| SIPCORE Working Group | I.B.C. Baz Castillo |
| Internet-Draft | J.L.M.V. Millan Villegas |
| Updates: 3261 (if approved) | Versatica |
| Intended status: Standards Track | V.P. Pascual |
| Expires: September 02, 2013 | Acme Packet |
| March 2013 |
The WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP)
draft-ietf-sipcore-sip-websocket-08
The WebSocket protocol enables two-way realtime communication between clients and servers in web-based applications. This document specifies a WebSocket sub-protocol as a reliable transport mechanism between SIP (Session Initiation Protocol) entities to enable usage of SIP in web-oriented deployments. This document normatively updates RFC 3261.
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 September 02, 2013.
Copyright (c) 2013 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 (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.
The WebSocket [RFC6455] protocol enables message exchange between clients and servers on top of a persistent TCP connection (optionally secured with TLS [RFC5246]). The initial protocol handshake makes use of HTTP [RFC2616] semantics, allowing the WebSocket protocol to reuse existing HTTP infrastructure.
Modern web browsers include a WebSocket client stack complying with the WebSocket API [WS-API] as specified by the W3C. It is expected that other client applications (those running in personal computers and devices such as smartphones) will also make a WebSocket client stack available. The specification in this document enables usage of SIP in these scenarios.
This specification defines a WebSocket sub-protocol (as defined in section 1.9 in [RFC6455]) for transporting SIP messages between a WebSocket client and server, a reliable and message-boundary preserving transport for SIP, DNS NAPTR [RFC3403] service values and procedures for SIP entities implementing the WebSocket transport. Media transport is out of the scope of this document.
Section 3 in this specification relaxes the requirement in [RFC3261] by which the SIP server transport MUST add a "received" parameter in the top Via header in certain circumstances.
All diagrams, examples, and notes in this specification are non-normative, as are all sections explicitly marked non-normative. Everything else in this specification is normative.
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 [RFC2119].
_This section is non-normative._
The WebSocket protocol [RFC6455] is a transport layer on top of TCP (optionally secured with TLS [RFC5246]) in which both client and server exchange message units in both directions. The protocol defines a connection handshake, WebSocket sub-protocol and extensions negotiation, a frame format for sending application and control data, a masking mechanism, and status codes for indicating disconnection causes.
The WebSocket connection handshake is based on HTTP [RFC2616] and utilizes the HTTP GET method with an "Upgrade" request. This is sent by the client and then answered by the server (if the negotiation succeeded) with an HTTP 101 status code. Once the handshake is completed the connection upgrades from HTTP to the WebSocket protocol. This handshake procedure is designed to reuse the existing HTTP infrastructure. During the connection handshake, client and server agree on the application protocol to use on top of the WebSocket transport. Such application protocol (also known as a "WebSocket sub-protocol") defines the format and semantics of the messages exchanged by the endpoints. This could be a custom protocol or a standardized one (as the WebSocket SIP sub-protocol defined in this document). Once the HTTP 101 response is processed both client and server reuse the underlying TCP connection for sending WebSocket messages and control frames to each other. Unlike plain HTTP, this connection is persistent and can be used for multiple message exchanges.
WebSocket defines message units to be used by applications for the exchange of data, so it provides a message boundary-preserving transport layer. These message units can contain either UTF-8 text or binary data, and can be split into multiple WebSocket text/binary transport frames as needed by the WebSocket stack.
The term WebSocket sub-protocol refers to an application-level protocol layered on top of a WebSocket connection. This document specifies the WebSocket SIP sub-protocol for carrying SIP requests and responses through a WebSocket connection.
The SIP WebSocket Client and SIP WebSocket Server negotiate usage of the WebSocket SIP sub-protocol during the WebSocket handshake procedure as defined in section 1.3 of [RFC6455]. The Client MUST include the value "sip" in the Sec-WebSocket-Protocol header in its handshake request. The 101 reply from the Server MUST contain "sip" in its corresponding Sec-WebSocket-Protocol header.
GET / HTTP/1.1 Host: sip-ws.example.com Upgrade: websocket Connection: Upgrade Sec-WebSocket-Key: dGhlIHNhbXBsZSBub25jZQ== Origin: http://www.example.com Sec-WebSocket-Protocol: sip Sec-WebSocket-Version: 13
Below is an example of a WebSocket handshake in which the Client requests the WebSocket SIP sub-protocol support from the Server:
HTTP/1.1 101 Switching Protocols Upgrade: websocket Connection: Upgrade Sec-WebSocket-Accept: s3pPLMBiTxaQ9kYGzzhZRbK+xOo= Sec-WebSocket-Protocol: sip
The handshake response from the Server accepting the WebSocket SIP sub-protocol would look as follows:
Once the negotiation has been completed, the WebSocket connection is established and can be used for the transport of SIP requests and responses. The WebSocket messages transmitted over this connection MUST conform to the negotiated WebSocket sub-protocol.
WebSocket messages can be transported in either UTF-8 text frames or binary frames. SIP [RFC3261] allows both text and binary bodies in SIP requests and responses. Therefore SIP WebSocket Clients and SIP WebSocket Servers MUST accept both text and binary frames.
WebSocket [RFC6455] is a reliable protocol and therefore the SIP WebSocket sub-protocol defined by this document is a reliable SIP transport. Thus, client and server transactions using WebSocket for transport MUST follow the procedures and timer values for reliable transports as defined in [RFC3261].
Each SIP message MUST be carried within a single WebSocket message, and a WebSocket message MUST NOT contain more than one SIP message. Because the WebSocket transport preserves message boundaries, the use of the Content-Length header in SIP messages is optional when they are transported using the WebSocket sub-protocol.
Via header fields in SIP messages carry a transport protocol identifier. This document defines the value "WS" to be used for requests over plain WebSocket connections and "WSS" for requests over secure WebSocket connections (in which the WebSocket connection is established using TLS [RFC5246] with TCP transport).
transport =/ "WS" / "WSS"
The updated augmented BNF (Backus-Naur Form) [RFC5234] for this parameter is the following (the original BNF for this parameter can be found in [RFC3261], which was then updated by [RFC4168]):
This document defines the value "ws" as the transport parameter value for a SIP URI [RFC3986] to be contacted using the SIP WebSocket sub-protocol as transport.
transport-param =/ "transport=" "ws"
The updated augmented BNF (Backus-Naur Form) for this parameter is the following (the original BNF for this parameter can be found in [RFC3261], which was then updated by [RFC4168]):
[RFC3261] section 18.2.1 "Receiving Requests" states the following:
The requirement of adding the "received" parameter does not fit well into the WebSocket protocol design. The WebSocket connection handshake reuses existing HTTP infrastructure in which there could be an unknown number of HTTP proxies and/or TCP load balancers between the SIP WebSocket Client and Server, so the source address the server would write into the Via "received" parameter would be the address of the HTTP/TCP intermediary in front of it. This could reveal sensitive information about the internal topology of the Server's network to the Client.
Given the fact that SIP responses can only be sent over the existing WebSocket connection, the Via "received" parameter is of little use. Therefore, in order to allow hiding possible sensitive information about the SIP WebSocket Server's network, this document updates [RFC3261] section 18.2.1 by stating:
[RFC3261] section 18 "Transport" states the following:
The specification of this transport enables SIP to be used as a session establishment protocol in scenarios where none of other transport protocols defined for SIP can be used. Since some environments do not enable SIP elements to use UDP and TCP as SIP transport protocols, a SIP element acting as a SIP WebSocket Client is not mandated to implement support of UDP and TCP and thus MAY just implement the WebSocket transport defined by this specification.
[RFC3263] specifies the procedures which should be followed by SIP entities for locating SIP servers. This specification defines the NAPTR service value "SIP+D2W" for SIP WebSocket Servers that support plain WebSocket connections and "SIPS+D2W" for SIP WebSocket Servers that support secure WebSocket connections.
In the absence of DNS SRV resource records or an explicit port, the default port for a SIP URI using the "sip" scheme and the "ws" transport parameter is 80, and the default port for a SIP URI using the "sips" scheme and the "ws" transport parameter is 443.
_This section is non-normative._
SIP WebSocket Clients and Servers may keep their WebSocket connections open by sending periodic WebSocket "Ping" frames as described in [RFC6455] section 5.5.2.
The indication and use of the CRLF NAT keep-alive mechanism defined for SIP connection-oriented transports in [RFC5626] section 3.5.1 or [RFC6223] are, of course, usable over the transport defined in this specification.
_This section is non-normative._
This section describes how authentication is achieved through the requirements in [RFC6455], [RFC6265] and [RFC3261].
Prior to sending SIP requests, a SIP WebSocket Client connects to a SIP WebSocket Server and performs the connection handshake. As described in Section 3 the handshake procedure involves a HTTP GET method request from the Client and a response from the Server including an HTTP 101 status code.
In order to authorize the WebSocket connection, the SIP WebSocket Server is allowed to inspect any Cookie [RFC6265] headers present in the HTTP GET request. For many web applications the value of such a Cookie is provided by the web server once the user has authenticated themselves to the web server, which could be done by many existing mechanisms. As an alternative method, the SIP WebSocket Server could request HTTP authentication by replying to the Client's GET method request with a HTTP 401 status code. The WebSocket protocol [RFC6455] covers this usage in section 4.1:
Regardless of whether the SIP WebSocket Server requires authentication during the WebSocket handshake, authentication can be requested at SIP protocol level. Note that RFC 3261 requires that all SIP implementations (which includes implementations of this specification) implement Digest Authorization ([RFC3261] section 26.3.1).
Alice (SIP WSS) proxy.example.com | | |HTTP GET (WS handshake) F1 | |---------------------------->| |101 Switching Protocols F2 | |<----------------------------| | | |REGISTER F3 | |---------------------------->| |200 OK F4 | |<----------------------------| | |
Alice loads a web page using her web browser and retrieves JavaScript code implementing the WebSocket SIP sub-protocol defined in this document. The JavaScript code (a SIP WebSocket Client) establishes a secure WebSocket connection with a SIP proxy/registrar (a SIP WebSocket Server) at proxy.example.com. Upon WebSocket connection, Alice constructs and sends a SIP REGISTER request including Outbound and GRUU support. Since the JavaScript stack in a browser has no way to determine the local address from which the WebSocket connection was made, this implementation uses a random ".invalid" domain name for the Via header sent-by parameter and for the hostport of the URI in the Contact header (see Appendix A.1).
Message details (authentication and SDP bodies are omitted for simplicity):
F1 HTTP GET (WS handshake) Alice -> proxy.example.com (TLS) GET / HTTP/1.1 Host: proxy.example.com Upgrade: websocket Connection: Upgrade Sec-WebSocket-Key: dGhlIHNhbXBsZSBub25jZQ== Origin: https://www.example.com Sec-WebSocket-Protocol: sip Sec-WebSocket-Version: 13 F2 101 Switching Protocols proxy.example.com -> Alice (TLS) HTTP/1.1 101 Switching Protocols Upgrade: websocket Connection: Upgrade Sec-WebSocket-Accept: s3pPLMBiTxaQ9kYGzzhZRbK+xOo= Sec-WebSocket-Protocol: sip F3 REGISTER Alice -> proxy.example.com (transport WSS) REGISTER sip:proxy.example.com SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKasudf From: sip:alice@example.com;tag=65bnmj.34asd To: sip:alice@example.com Call-ID: aiuy7k9njasd CSeq: 1 REGISTER Max-Forwards: 70 Supported: path, outbound, gruu Contact: <sip:alice@df7jal23ls0d.invalid;transport=ws> ;reg-id=1 ;+sip.instance="<urn:uuid:f81-7dec-14a06cf1>" F4 200 OK proxy.example.com -> Alice (transport WSS) SIP/2.0 200 OK Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKasudf From: sip:alice@example.com;tag=65bnmj.34asd To: sip:alice@example.com;tag=12isjljn8 Call-ID: aiuy7k9njasd CSeq: 1 REGISTER Supported: outbound, gruu Contact: <sip:alice@df7jal23ls0d.invalid;transport=ws> ;reg-id=1 ;+sip.instance="<urn:uuid:f81-7dec-14a06cf1>" ;pub-gruu="sip:alice@example.com;gr=urn:uuid:f81-7dec-14a06cf1" ;temp-gruu="sip:87ash54=3dd.98a@example.com;gr" ;expires=3600
Alice (SIP WSS) proxy.example.com (SIP UDP) Bob | | | |INVITE F1 | | |---------------------------->| | |100 Trying F2 | | |<----------------------------| | | |INVITE F3 | | |---------------------------->| | |200 OK F4 | | |<----------------------------| |200 OK F5 | | |<----------------------------| | | | | |ACK F6 | | |---------------------------->| | | |ACK F7 | | |---------------------------->| | | | | Bidirectional RTP Media | |<=========================================================>| | | | | |BYE F8 | | |<----------------------------| |BYE F9 | | |<----------------------------| | |200 OK F10 | | |---------------------------->| | | |200 OK F11 | | |---------------------------->| | | |
In the same scenario Alice places a call to Bob's AoR (Address Of Record). The SIP WebSocket Server at proxy.example.com acts as a SIP proxy, routing the INVITE to Bob's contact address (which happens to be using SIP transported over UDP). Bob answers the call and then terminates it.
Message details (authentication and SDP bodies are omitted for simplicity):
F1 INVITE Alice -> proxy.example.com (transport WSS) INVITE sip:bob@example.com SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks From: sip:alice@example.com;tag=asdyka899 To: sip:bob@example.com Call-ID: asidkj3ss CSeq: 1 INVITE Max-Forwards: 70 Supported: path, outbound, gruu Route: <sip:proxy.example.com:443;transport=ws;lr> Contact: <sip:alice@example.com ;gr=urn:uuid:f81-7dec-14a06cf1;ob> Content-Type: application/sdp F2 100 Trying proxy.example.com -> Alice (transport WSS) SIP/2.0 100 Trying Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks From: sip:alice@example.com;tag=asdyka899 To: sip:bob@example.com Call-ID: asidkj3ss CSeq: 1 INVITE F3 INVITE proxy.example.com -> Bob (transport UDP) INVITE sip:bob@203.0.113.22:5060 SIP/2.0 Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bKhjhjqw32c Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks Record-Route: <sip:proxy.example.com;transport=udp;lr>, <sip:h7kjh12s@proxy.example.com:443;transport=ws;lr> From: sip:alice@example.com;tag=asdyka899 To: sip:bob@example.com Call-ID: asidkj3ss CSeq: 1 INVITE Max-Forwards: 69 Supported: path, outbound, gruu Contact: <sip:alice@example.com ;gr=urn:uuid:f81-7dec-14a06cf1;ob> Content-Type: application/sdp F4 200 OK Bob -> proxy.example.com (transport UDP) SIP/2.0 200 OK Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bKhjhjqw32c ;received=192.0.2.10 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks Record-Route: <sip:proxy.example.com;transport=udp;lr>, <sip:h7kjh12s@proxy.example.com:443;transport=ws;lr> From: sip:alice@example.com;tag=asdyka899 To: sip:bob@example.com;tag=bmqkjhsd Call-ID: asidkj3ss CSeq: 1 INVITE Contact: <sip:bob@203.0.113.22:5060;transport=udp> Content-Type: application/sdp F5 200 OK proxy.example.com -> Alice (transport WSS) SIP/2.0 200 OK Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks Record-Route: <sip:proxy.example.com;transport=udp;lr>, <sip:h7kjh12s@proxy.example.com:443;transport=ws;lr> From: sip:alice@example.com;tag=asdyka899 To: sip:bob@example.com;tag=bmqkjhsd Call-ID: asidkj3ss CSeq: 1 INVITE Contact: <sip:bob@203.0.113.22:5060;transport=udp> Content-Type: application/sdp F6 ACK Alice -> proxy.example.com (transport WSS) ACK sip:bob@203.0.113.22:5060;transport=udp SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKhgqqp090 Route: <sip:h7kjh12s@proxy.example.com:443;transport=ws;lr>, <sip:proxy.example.com;transport=udp;lr>, From: sip:alice@example.com;tag=asdyka899 To: sip:bob@example.com;tag=bmqkjhsd Call-ID: asidkj3ss CSeq: 1 ACK Max-Forwards: 70 F7 ACK proxy.example.com -> Bob (transport UDP) ACK sip:bob@203.0.113.22:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bKhwpoc80zzx Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKhgqqp090 From: sip:alice@example.com;tag=asdyka899 To: sip:bob@example.com;tag=bmqkjhsd Call-ID: asidkj3ss CSeq: 1 ACK Max-Forwards: 69 F8 BYE Bob -> proxy.example.com (transport UDP) BYE sip:alice@example.com;gr=urn:uuid:f81-7dec-14a06cf1;ob SIP/2.0 Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001 Route: <sip:proxy.example.com;transport=udp;lr>, <sip:h7kjh12s@proxy.example.com:443;transport=ws;lr> From: sip:bob@example.com;tag=bmqkjhsd To: sip:alice@example.com;tag=asdyka899 Call-ID: asidkj3ss CSeq: 1201 BYE Max-Forwards: 70 F9 BYE proxy.example.com -> Alice (transport WSS) BYE sip:alice@example.com;gr=urn:uuid:f81-7dec-14a06cf1;ob SIP/2.0 Via: SIP/2.0/WSS proxy.example.com:443;branch=z9hG4bKmma01m3r5 Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001 From: sip:bob@example.com;tag=bmqkjhsd To: sip:alice@example.com;tag=asdyka899 Call-ID: asidkj3ss CSeq: 1201 BYE Max-Forwards: 69 F10 200 OK Alice -> proxy.example.com (transport WSS) SIP/2.0 200 OK Via: SIP/2.0/WSS proxy.example.com:443;branch=z9hG4bKmma01m3r5 Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001 From: sip:bob@example.com;tag=bmqkjhsd To: sip:alice@example.com;tag=asdyka899 Call-ID: asidkj3ss CSeq: 1201 BYE F11 200 OK proxy.example.com -> Bob (transport UDP) SIP/2.0 200 OK Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001 From: sip:bob@example.com;tag=bmqkjhsd To: sip:alice@example.com;tag=asdyka899 Call-ID: asidkj3ss CSeq: 1201 BYE
It is recommended that the SIP traffic transported over a WebSocket communication be protected by using a secure WebSocket connection (using TLS [RFC5246] over TCP).
However none of the SIP TLS certificate checks specified in [RFC3261] or [RFC5922] will be made when using SIP over secure WebSocket transport. Instead, only the checks specified by [RFC6455] will be made. The certificates that are appropriate for SIP over TLS over TCP will probably not be appropriate for SIP over secure WebSocket connections.
The SIPS scheme in a SIP URI dictates that the entire request path to the target be secure. If such a path includes a WebSocket connection it MUST be a secure WebSocket connection.
RFC Editor Note: Please set the RFC number assigned for this document in the sub-sections below and remove this note.
This specification requests IANA to register the WebSocket SIP sub-protocol under the "WebSocket Subprotocol Name" Registry with the following data:
Services Field Protocol Reference -------------- -------- --------- SIP+D2W WS TBD: this document SIPS+D2W WS TBD: this document
This document defines two new NAPTR service field values (SIP+D2W and SIPS+D2W) and requests IANA to register these values under the "Registry for the Session Initiation Protocol (SIP) NAPTR Resource Record Services Field". The resulting entries are as follows:
Parameter Name Predefined Values Reference -------------- ----------------- --------- transport Yes [RFC3261][TBD: this document]
This specification requests IANA to add a reference to this document under the "SIP/SIPS URI Parameters" Sub-Registry within the "Session Initiation Protocol (SIP) Parameters" Registry:
Header Name compact Reference ----------- ------- --------- Via v [RFC3261][TBD: this document]
This specification requests IANA to add a reference to this document under the "Header Fields" Sub-Registry within the "Session Initiation Protocol (SIP) Parameters" Registry:
Predefined
Header Field Parameter Name Values Reference
------------ -------------- ------ ---------
Via received No [RFC3261][TBD: this document]
This specification requests IANA to add a reference to this document under the "Header Field Parameters and Parameter Values" Sub-Registry within the "Session Initiation Protocol (SIP) Parameters" Registry:
+------------+------------------------+ | Transport | Reference | +------------+------------------------+ | UDP | [RFC 3261] | | TCP | [RFC 3261] | | TLS | [RFC 3261] | | SCTP | [RFC 3261], [RFC 4168] | | TLS-SCTP | [RFC 4168] | | WS | [TBD: this document] | | WSS | [TBD: this document] | +------------+------------------------+
This document adds a new registry, "SIP Transport", to the "Session Initiation Protocol (SIP) Parameters" Registry. Its format and initial values are as shown in the following table:
The policy for registration of values in this registry is "Standards Action", as that term is defined by [RFC5226].
Special thanks to the following people who participated in discussions on the SIPCORE and RTCWEB WG mailing lists and contributed ideas and/or provided detailed reviews (the list is likely to be incomplete): Hadriel Kaplan, Paul Kyzivat, Robert Sparks, Adam Roach, Ranjit Avasarala, Xavier Marjou, Nataraju A. B., Martin Vopatek, Alexey Melnikov, Alan Johnston, Christer Holmberg, Salvatore Loreto, Kevin P. Fleming (complete grammatical review), Saul Ibarra Corretge.
| [RFC2119] | Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. |
| [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. |
| [RFC3263] | Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol (SIP): Locating SIP Servers", RFC 3263, June 2002. |
| [RFC3403] | Mealling, M., "Dynamic Delegation Discovery System (DDDS) Part Three: The Domain Name System (DNS) Database", RFC 3403, October 2002. |
| [RFC5234] | Crocker, D. and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", STD 68, RFC 5234, January 2008. |
| [RFC5226] | Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA Considerations Section in RFCs", BCP 26, RFC 5226, May 2008. |
| [RFC6455] | Fette, I. and A. Melnikov, "The WebSocket Protocol", RFC 6455, December 2011. |
_This section is non-normative._
Let us assume a scenario in which the users access with their web browsers (probably behind NAT) an application provided by a server on an intranet, login by entering their user identifier and credentials, and retrieve a JavaScript application (along with the HTML) implementing a SIP WebSocket Client.
Such a SIP stack connects to a given SIP WebSocket Server (an outbound SIP proxy which also implements classic SIP transports such as UDP and TCP). The HTTP GET method request sent by the web browser for the WebSocket handshake includes a Cookie [RFC6265] header with the value previously provided by the server after the successful login procedure. The Cookie value is then inspected by the WebSocket server to authorize the connection. Once the WebSocket connection is established, the SIP WebSocket Client performs a SIP registration to a SIP registrar server that is reachable through the proxy. After registration, the SIP WebSocket Client and Server exchange SIP messages as would normally be expected.
This scenario is quite similar to ones in which SIP UAs behind NATs connect to a proxy and must reuse the same TCP connection for incoming requests (because they are not directly reachable by the proxy otherwise). In both cases, the SIP UAs are only reachable through the proxy they are connected to.
The SIP Outbound extension [RFC5626] seems an appropriate solution for this scenario. Therefore these SIP WebSocket Clients and the SIP registrar implement both the Outbound and Path [RFC3327] extensions, and the SIP proxy acts as an Outbound Edge Proxy (as defined in [RFC5626] section 3.4).
SIP WebSocket Clients in this scenario receive incoming SIP requests via the SIP WebSocket Server they are connected to. Therefore, in some call transfer cases the usage of GRUU [RFC5627] (which should be implemented in both the SIP WebSocket Clients and SIP registrar) is valuable.
The JavaScript stack in web browsers does not have the ability to discover the local transport address used for originating WebSocket connections. A SIP WebSocket client running in such an environment can construct a domain name consisting of a random token followed by the ".invalid" top-level domain name, as stated in [RFC2606], and uses it within its Via and Contact headers.
Both the Outbound and GRUU specifications require a SIP UA to include a Uniform Resource Name (URN) in a "+sip.instance" parameter of the Contact header they include their SIP REGISTER requests. The client device is responsible for generating or collecting a suitable value for this purpose.
The SIP WebSocket Server in this scenario behaves as a SIP Outbound Edge Proxy, which involves support for Outbound [RFC5626] and Path [RFC3327].
The proxy performs Loose Routing and remains in the path of dialogs as specified in [RFC3261]. If it did not do this, in-dialog requests would fail since SIP WebSocket Clients make use of their SIP WebSocket Server in order to send and receive SIP messages.