PSTN) Signaling Information", RFC 4769, November 2006. [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] Barnes, M., Audet, F., Schubert, S., Netherlands, T., and C. Holmberg, "An Extension to the Session Initiation Protocol (SIP) for Request History Information", draft-ietf-sipcore-rfc4244bis-00 (work in progress), February 2010. Authors' Addresses Mary Barnes Polycom TX US Email: mary.ietf.barnes@gmail.com Barnes, et al. Expires December 26, 2010 [Page 24] Internet-Draft History-Info Call Flows June 2010 Francois Audet Skype Email: francois.audet@skype.net Shida Schubert NTT Email: shida@agnada.com 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 Barnes, et al. Expires December 26, 2010 [Page 25]