Network Working Group C. Jennings Internet-Draft Cisco Systems Expires: May 28, 2006 F. Audet Nortel Networks J. Elwell Siemens Communications November 24, 2005 Session Initiation Protocol (SIP) URIs for Applications such as Voicemail and Interactive Voice Response (IVR) draft-jennings-sip-voicemail-uri-05 Status of this Memo By submitting this Internet-Draft, each author represents that any applicable patent or other IPR claims of which he or she is aware have been or will be disclosed, and any of which he or she becomes aware will be disclosed, in accordance with Section 6 of BCP 79. 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 material 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/1id-abstracts.txt. The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. This Internet-Draft will expire on May 28, 2006. Copyright Notice Copyright (C) The Internet Society (2005). Abstract The Session Initiation Protocol (SIP) is often used to initiate connections to applications such as voicemail or interactive voice recognition systems. This specification describes a convention for forming SIP service URIs that request particular services based on Jennings, et al. Expires May 28, 2006 [Page 1] Internet-Draft SIP Voicemail URI November 2005 redirecting targets from such applications. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Mechanism (UAS and Proxy) . . . . . . . . . . . . . . . . . . 4 2.1. Target . . . . . . . . . . . . . . . . . . . . . . . . . . 4 2.2. Cause . . . . . . . . . . . . . . . . . . . . . . . . . . 4 2.3. Retrieving Messages . . . . . . . . . . . . . . . . . . . 5 3. Interaction with Request History Information . . . . . . . . . 5 4. Limitations of Voicemail URI . . . . . . . . . . . . . . . . . 6 5. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 6 5.1. Proxy Forwards Busy to Voicemail . . . . . . . . . . . . . 7 5.2. Endpoint forwards busy to Voicemail . . . . . . . . . . . 9 5.3. Endpoint forwards busy to TDM via a gateway . . . . . . . 11 5.4. Endpoint forwards busy to Voicemail with History Info . . 12 5.5. Zero Configuration UM System . . . . . . . . . . . . . . . 14 5.6. Call Coverage . . . . . . . . . . . . . . . . . . . . . . 14 6. Syntax . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15 8. Security Considerations . . . . . . . . . . . . . . . . . . . 15 8.1. Integrity Protection of Forwarding in SIP . . . . . . . . 16 8.2. Privacy Related Issues on the Second Call Leg . . . . . . 16 9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 17 10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 18 10.1. Normative References . . . . . . . . . . . . . . . . . . . 18 10.2. Informative References . . . . . . . . . . . . . . . . . . 18 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 19 Intellectual Property and Copyright Statements . . . . . . . . . . 20 Jennings, et al. Expires May 28, 2006 [Page 2] Internet-Draft SIP Voicemail URI November 2005 1. Introduction Many applications such as Unified Messaging (UM systems and Interactive Voice Recognition (IVR) systems have been developed out of traditional telephony. They can be used for storing and interacting with voice, video, faxes, email, and instant messaging services. Users often use SIP to initiate communications with these applications. When a SIP call is routed to an application, it is necessary that the application be able to obtain several bits of information from the session initiation message so that it can deliver the desired services. For the purpose of this document, we will use UM as the main example, but other applications may use the mechanism defined in this document. The UM needs to know what mailbox should be used and possible reasons for the type of service desired from the UM. Many voice mail systems provide different greetings depending whether the call went to voicemail because the user was busy or because the user did not answer. All of this information can be delivered in existing SIP signaling from the call control that retargets the call to the UM, but there are no conventions for describing how the desired mailbox and the service requested are expressed. It would be possible for every vendor to make this configurable so that any site could get it to work; however, this approach is unrealistic for achieving interoperability among call control, gateways, and unified messaging systems from different vendors. This specification describes a convention for describing this mailbox and service information in the SIP URI so that vendors and operators can build interoperable systems. If there were no need to interoperate with TDM based voicemail systems or to allow TDM systems to use VoIP unified messaging systems, this problem would be a little easier. The problem that is introduced in the VoIP to TDM case is as follows. The SIP system needs to tell a PSTN GW both the subscriber's mailbox identifier (which typically looks like a phone number) and the address of the voicemail system in the TDM network (again a phone number). The question has been asked why the To header cannot be used to specify which mailbox to use. One problem is that the call control proxies cannot modify the To header, and the UACs often set it incorrectly because they do not have information about the subscribers in the domain they are trying to call. This happens because the routing of the call often translates the URI multiple times before it results in an identifier for the desired user that is valid in the namespace that the UM system understands. Jennings, et al. Expires May 28, 2006 [Page 3] Internet-Draft SIP Voicemail URI November 2005 2. Mechanism (UAS and Proxy) The mechanism works by encoding the information for the desired service in the SIP Request-URI that is sent to the UM system. Two chunks of information are encoded, the first being the target mailbox to use and the second being the SIP error code that caused this retargeting and indicates the desired service. The userinfo and hostport parts of the Request-URI will identify the voicemail service, the target mailbox can be put in the target parameter and the reason can be put in the cause parameter. For example, if the proxy wished to use Bob's mailbox because his phone was busy, the URI sent to the UM system could be something like: sip:voicemail@example.com;target=bob%40example.com;cause=486 2.1. Target Target is a URI parameter that indicates the address of the retargeting entity: in the context of UM, this can be the mailbox number. For example, in the case of a voice mail system on the PSTN, the user portion will contain the phone number of the voice mail system, while the target will contain the phone number of the subscriber's mailbox. 2.2. Cause Cause is a URI parameter that is used to indicate the service that the UAS receiving the message should perform. The following values for this URI parameter are defined: +---------------------------------+-------+ | Redirecting Reason | Value | +---------------------------------+-------+ | Unknown/Not available | 404 | | User Busy | 486 | | No Reply | 408 | | Unconditional | 302 | | Deflection during alerting | 487 | | Deflection immediate response | 480 | | Mobile subscriber not reachable | 503 | +---------------------------------+-------+ The mapping to PSTN protocols is important both for gateways that connect the IP network to existing TDM customer's equipment, such as PBXs and voicemail systems, and for gateways that connect the IP network to the PSTN network. ISUP has signaling encodings for this information that can be treated as roughly equivalent for the purposes here. For this reason, this specification uses the names of Jennings, et al. Expires May 28, 2006 [Page 4] Internet-Draft SIP Voicemail URI November 2005 Redirecting Reason values defined in ITU-T Q.732.2-5 [7]. In this specification, the Redirecting Reason Values are referred to as "Causes". It should be understood that the term "Cause" has nothing to do with PSTN "Cause values" (as per ITU-T Q.850 [8] and RFC 3398 [4]) but are instead mapped to ITU-T Q.732.2-5 Redirecting Reasons. Since ISUP interoperates with other PSTN networks such as Q.931 [9] and QSIG [10] using well-known rules, it makes sense to use the ISUP names as the most appropriate superset. If no appropriate mapping to a cause value defined in this specification exists in a network, it would be mapped to 302 "Unconditional". Similarly, if the mapping occurs from one of the causes defined in this specification to a PSTN system that does not have an equivalent reason value, it would be mapped to that network's equivalent of "Unconditional". If a new cause parameter needs to be defined, this specification will have to be updated. The user portion of the URI SHOULD be used as the address of the voicemail system on the PSTN, while the target SHOULD be mapped to the original redirecting number on the PSTN side. The redirection counters SHOULD be set to one unless additional information is available. 2.3. Retrieving Messages The UM system MAY use the fact that the From header is the same as the URI target as a hint that the user wishes to retrieve messages. 3. Interaction with Request History Information The Request History mechanism [5] provides more information relating to multiple retargetings. It is reasonable to have systems in which both the information in this specification and the History information are included and one or both are used. History-Info specifies a means of providing the UAS and UAC with information about the retargeting of a request. This information includes the initial Request-URI and any retarget-to URIs. This information is placed in the History-Info header field, which, except where prevented by privacy considerations, is built up as the request progresses and, upon reaching the UAS, is returned in certain responses. History-Info, when deployed at relevant SIP entities, is intended to provide a comprehensive trace of retargeting for a SIP request, along with the SIP response codes that led to retargeting. Jennings, et al. Expires May 28, 2006 [Page 5] Internet-Draft SIP Voicemail URI November 2005 History-Info can complement this specification. In particular, when a proxy inserts a URI containing the parameters defined in this specification into the Request-URI of a forwarded request, the proxy can also insert a History-Info header field entry into the forwarded request and the URI in that entry will incorporate these parameters. Therefore even if the Request-URI is replaced as a result of rerouting by a downstream proxy, the History-Info header field will still contain these parameters, which may be of use to the UAS. Consequently, UAS that make use of this information may find the information in the History-Info header, and/or in the Request-URI, depending on the capability of the proxy to support generation of History-Info, or the behavior of downstream proxies, and therefore applications need to take this into account. 4. Limitations of Voicemail URI This specification requires the proxy that is requesting the service to understand what are valid targets on the UM system. For practical purposes this means that the approach is unlikely to work in many cases in which the proxy is not configured with information about the UM system or is not in the same administrative domain. This approach only works when the service the call control wants applied is fairly simple. For example it does not allow the proxy to express information like "Do not offer to connect to the target's colleague because that address has already been tried". The limitations discussed in this section are addressed by History- Info [5]. 5. Examples This section provides some example use cases for the solution proposed in this document. For the purpose of this document, UM is used as the main example, but other applications may use this mechanism. The examples are intended to highlight the potential applicability of this solution and are not intended to limit its applicability. Also the examples show just service retargeting on busy, but can easily be adapted to show other forms of retargeting. In several of the examples, the URI are broken across more than one line. This was only done for formatting and is not a valid SIP messages. Some of the characters in the URIs are not correctly escaped to improve readability. The examples are all shown using Jennings, et al. Expires May 28, 2006 [Page 6] Internet-Draft SIP Voicemail URI November 2005 sip: with UDP transport, for readability. It should be understood that using sips: with TLS transport is preferable. 5.1. Proxy Forwards Busy to Voicemail In this example, Alice calls Bob. Bob's proxy determines that Bob is busy, and the proxy forwards the call to Bob's voicemail. Alice's phone is at 192.0.2.1 while Bob's phone is at 192.0.2.2. The important thing to note is the URI in message F7. Alice Proxy Bob voicemail | | | | | INVITE F1 | | | |--------------->| INVITE F2 | | | |------------->| | |(100 Trying) F3 | | | |<---------------| 486 Busy F4 | | | |<-------------| | | | ACK F5 | | | |------------->| | |(181 Call is Being Forwarded) F6 | |<---------------| | INVITE F7 | | |--------------------------------->| * Rest of flow not shown * F1: INVITE 192.0.2.1 -> proxy.example.com INVITE sip:+15555551002@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:+15555551002@example.com;user=phone Call-ID: c3x842276298220188511 CSeq: 1 INVITE Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: *Body length goes here* * SDP goes here* Jennings, et al. Expires May 28, 2006 [Page 7] Internet-Draft SIP Voicemail URI November 2005 F2: INVITE proxy.example.com -> 192.0.2.2 INVITE sip:+15555551002@192.0.2.2 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:+15555551002@example.com;user=phone Call-ID: c3x842276298220188511 CSeq: 1 INVITE Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: *Body length goes here* * SDP goes here* F4: 486 192.0.2.2 -> proxy.example.com SIP/2.0 486 Busy Here 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:+15555551002@example.com;user=phone;tag=09xde23d80 Call-ID: c3x842276298220188511 CSeq: 1 INVITE Content-Length: 0 F7: INVITE proxy.example.com -> um.example.com INVITE sip:voicemail@example.com;\ target=sip:+15555551002%40example.com;user=phone;\ cause=486 SIP/2.0 Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-2 Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9 From: Alice ;tag=9fxced76sl To: sip:+15555551002@example.com;user=phone Call-ID: c3x842276298220188511 CSeq: 1 INVITE Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: *Body length goes here* * SDP goes here* Jennings, et al. Expires May 28, 2006 [Page 8] Internet-Draft SIP Voicemail URI November 2005 5.2. Endpoint forwards busy to Voicemail In this example, Alice calls Bob. Bob is busy, but forwards the session directly to his voicemail. Alice's phone is at 192.0.2.1 while Bob's phone is at 192.0.2.2. The important thing to note is the URI in the Contact in message F3. Alice Proxy Bob voicemail | | | | | INVITE F1 | | | |--------------->| INVITE F2 | | | |------------->| | | | 302 Moved F3 | | | 302 Moved F4 |<-------------| | |<---------------| | | | ACK F5 | | | |--------------->| ACK F6 | | | |------------->| | | INVITE F7 | |-------------------------------------------------->| * Rest of flow not shown * F1: INVITE 192.0.2.1 -> proxy.example.com INVITE sip:+15555551002@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:+15555551002@example.com;user=phone Call-ID: c3x842276298220188511 CSeq: 1 INVITE Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: *Body length goes here* * SDP goes here* Jennings, et al. Expires May 28, 2006 [Page 9] Internet-Draft SIP Voicemail URI November 2005 F2: INVITE proxy.example.com -> 192.0.2.2 INVITE sip:line1@192.0.2.2 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:+15555551002@example.com;user=phone Call-ID: c3x842276298220188511 CSeq: 1 INVITE Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: *Body length goes here* * SDP goes here* F3: 302 192.0.2.2 -> proxy.example.com SIP/2.0 302 Moved Temporarily 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:+15555551002@example.com;user=phone;tag=09xde23d80 Call-ID: c3x842276298220188511 CSeq: 1 INVITE Contact: Content-Length: 0 F7: INVITE proxy.example.com -> um.example.com INVITE sip: voicemail@example.com;\ target=sip:+15555551002%40example.com;user=phone;\ cause=486 SIP/2.0 Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-2 Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9 From: Alice ;tag=9fxced76sl To: sip:+15555551002@example.com;user=phone Call-ID: c3x842276298220188511 CSeq: 1 INVITE Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: *Body length goes here* Jennings, et al. Expires May 28, 2006 [Page 10] Internet-Draft SIP Voicemail URI November 2005 * SDP goes here* 5.3. Endpoint forwards busy to TDM via a gateway In this example, the voicemail is reached via a gateway to a TDM network. Bob's number is +1 555 555-1002, while voicemail's number on the TDM network is +1-555-555-2000. The call flow is the same as in Section 5.2 except for the Contact URI in F4 and the Request URI in F7. Alice Proxy Bob voicemail | | | | | INVITE F1 | | | |--------------->| INVITE F2 | | | |------------->| | |(100 Trying) F3 | | | |<---------------| 302 Moved F4 | | | |<-------------| | | | ACK F5 | | | |------------->| | |(181 Call is Being Forwarded) F6 | |<---------------| | INVITE F7 | | |--------------------------------->| * Rest of flow not shown * F4: 486 192.0.2.2 -> proxy.example.com SIP/2.0 302 Moved temporarily 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:+15555551002@example.com;user=phone;tag=09xde23d80 Call-ID: c3x842276298220188511 CSeq: 1 INVITE Contact: Content-Length: 0 Jennings, et al. Expires May 28, 2006 [Page 11] Internet-Draft SIP Voicemail URI November 2005 F7: INVITE proxy.example.com -> gw.example.com INVITE sip:+15555552000@example.com;user=phone;\ target=tel:+15555551002;cause=486\ SIP/2.0 Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-2 Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9 From: Alice ;tag=9fxced76sl To: sip:+15555551002@example.com;user=phone Call-ID: c3x842276298220188511 CSeq: 1 INVITE Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: *Body length goes here* * SDP goes here* 5.4. Endpoint forwards busy to Voicemail with History Info This example illustrates how History Info works in conjunction with service retargeting. The scenario is the same as Section 5.1. F1: INVITE 192.0.2.1 -> proxy.example.com INVITE sip:+15555551002@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:+15555551002@example.com;user=phone Call-ID: c3x842276298220188511 CSeq: 1 INVITE Max-Forwards: 70 Contact: History-Info: ;index=1 Content-Type: application/sdp Content-Length: *Body length goes here* * SDP goes here* Jennings, et al. Expires May 28, 2006 [Page 12] Internet-Draft SIP Voicemail URI November 2005 F2: INVITE proxy.example.com -> 192.0.2.2 INVITE sip:line1@192.0.2.2 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:+15555551002@example.com;user=phone Call-ID: c3x842276298220188511 CSeq: 1 INVITE Max-Forwards: 70 Contact: History-Info: ;index=1, ;index=1.1 Content-Type: application/sdp Content-Length: *Body length goes here* * SDP goes here* F7: INVITE proxy.example.com -> um.example.com INVITE sip: voicemail@example.com;\ target=sip:+15555551002%40example.com;user=phone;\ cause=486 SIP/2.0 Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-2 Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9 From: Alice ;tag=9fxced76sl To: sip:+15555551002@example.com;user=phone Call-ID: c3x842276298220188511 CSeq: 1 INVITE Max-Forwards: 70 Contact: History-Info: ;index=1, ;index=1.1 ;index=2 Contact: Content-Type: application/sdp Content-Length: *Body length goes here* * SDP goes here* Jennings, et al. Expires May 28, 2006 [Page 13] Internet-Draft SIP Voicemail URI November 2005 5.5. Zero Configuration UM System In this example, the UM system has no configuration information specific to any user. The proxy is configured to pass a URI that provides the prompt to play and an email address in the user portion of the URI to which the recorded message is to be sent. The call flow is the same as in Section 5.1, except that the URI in F7 changes to specify the user part as Bob's email address, and the Netann [6] URI play parameter specifies where the greeting to play can be fetched from. F7: INVITE proxy.example.com -> voicemail.example.com INVITE sip:voicemail@example.com;target=mailto:bob%40example.com;\ cause=486;play=http://www.example.com/bob/busy.wav SIP/2.0 Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-2 Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9 From: Alice ;tag=9fxced76sl To: sip:+15555551002@example.com;user=phone Call-ID: c3x842276298220188511 CSeq: 1 INVITE Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: *Body length goes here* * SDP goes here* In addition, if the proxy wished to indicate a VXML script that the UM should execute, it could add a parameter to the URI in the above message that looked like: voicexml=http://www.example.com/bob/busy.vxml 5.6. Call Coverage In a Call Coverage example, a user on the PSTN calls a 800 number. The GW sends this to the proxy which recognizes that the helpdesk is the target. Alice and Bob are staffing the help desk and are tried sequentially but neither answers, so the call is forwarded to the helpdesk's voice mail. The details of this flow are trivial and not shown: the key item in this example is that the INVITE to Alice and Bob looks as follows: INVITE sip:voicemail@example.com;target=helpdesk%40example.com;\ cause=302 SIP/2.0 Jennings, et al. Expires May 28, 2006 [Page 14] Internet-Draft SIP Voicemail URI November 2005 6. Syntax This specification updates the ABNF in Section 25 of RFC 3261 [1] to add the target-param to the uri-parameter as shown below. uri-parameter = transport-param / user-param / method-param / ttl-param / maddr-param / lr-param / other-param / target-param / cause-param target-param = "target" EQUAL pvalue cause-param = "cause" EQUAL Status-Code It is worth noting that the ABNF requires some characters to be escaped if they occur in the value of the target parameters. For example, the "@" character needs to be escaped. 7. IANA Considerations This specification adds a new value to the IANA registration in the "SIP/SIPS URI Parameters" sub-registry at http://www.iana.org/assignments/sip-parameters as defined in [3]. Parameter Name Predefined Values Reference ____________________________________________ target No [RFCAAAA] cause Yes [RFCAAAA] [Note to IANA: Please replace AAAA with the RFC number of this specification. 8. Security Considerations This draft discusses transactions involving at least three parties, which increases the complexity of the privacy issues. The new URI parameters defined in this draft are generally sent from a Proxy or call control system to a Unified Messaging (UM) system or to a gateway to the PSTN and then to a voicemail system. These new parameters tell the UM what service the proxy wishes to have performed. Just as any message sent from the proxy to the UM needs to be integrity protected, these messages need to be integrity protected to stop attackers from, for example, causing a voicemail meant for a company's CEO to go to an attacker's mailbox. RFC 3261 provides a TLS mechanism suitable for performing this integrity Jennings, et al. Expires May 28, 2006 [Page 15] Internet-Draft SIP Voicemail URI November 2005 protection. The signaling from the Proxy to the UM or gateway will reveal who is calling whom and possibly some information about a user's presence based on whether the call was answered or sent to voicemail. This information can be protected by encrypting the SIP traffic between the Proxy and UM or gateway. Again, RFC 3261 contains mechanisms for accomplishing this using TLS. Implementations should implement and use TLS. 8.1. Integrity Protection of Forwarding in SIP The forwarding of a call in SIP brings up a very strange trust issue. Consider the normal case when A calls B and the call gets forwarded to C by a network element in B's domain, and then C answers the call. A has called B but ended up talking to C. This scenario may be hard to separate from a man in the middle attack. There are two possible solutions. One is that B sends back information to A saying don't call me, call C and signs it as B. The problem is that this solution involves revealing that B has forwarded to C, which B often may not want to do. For example, B may be a work phone that has been forwarded to a mobile or home phone. The user does not want to reveal their mobile or home phone number but, even more importantly, does not want to reveal that they are not in the office. The other possible solution is that A needs to trust B only to forward to a trusted identity. This requires a hop by hop transitive trust such that each hop will only send to a trusted next hop and each hop will only do things that the user at that hop desired. This solution is enforced in SIP using the SIPS URI and TLS based hop by hop security. It protects from an off axis attack, but if one of the hops is not trustworthy, the call may be diverted to an attacker. Any redirection of a call to an attacker's mailbox is serious. It is trivial for an attacker to make its mailbox seem very much like the real mailbox and forward the messages to the real mailbox so that the fact that the messages have been intercepted or even tampered with escapes detection. 8.2. Privacy Related Issues on the Second Call Leg In the case where A calls B and gets redirected to C, occasionally people suggest there is a a requirement for the call leg from B to C to be anonymous. The SIP case is not the PSTN and there is no call leg from B to C; instead there is a VoIP session between A and C. If Jennings, et al. Expires May 28, 2006 [Page 16] Internet-Draft SIP Voicemail URI November 2005 A has put a To header field value containing B in the initial invite message, unless something special is done about it, C would see that To header field value. If the person who answers phone C says "I think you dialed the wrong number, who were you trying to reach?" A will probably specify B. If A does not want C to see that the call was to B, A needs a special relationship with the forwarding Proxy to induce it not to reveal that information. The call should go through an anonymization service that provides session or user level privacy (as described in RFC 3323 [2]) service before going to C. It is not hard to figure out how to meet this requirement, but it is unclear why anyone would want this service. The scenario in which B wants to make sure that C does not see that the call was to B is easier to deal with but a bit weird. The usual argument is Bill wants to forward his phone to Monica but does not want Monica to find out his phone number. It is hard to imagine that Monica would want to accept all Bill's calls without knowing how to call Bill to complain. The only person Monica will be able to complain to is Hilary, when she tries to call Bill. Several popular web portals will send SMS alert message about things like stock prices and weather to mobile phone users today. Some of these contain no information about the account on the web portal that initiated them, making it nearly impossible for the mobile phone owner to stop them. This anonymous message forwarding has turned out to be a really bad idea even where no malice is present. Clearly some people are fairly dubious about the need for this, but never mind: let's look at how it is solved. In the general case, the proxy needs to route the call through an anonymization service and everything will be cleaned up. Any anonymization service that performs the "Privacy: Header" Service in RFC 3323 [2] must remove the cause and target URI parameters from the URI. Privacy of the parameters when they from part of a URI within the History-Info header is covered in History-Info [5]. This specification does not discuss the security considerations of mapping to a PSTN Gateway. Security implications of mapping to ISUP for example, are discussed in RFC 3398 [4]. 9. Acknowledgments Many thanks to Mary Barnes, Steve Levy, Dean Willis, Allison Mankin, Martin Dolly, Paul Kyzivat, Erick Sasaki, Lyndsay Campbell, Keith Drage, Miguel Garcia, Sebastien Garcin, Roland Jesske, Takumi Ohba and Rohan Mahy. Jennings, et al. Expires May 28, 2006 [Page 17] Internet-Draft SIP Voicemail URI November 2005 10. References 10.1. Normative References [1] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002. [2] Peterson, J., "A Privacy Mechanism for the Session Initiation Protocol (SIP)", RFC 3323, November 2002. [3] 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. 10.2. Informative References [4] Camarillo, G., Roach, A., Peterson, J., and L. Ong, "Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping", RFC 3398, December 2002. [5] Barnes, M., "An Extension to the Session Initiation Protocol (SIP) for Request History Information", RFC 4244, November 2005. [6] Burger, E., "Basic Network Media Services with SIP", draft-burger-sipping-netann-11 (work in progress), February 2005. [7] "Stage 3 description for call offering supplementary services using signalling system No. 7: Call diversion services", ITU- T Recommendation Q.732.2-5, December 1999. [8] "Usage of cause and location in the Digital Subscriber Signalling System No. 1 and the Signalling System No. 7 ISDN User Part", ITU-T Recommendation Q.850, May 1998. [9] "ISDN user-network interface layer 3 specification for basic call control", ITU-T Recommendation Q.931, May 1998. [10] "Information technology - Telecommunications and information exchange between systems - Private Integrated Services Network - Circuit mode bearer services - Inter-exchange signalling procedures and protocol", ISO/IEC 11572, March 2000. Jennings, et al. Expires May 28, 2006 [Page 18] Internet-Draft SIP Voicemail URI November 2005 Authors' Addresses Cullen Jennings Cisco Systems 170 West Tasman Drive Mailstop SJC-21/2 San Jose, CA 95134 USA Phone: +1 408 421-9990 Email: fluffy@cisco.com Francois Audet Nortel Networks 4655 Great America Parkway Santa Clara, CA 95054 US Phone: +1 408 495 3756 Email: audet@nortel.com John Elwell Siemens Communications Technology Drive Beeston, Nottingham NG9 1LA UK Email: john.elwell@siemens.com Jennings, et al. 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Acknowledgment Funding for the RFC Editor function is currently provided by the Internet Society. Jennings, et al. Expires May 28, 2006 [Page 20]