Guidelines and Protocol Extensions for Combining SIP Based Real-time Media Sessions With XMPP Based Instant Messaging and Presence Service.

Currently most standards-based Voice over IP (VoIP) deployments use Session Initiation Protocol (SIP). In addition to providing basic voice service SIP has an extensive support for more advanced telephony features including interworking with the traditional Public Switched Telephone Network (PSTN). SIP is also gaining popularity in the field of video communication. At the same time, the Extensible Messaging and Presence Protocol (XMPP) is enjoying widespread usage in instant messaging and presence services. An interesting scenario arises when a SIP based voice and video service is combined together with an XMPP based instant messaging and presence service. This memo describes how SIP based real-tome sessions and XMPP based IM and presence can be offered using existing server implementations. This memo also presents a set of requirements and protocol extensions for SIP User Ageng and XMPP client implementations in order to offer a seamless usage experience when using SIP based VoIP with XMPP based instant messaging and presence. Combining SIP based real-time services with XMPP based presence and IM service can be accomplished for the most part in the endpoints; little if anything needs to be done in the service infrastructure. It is also possible to achieve seamless integration even when SIP and XMPP services are offered by different service providers. The main issues that need to be addressed when offering such combined services are identities and addressing. SIP and XMPP both use a similar addressing scheme (based on "user@domain" structure) to identify users and endpoints but there are some subtle differencies as well. It is not possible to assume any algorithmic correlation between SIP and XMPP Universal Resource Identifiers (URI), even when they identify the same user or endpoint. New protocol mechanisms are needed to tie together communication contexts that are based on the two protocols. We do not discuss how protocol translation through a gateway could be performed between the protocols; this is the subject of other work, see for example . We focus on one-to-one communication only. Multiparty use cases such as combining SIP voice conference with XMPP IM group chat are beyond the scope. The document structure is as follows: present the document conventions and definitions, presents deployment scenarios and use cases, lists the requirements, provides an operview of the protocol operation, provides the defintions, and examples are presented in .

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14, RFC 2119 and indicate requirement levels for compliant implementations. The following definitions are used in this memo: Integrated endpoint is an implementation that combines the functionality of SIP User Agent and XMPP client and can offer its user a seamless user experience in the sense that a single UI and contact information can be user for voice and video communication using the SIP protocol as well as instant messaging and presence sharing using XMPP. We assume an integrated endpoint is able to support SIP and XMPP protocol extensions defined in this memo. Separated endpoint refers to independent SIP User Agent and XMPP client implementations that are not aware of each other if they are used by the same user. The users uses SIP UA for voice and video, while using XMPP client for IM and presence. It is assumed that a separated endpoint does not support any SIP or XMPP protocol extensions defined in this memo.

This section presents the assumptions we make about SIP and XMPP deployments with respect to endpoints and server infrastructure. We also enumerate the actual use cases that the combined service must accommodate for.

The assumption is that the server infrastructure for SIP and XMPP are totally separated, thus no exchange of information is expected between them. There is no assumption that SIP and XMPP services are even offered by the same service provider. This means that the user identities can even be from two different domains. However, if the same service provider offers both SIP and XMPP service, it is recommended that the URIs sip:user@domain and xmpp:user@domain correspond to the same user. We assume that the user intially only knows the contact address of the other user in one protocol space. The contact address for the other protocol must be learned through this. We consider only cases where two integrated endpoints interact. When an intergrated endpoint communicates with a separated endpoint, normal SIP and XMPP procedures apply and no extensions defined in this memo are used.

Two users who both use an integrated endpoint start an (XMPP) IM conversation. After the exchange of initial messages, their UIs show that the other party is capable of (SIP) voice and/or video in addition to IM. Either user can at any point add voice and/or video component to the conversation in such a way that they end up in the same endpoint and conversation context where the IM exchange is already taking place. (Note that for this use case the conversation initiatior initially only needs to know the other user’s XMPP user id.) Two users who both use an integrated endpoint start a (SIP) voice and/or video call. As the call is established, their UIs show that the other party is capable for (XMPP) IM. Either user can at any point add an IM component to the conversation in such a way that they end up in the same endpoint and conversation context where the voice and/or call is already taking place. (Note that for this use case the caller initially only needs to know the other user’s SIP user id.) It is possible to vary the two cases above so that one fo the users initiates a "multimedia call" to the other one, i.e. SIP voice and/or video and XMPP IM are all active from the stary. Tehcnically this happens according to the two-phased approach above, and it inivisble from the end-user. A user of an integrated endpoint is able to publish her SIP voice and video related presence status as part of her XMPP presence. The status includes information such as user’s SIP contact address (for the integrated endpoint), media capability and availability and whether the user is currently “on the phone”. Another user of an integrated endpoint can see the presence status (assuming she is authorized for that) and based on that initiate calls. For instance watcher’s UI can now for certainty show that the user on her roster is capable of receiving “multimedia calls” (Note that the watcher initially only needs to know the other user’s XMPP user id.) OPEN ISSUE: Is there a use case for discovering other user’s XMPP identity based on her SIP identity without needing to establish a media session. SIP OPTIONS would be one possibility for that (as we do not assume SIP presence support).

This section presents the protocol requirements. It must be possible for the sender of an XMPP message to include its SIP contact information within the message. The contact information must allow the recipient to establish the SIP session such that the session is routed to the same endpoint which is hosting the XMPP conversation. As including the same information in every message would create some overhead, it is desirable to be able to convey the contact only once per IM conversation/thread. It must be possible for the caller to convey in the SIP session initiation information which allows the callee to correlate the session with an ongoing XMPP conversation. It must be possible for the SIP User Agent Client and User Agent Server that establish a real-time media session to exchange their XMPP contact information so that either endpoint can at any time send XMPP messages to the other endpoint. It must be possible for the sender to convey in the XMPP message information which allows the recipient to correlate the message with an ongoing SIP session. It must be possible to include SIP real-time media related contact and status in XMPP presence information. The information must contain at least SIP contact address to identify a user or a user agent instance, media capabilities and general availability status OPEN ISSUE: Should we define requirements related to “error” or “corner” cases here. For instance what happens to communication attempts after the other party has closed the conversation context, e.g. a correlated XMPP message that is sent after the related SIP session is terminated. Or a SIP INVITE that appears two days after the XMPP IM conversation was ended. NOTE: There is also an implicit requirement that we must take Session Border Controllers into account when defining how SIP User Agents are able to identify an existing session between them in a common manner.

Both SIP and XMPP allow registration of multiple endpoints using the same identifier, either a SIP AOR or XMPP Jabber ID (JID). When two endpoints are enganged in an IM conversation, for example, and wish to add a voice component to the communication, it has be ensured that the resulting SIP dialog is targeted to the same endoint that is running the IM conversation. Fortunately, both XMPP and SIP provide a mechanism for this. defines mechanisms for a SIP UA to obtain and use a Globally Routable User Agent (UA) URI. A GRUU will route a call to a specific UA instance. Unfortunately, not all SIP registrars support the optional GRUU mechanism. In that case the SIP UA has not other option but to use its AOR in place of GRUU. In XMPP, a "full JID" consists of a name, domain and a resource identifier in the form of <name@domain/resource>. The resource identifier can be used to identify a specific endpoint.

| | | | (F1) INVITE | |---------------------->| | | | (F2) 200OK | |<----------------------| | | | (F3) ACK | |---------------------->| | | | RTP media | |<=====================>| | | ]]> This case starts by one endpoint (Bob) sending a message stanza to another (Alice). Bob includes <thread> element in the message and chooses a unique value for it. In his first message Bob also includes his SIP URI in <sip-contact> element, defined in . If Bob has been able to obtain a GRUU from his registrar, he populates the <sip-contact> with that. Otherwise a mere AOR is used. When Alice receives Bob’s SIP URI Alice stores it associated with the current <thread>. When responding to Bob’s messages Alice also includes <thread> and her SIP URI (GRUU or AOR) in <sip-contact>. Upon receiving Alice’s first message Bob stores Alice’s SIP URI associated with the current <thread>. In addition to containing a SIP URI, <sip-contact> also conveys the information whether an endpoint supports audio or video or both medias. So, based on exchanged <sip-contact> elements, both endpoints now know each others SIP URIs and media capabilities. The same <thread> value is used in all further messages by both endpoints to keep track of the conversation. As long as the <thread> value is unchanged, the <sip-contact> element need not be repeated, unless either endpoint’s SIP GRUU changes for some reason. When either party wants to extend the IM conversation by adding SIP voice or video session, they address a SIP INVITE to the SIP URI learned in <sip-contact>. If <contact> contained a GRUU, it ensures that the INVITE will be routed to the correct endpoint. The caller populates XMPP-Thread header, defined in , in the INVITE with the value of <thread>. The callee is thus able to correlate the SIP session to the IM conversation. The callee replays XMPP-Thread in responses to INVITE to indicate that the correlation was successful.

In this case two endpoints first have a SIP voice or video session. They exchange their full JIDs within the session establishment. The caller (Bob) adds XMPP-Contact header, defined in , in INVITE populating it with his full JID. XMPP-Contact also includes an opaque end-to-end identifier for the session common to both endpoints. The callee (Alice) stores this information as part of the session state. In 200 OK response to INVITE Alice includes similar XMPP-Contact header with her full JID, and replays the end-to-end session identifier. Bob stores this information as part of his session state. Both endpoints now know each others full JIDs and have a common reference to the session. OPEN ISSUE: Instead of defining XMPP-specific session identifier we could use SIP call-id as is. However, there is a concern that call-id may be changed by SBCs en route is thus might not be a useful as a common reference for both User Agents. defines a Session-ID header that potentially could also be used. When either endpoint wants to send XMPP messages to each other, they address them to the full JID learned from XMPP-Contact header. This ensures that the messages reach the correct endpoint. In the very first message the sender also includes <sip-correlation> element, defined in , with the session identifier value learned from XMPP-Contact. The recipient uses the value to correlate the message with the SIP session and echoes it back the first message it sends to indicate that the correlation was succesful.

SIP related presence information is encoded in XMPP presence schema as an extension. It includes endpoints SIP URI (preferably GRUU but can be also AOR), media capabilities (audio, video), and availability (open, closed, busy). Based on this information XMPP Presence watcher is able to initiate SIP voice and video sessions.

In this section we define protocol extensions to meet the requirements stated in the previous section.

The child elements of the message stanza can be extended with elements from other namespaces. For the purposes of carrying a SIP identifier in the message stanza, we define two new elements, the <sip-contact> element and <sip-correlation> element.

The <sip-contact> element, qualified by "http://jabber.org/protocol/sip-contact" namespace, has one mandatory attribute, "target", which defines the target's SIP URI. The format of the "target" attribute is an absoluteURI defined in . When an endpoint initiates an XMPP IM conversation, and wants to offer a possibility to later add a SIP real-time media session, it MUST include a <sip-contact> element as a child element in the first the <message> stanza it sends, and MUST add a <thread> element and populate its value according to . The endpoint MUST include in the “target” attribute of the <sip-contact> element the SIP URI it wishes to be contacted at. If the endpoint is aware of its GRUU, it SHOULD use that as the value in the “target” attribute; otherwise it MAY use its AOR. The endpoint receiving an XMPP <message> stanza that includes <thread> and <sip-contact> elements MUST copy the <thread> element value to the first <message> stanza it sends back, as defined in , and MUST include a <sip-contact> element and set the “target” attribute value to the SIP URI it wishes to be contacted at. An endpoint MUST add its audio and video capabilities defined in to the contact address in the “target” attribute, and MUST understand those capabilities if received from the other endpoint. An endpoint MAY add other media capabilities. When an endpoint receives a <sip-contact> element in a <message> stanza, it MUST store the value of the target attribute, and use it as the SIP URI in an INVITE request if the user of the endpoint would like to add a SIP session to the IM conversation context For example, a <sip-contact> element carrying a SIP Globally Routable Unique URI (GRUU) would be

;gr=urn: uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6; audio;video"> ]]>

In order to indicate that an XMPP IM conversation is related to an existing SIP session, we define a new element in the message stanza called <sip-correlation>. The <sip-correlation>, qualified by the "http://jabber.org/protocol/sip-correlation" namespace, has one mandatory attribute, "value". The endpoint sending the <message> stanza MUST set the "value" attribute to the value of the correlation-value parameter of the SIP XMPP-Contact header. The XMPP-Contact header is exhanged during the setup of the SIP session. The endpoint MUST also include a <thread> element in the message. An endpoint receiving a <message> stanza which includes a <sip-correlation> element MUST first compare the "from" attribute value of the <message> stanza to the XMPP JID in the contact-value of the XMPP-Contact header of its active SIP sessions. If a matching SIP session is found, the endpoint MUST compare the "value" attribute to the correlation-value of the XMPP-Contact header of that SIP session. If the "value" attribute matches to correlation-value of an XMPP-Contact header, the <message> stanza is correlated to that SIP session. If the user replies to the message, the values of the <thread> element and the "value" attribute of the <sip-correlation> element in the first reply MUST be the same as in the original message. This indicates that the correlation was successful. The correlation is valid as long as the messages are exchanged with the same <thread> value. As an example, a <sip-correlation> element carrying the XMPP-Contact header correlation-value parameter of an existing SIP session would be <sip-correlation value="xyz123"/>

OPEN ISSUE: XML Schemas to be provided

The XMPP presence stanza defined in can be extended with any properly-namespaced child element. We define a new optional element called <contact> which, qualified by the http://jabber.org/protocol/contact namespace, MAY appear as a child element in the presence stanza. The contact element SHOULD be set to the SIP address (GRUU or AOR) the endpoint wishes to be contacted at for further communication. Exact syntax and XML Schema of the correlation element is TBD.

In order to indicate that the SIP dialog is related to an existing XMPP messaging session, we define a new SIP header, called XMPP-Thread. The XMPP-Thread contains information that can be used by the terminating endpoint to correlate the SIP session establishment to an existing XMPP conversation. The endpoint sending a SIP INVITE request MUST include an XMPP-Thread header, and set its value to the value of the <thread> element used in the XMPP IM conversation. The endpoint receiving a SIP INVITE which inludes an XMPP-Thread header act as follows: it first compares the Contact header value with all SIP GRUUs from <sip-contact> elements in active XMPP IM conversations, and unless a match is found, compares P-Asserted-Identity header value with all other SIP URIs from <sip-contact> elements in active XMPP IM conversations if a single match is found, the receiving endpoint MUST the value of the XMPP-Thread element to the <thread> element values of existing XMPP IM conversations the endpoint has active, and if the value matches, the SIP INVITE is correlated to the IM conversation. The endpoint MUST copy the XMPP-Thread header to any of the 2xx series responses. defines support of XMPP-Contact header in SIP requests and responses, and extends Table 2 of . MESSAGE, SUBCRIBE and NOTIFY, REFER, INFO, UPDATE, PRACK, and PUBLISH are defined in , , , , , , and , respectively.

Header field where proxy ACK BYE CAN INV OPT REG MSG ------------ ----- ----- --- --- --- --- --- --- --- XMPP-Thread R - - - o - - - XMPP-Thread 2xx - - - o - - - SUB NOT REF INF UPD PRA PUB --- --- --- --- --- --- --- XMPP-Thread R - - - - - - - XMPP-Thread 2xx - - - - - - - The syntax of the XMPP-Thread using augmented Backur-Naur Form (ABNF) is defined as follows:

XMPP-Thread = "XMPP-Thread" HCOLON thread-value thread-value = token ;defined in RFC3261

The XMPP-Contact header is used to carry the XMPP JID and an opaque token that can be used for correlation purposes. When an endpoint initiates a SIP session, and wants to offer the possibility to later add an XMPP IM conversation, it MUST include an XMPP-Contact header in the intitial SIP request. The contact-value of the XMPP-Contact header MUST be set to the full XMPP JID the endpoint wishes to be contacted at, and the correlation-value SHOULD be set to the value of the Call-ID of the SIP session. If the Call-ID cannot be used, the endpoint MUST select the correlation-value such that it fulfills the same uniqueness requirements defined for Call-ID in Section 8.1.1.4 of . An endpoint sending a 2xx series response to an INVITE that contains XMPP-Contact header MUST include a XMPP-Contact header in the response, MUST set the contact-value of the header to the full XMPP JID the endpoint wishes to be contacted at, and MUST copy the correlation-value from the INVITE to the 2xx response. The endpoint receiving a SIP request or response with an XMPP-Contact header, MUST store the value of the correlation-value in order to be able to later correlate an XMPP IM conversation with the SIP session. An endpoint initiating a correlated XMPP IM conversation MUST use the correlation-value in the <sip-correlation> element as specified in . defines support of XMPP-Contact header in SIP requests and responses, and extends Table 2 of . MESSAGE, SUBCRIBE and NOTIFY, REFER, INFO, UPDATE, PRACK, and PUBLISH are defined in , , , , , , and , respectively.

Header field where proxy ACK BYE CAN INV OPT REG MSG ------------ ----- ----- --- --- --- --- --- --- --- XMPP-Contact R - - - o - - - XMPP-Contact 2xx - - - o - - - SUB NOT REF INF UPD PRA PUB --- --- --- --- --- --- --- XMPP-Contact R - - - - - - - XMPP-Contact 2xx - - - o - - - The syntax of the XMPP-Contact using augmented Backur-Naur Form (ABNF) is defined as follows:

XMPP-Contact = "XMPP-Contact" HCOLON contact-value SEMI correlation-value contact-value = token ;defined in RFC3261 correlation-value =token

| |---------------------->| | | | (F2) | |<----------------------| | | | (F3) INVITE | |---------------------->| | | | (F4) 200OK | |<----------------------| | | | (F5) ACK | |---------------------->| | | | RTP media | |<=====================>| | | ]]> Bob and Alice are engaged in an XMPP IM session, when Bob would like to add voice/video component to the discussion. When Bob and Alice exhange message stanzas, they also include the SIP address they would like to be contacted at. In this example, Bob is aware of its GRUU, while Alice is merely aware of her SIP AOR. Both include the SIP identifier in a contact element in the message stanza.

Hi! e0ffe42b28561960c6b12b944a092794b9683a38

;gr=urn: uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6; audio;video" xmlns="http//jabber.org/protocol/contact"/> ]]>

In the above message, Bob includes his GRUU, and also the media capabilities Bob is capable of handling (audio and video). Alice sends back a message stanza containing her SIP contact information.

Hello there! e0ffe42b28561960c6b12b944a092794b9683a38

;audio;video" xmlns="http//jabber.org/protocol/contact"/> ]]>

Bob then decides to add SIP voice call to the existing XMPP conversation. He picks up Alice's contact information that Alice sent to him in a message stanza, and issues a SIP INVITE request to that URI. The XMPP-Thread carries the value of the <thread> element.

From: Bob ;tag=18593756298 Call-ID: a84b4c76e66710@ua-991.domain.com CSeq: 314159 INVITE XMPP-Thread:e0ffe42b28561960c6b12b944a092794b9683a38 Contact: ";gr=urn: uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6; audio;video" Content-Type: application/sdp Content-Length: 142 (SDP not shown) ]]> Alice responds with 200OK accepting the session invitiation request. Alice also includes the XMPP-Thread element to indicate that she has received the thread and successfully correlated the session invitation to the XMPP conversation.

;tag=c09hh1lsn From: Bob ;tag=18593756298 Call-ID: a84b4c76e66710@ua-991.domain.com CSeq: 314159 INVITE XMPP-Thread:e0ffe42b28561960c6b12b944a092794b9683a38 Contact: Content-Type: application/sdp Content-Length: 131 (SDP not shown) ]]> Bob then sends a ACK as per normal SIP procedures.

| | | | (F2) 200OK | |<----------------------| | | | (F3) ACK | |---------------------->| | | | RTP media | |<=====================>| | | | (F4) | |---------------------->| | | | (F5) | |<----------------------| ]]> Bob invites Alice to a SIP session. In the INVITE request, Bob includes the XMPP-Contact header including his XMPP JID.

From: Bob ;tag=18593756298 Call-ID: a84b4c76e66710@ua-991.domain.com CSeq: 314159 INVITE XMPP-contact: ;a84b4c76e66710@ua-991.domain.com Contact: ";gr=urn: uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6; audio;video" Content-Type: application/sdp Content-Length: 142 (SDP not shown) ]]>

;tag=c09hh1lsn From: Bob ;tag=18593756298 Call-ID: a84b4c76e66710@ua-991.domain.com CSeq: 314159 INVITE XMPP-Contact: ;a84b4c76e66710@ua-991.domain.com Contact: Content-Type: application/sdp Content-Length: 131 (SDP not shown) ]]>

Hi! e0ffe42b28561960c6b12b944a092794b9683a38

]]>

Alice sends back a message stanza copying the sip-correlation value indicating the the correlation was successful.

Hello there! e0ffe42b28561960c6b12b944a092794b9683a38

]]>

TBD

The contact and correlation iformation is sensitive and we need to prevent connection hijacking and impersonation. If the contact information that is sent over one protocol is forged, the identity verification mechanism in the other no longer help as an attacker is able to assert the false identity.

TBD