[Docs] [txt|pdf|xml|html] [Tracker] [Email] [Diff1] [Diff2] [Nits]

Versions: (draft-ibc-rtcweb-sip-websocket) 00 01 02 draft-ietf-sipcore-sip-websocket

SIPCORE Working Group                                    I. Baz Castillo
Internet-Draft                                            J. Luis Millan
Intended status: Standards Track                        XtraTelecom S.A.
Expires: July 18, 2012                                        V. Pascual
                                                             Acme Packet
                                                        January 15, 2012


    The WebSocket Protocol as a Transport for the Session Initiation
                             Protocol (SIP)
                   draft-ibc-sipcore-sip-websocket-01

Abstract

   This document specifies a WebSocket Sub-Protocol for a new transport
   in SIP (Session Initiation Protocol).  The WebSocket protocol enables
   two-way realtime communication between clients and servers.

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 July 18, 2012.

Copyright Notice

   Copyright (c) 2012 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (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



Baz Castillo, et al.      Expires July 18, 2012                 [Page 1]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  The WebSocket Protocol . . . . . . . . . . . . . . . . . . . .  5
   4.  The WebSocket SIP Sub-Protocol . . . . . . . . . . . . . . . .  6
   5.  SIP WebSocket Transport  . . . . . . . . . . . . . . . . . . .  8
     5.1.  Via Transport Parameter  . . . . . . . . . . . . . . . . .  8
     5.2.  SIP URI Transport Parameter  . . . . . . . . . . . . . . .  8
     5.3.  Sending Responses  . . . . . . . . . . . . . . . . . . . .  8
   6.  Outbound and GRUU Usage  . . . . . . . . . . . . . . . . . . . 10
   7.  Locating a SIP Server  . . . . . . . . . . . . . . . . . . . . 11
   8.  WebSocket Client Usage . . . . . . . . . . . . . . . . . . . . 12
     8.1.  WebSocket Disconnection  . . . . . . . . . . . . . . . . . 12
   9.  WebSocket Server Usage . . . . . . . . . . . . . . . . . . . . 13
     9.1.  SIP Proxy Considerations . . . . . . . . . . . . . . . . . 13
   10. Connection Keep Alive  . . . . . . . . . . . . . . . . . . . . 14
   11. Authentication . . . . . . . . . . . . . . . . . . . . . . . . 15
   12. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
     12.1. Registration . . . . . . . . . . . . . . . . . . . . . . . 16
     12.2. INVITE dialog through a proxy  . . . . . . . . . . . . . . 17
   13. Security Considerations  . . . . . . . . . . . . . . . . . . . 22
     13.1. Secure WebSocket Connection  . . . . . . . . . . . . . . . 22
     13.2. Usage of SIPS Schema . . . . . . . . . . . . . . . . . . . 22
     13.3. WebSocket Topology Hiding  . . . . . . . . . . . . . . . . 22
   14. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 24
     14.1. Registration of the WebSocket SIP Sub-Protocol . . . . . . 24
     14.2. Registration of new Via transports . . . . . . . . . . . . 24
     14.3. Registration of new SIP URI transport  . . . . . . . . . . 24
     14.4. Registration of new NAPTR service field values . . . . . . 24
   15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 25
   16. References . . . . . . . . . . . . . . . . . . . . . . . . . . 26
     16.1. Normative References . . . . . . . . . . . . . . . . . . . 26
     16.2. Informative References . . . . . . . . . . . . . . . . . . 26
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 28













Baz Castillo, et al.      Expires July 18, 2012                 [Page 2]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


1.  Introduction

   This specification defines a new WebSocket Sub-Protocol for
   transporting SIP messages between a WebSocket client and server, a
   new transport for the SIP protocol and procedures for SIP servers
   when bridging WebSocket and other SIP transports.

   This specification is focused on integrating the SIP protocol within
   client applications running a WebSocket stack.  Other aspects such as
   the usage of WebSocket as a transport between SIP servers are not
   fully covered by this specification.

      This is because WebSocket client agents are expected to be mostly
      implemented in client applications running in personal computers
      and devices as smartphones, being applications that typically are
      not able to manage TCP or UDP connections directly.  Therefore
      using WebSocket as a SIP transport between two proxies or servers
      is of little use given the fact that those servers can typically
      access to the UDP/TCP layer rather than having to use an extra
      layer on top of it.































Baz Castillo, et al.      Expires July 18, 2012                 [Page 3]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


2.  Terminology

   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].














































Baz Castillo, et al.      Expires July 18, 2012                 [Page 4]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


3.  The WebSocket Protocol

   WebSocket protocol [RFC6455] is a transport layer on top of TCP 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]
   protocol by means of a specific HTTP GET request sent by the client,
   typically a web browser, which is answered by the server (if the
   negotiation succeeded) with HTTP 101 status code.  This handshake
   procedure is designed to reuse the existing HTTP infrastructure.
   During the connection handshake, client and server agree in the
   application protocol to use on top of the WebSocket transport.  Such
   application protocol (also known as the "WebSocket Sub-Protocol")
   defines the format and semantics of the messages exchanged between
   both endpoints.  The WebSocket Sub-Protocol to be used is up to the
   application developer.  It may be a custom protocol or a standarized
   one (as the WebSocket SIP Sub-Protocol proposed in this document).
   Once the HTTP 101 response is processed both client and server reuse
   the existing TCP connection for sending application messages and
   control frames to each other in a persistent way.

   WebSocket defines message units as application data exchange for
   communication endpoints, becoming a message boundary transport layer.
   These messages can contain UTF-8 text or binary data, and can be
   splitted into various WebSocket text/binary frames.  However, the
   WebSocket API [WS-API] for web browsers just includes JavaScript
   callbacks that are invoked upon receipt of an entire message,
   regardless it has been received in a single or multiple WebSocket
   frames.


















Baz Castillo, et al.      Expires July 18, 2012                 [Page 5]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


4.  The WebSocket SIP Sub-Protocol

   The term WebSocket Sub-Protocol refers to the application-level
   protocol layered over a WebSocket connection.  This document
   specifies the WebSocket SIP Sub-Protocol for carrying SIP requests
   and responses through a WebSocket connection.

   The WebSocket client and server need to agree on this 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 WebSocket server MUST contain "sip" in its own Sec-
   WebSocket-Protocol header.

   Below is an example of the WebSocket handshake in which the client
   requests SIP Sub-Protocol support from the server:

     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

   The handshake response from the server supporting the WebSocket SIP
   Sub-Protocol would look like:

     HTTP/1.1 101 Switching Protocols
     Upgrade: websocket
     Connection: Upgrade
     Sec-WebSocket-Accept: s3pPLMBiTxaQ9kYGzzhZRbK+xOo=
     Sec-WebSocket-Protocol: sip

   Once the negotiation is done, the WebSocket connection is established
   with SIP as the WebSocket Sub-Protocol.  The WebSocket messages to be
   transmitted over this connection MUST conform to the established
   signaling protocol.

   WebSocket messages are carried on top of WebSocket UTF-8 text frames
   or binary frames.  SIP protocol [RFC3261] allows both text and binary
   bodies in SIP messages.  Therefore a client and server implementing
   the WebSocket SIP Sub-Protocol MUST accept both WebSocket text and
   binary frames.

   Each SIP message MUST be carried within a single WebSocket message
   and MUST be a complete SIP message, so a Content-Length header field



Baz Castillo, et al.      Expires July 18, 2012                 [Page 6]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


   is not mandatory.  Sending more than one SIP message within a single
   WebSocket message is not allowed, neither sending an incomplete SIP
   message.

      This makes parsing of SIP messages easier on client side
      (typically web-based applications with an strict and simple API
      for receiving WebSocket messages).  There is no need to establish
      boundaries (using Content-Length headers) between different
      messages.  Same advantage is present in other message-based SIP
      transports as UDP or SCTP [RFC4168].









































Baz Castillo, et al.      Expires July 18, 2012                 [Page 7]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


5.  SIP WebSocket Transport

   WebSocket [RFC6455] is a reliable protocol and therefore the
   WebSocket sub-protocol for a SIP transport defined by this document
   is also a reliable transport.  Thus, client and server transactions
   using WebSocket transport MUST follow the procedures and timer values
   for reliable transports as defined in [RFC3261].

5.1.  Via Transport Parameter

   Via header fields carry the transport protocol identifier.  This
   document defines the value "WS" to be used for requests over plain
   WebSocket protocol and "WSS" for requests over secure WebSocket
   protocol (in which the WebSocket connection is established on top of
   TLS [RFC5246] over TCP transport).

   The updated augmented BNF (Backus-Naur Form) [RFC5234] for this
   parameter is the following:

     transport  =  "UDP" / "TCP" / "TLS" / "SCTP" / "WS" / "WSS"
                   / other-transport

5.2.  SIP URI Transport Parameter

   This document defines the value "ws" as the transport parameter value
   for a SIP URI [RFC3986] to be contacted using WebSocket protocol as
   transport.

   The updated augmented BNF (Backus-Naur Form) [RFC5234] for this
   parameter is the following:

     transport-param  =  "transport="
                         ( "udp" / "tcp" / "sctp" / "tls" / "sctp"
                         / "ws"
                         / other-transport )

5.3.  Sending Responses

   The SIP server transport uses the value of the top Via header field
   in order to determine where to send a response.  If the "sent-
   protocol" is "WS" or "WSS" the response MUST be sent using the
   existing WebSocket connection to the source of the original request,
   if that connection is still open.  This requires the server transport
   to maintain an association between server transactions and transport
   connections.  If that connection is no longer open, the server MUST
   NOT attempt to open a WebSocket connection to the Via "sent-
   by"/"received"/"rport".  In that case the SIP server transport SHOULD
   inform the transport user of a failure in sending.



Baz Castillo, et al.      Expires July 18, 2012                 [Page 8]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


      This is due to the nature of the WebSocket protocol in which just
      the WebSocket client can establish a connection with the WebSocket
      server.  A WebSocket client does not listen for incoming
      connections.















































Baz Castillo, et al.      Expires July 18, 2012                 [Page 9]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


6.  Outbound and GRUU Usage

   WebSocket requires the client to open a TCP connection with the
   server and perform the WebSocket handshake.  A WebSocket client does
   not listen for incoming connections so it can only receive SIP
   requests from the WebSocket server it is connected to.  WebSocket
   clients may use either public or private addressing but it is
   expected that many of them will run the latter.  Unfortunately, some
   implementations may not have the ability to discover the local
   transport address which the WebSocket connection is originated from
   (e.g. a JavaScript stack within a web browser).  Those
   implementations are encouraged to create a domain consisting of a
   random token followed by .invalid top domain name, as stated in
   [RFC2606], and use it within the Via and Contact header.

   Therefore clients and servers implementing SIP over the WebSocket
   transport MUST implement the Outbound mechanism [RFC5626], being this
   the most suitable solution for SIP clients behind Network Address
   Translation (NAT) using reliable transports for contacting SIP
   servers.

   A client implementing SIP over the WebSocket transport SHOULD also
   implement GRUU [RFC5627].  The registrar responsible for the
   registration of SIP clients using the WebSocket transport SHOULD
   implement GRUU as well.

      If a REFER request is sent to a SIP User Agent indicating the
      Contact URI of a WebSocket client as the target in the Refer-To
      header field, such a URI will be reachable by the SIP UA just in
      the case it is a globally routable URI obtained from a SIP
      registrar implementing GRUU.

   Both Outbound and GRUU require the client to indicate a Uniform
   Resource Name (URN) in the "+sip.instance" parameter of the Contact
   header during the registration.  The client device is responsible for
   getting such a constant and unique value.

      In the case of web browsers it is hard to get a URN value from the
      browser itself.  This specification suggests that value is
      generated according to [RFC5626] section 4.1 by the web
      application running in the browser the first time it loads the web
      page, and then it is stored as a Cookie [RFC6265] within the
      browser data and loaded every time the same web page is visited.
      The application developer could choose any other mechanism which
      accomplishes the requirements of a URN.






Baz Castillo, et al.      Expires July 18, 2012                [Page 10]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


7.  Locating a SIP Server

   SIP entities follow normal SIP procedures in [RFC3263] to discover a
   SIP server.  This specification defines the NAPTR service value "SIP+
   D2W" for servers that support plain WebSocket transport and "SIPS+
   D2W" for servers that support secure WebSocket transport.

   A SIP entity using the WebSocket transport SHOULD perform procedures
   in [RFC3263] for the given WebSocket URI it will connect to.  If the
   WebSocket URI has "wss" schema the SIP entity MUST only consider
   "SIPS+D2W" resource records.  If the WebSocket URI does not contain a
   domain in the host part or does include a port, the SIP entity MUST
   follow procedures in [RFC6455] section 3 instead.

      Unfortunately the JavaScript stack running in web browsers cannot
      perform DNS NAPTR/SRV queries, neither the WebSocket stack running
      in web browsers can do it.  Thus, a WebSocket URI given within a
      web application needs to have a numeric network address or a
      hostname with associated DNS A/AAAA resource record(s) in its host
      part.































Baz Castillo, et al.      Expires July 18, 2012                [Page 11]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


8.  WebSocket Client Usage

   The WebSocket connection MUST be established in order to allow the
   client application to send and receive SIP requests.

      Based on local policy, this might occur once the JavaScript SIP
      application has been downloaded from the web server, or when the
      SIP user using the web browser application registers itself to a
      SIP registrar (assuming that SIP requests cannot be sent or
      received before then).

   In case the client application decides to close the WebSocket
   connection (for example when performing "logout" in a web
   application) it is recommended to remove the existing SIP
   registration binding (if present) by means of a REGISTER with
   expiration value of 0 and the associated "+sip.instance" Contact
   header parameter as per [RFC5626].

8.1.  WebSocket Disconnection

   In some circumstances the WebSocket connection could be terminated by
   the WebSocket server (for example when the server is restarted).  If
   the client application wants to become reachable again it SHOULD
   reconnect to the WebSocket server and perform a new SIP registration
   with same "+sip.instance" and "reg-id" Contact header parameters (as
   stated in [RFC5626]).

























Baz Castillo, et al.      Expires July 18, 2012                [Page 12]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


9.  WebSocket Server Usage

   How a SIP server authorizes WebSocket connection attemps from clients
   is out of the scope of this specification.  However some
   informational guidelines are provided in Section 11.  Once the
   WebSocket SIP Sub-Protocol is agreed, both client and server can send
   SIP messages to each other.

9.1.  SIP Proxy Considerations

   When a SIP proxy bridges WebSocket and any other SIP transport
   (including WebSocket transport) it MUST perform Loose Routing as
   specified in [RFC3261].  Otherwise in-dialog requests would fail
   since WebSocket clients cannot contact destinations other than their
   WebSocket server, and non-WebSocket SIP entities cannot establish a
   connection to WebSocket clients.  It is also recommended that SIP
   proxy implementations use double Record-Route techniques (as
   specified in [RFC5658]).

   In the same way, if the SIP proxy implementing the WebSocket server
   behaves as an outbound proxy for REGISTER requests, it MUST add a
   Path header field as described in [RFC3327].  Otherwise the WebSocket
   client would never receive incoming requests from the SIP registrar
   server after the lookup procedures in the SIP location service.



























Baz Castillo, et al.      Expires July 18, 2012                [Page 13]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


10.  Connection Keep Alive

   It is recommended that the WebSocket client or server keeps the
   WebSocket connection open by sending periodic WebSocket Ping frames
   as described in [RFC6455] section 5.5.2.  The decision for a
   WebSocket endpoint to maintain, or not, the connection over time is
   out of scope of this document.

   The client application MAY also use Network Address Translation (NAT)
   keep-alive mechanisms defined for the SIP protocol, such as the CRLF
   Keep-Alive Technique mechanism described in [RFC5626] section 3.5.1.
   Therefore, a SIP server implementing the WebSocket transport MUST
   support the CRLF Keep-Alive Technique.






































Baz Castillo, et al.      Expires July 18, 2012                [Page 14]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


11.  Authentication

   Prior to sending SIP requests, the WebSocket client implementing the
   SIP protocol connects to the WebSocket server and performs the
   connection handshake.  As described in Section 3 the handshake
   procedure involves an HTTP GET request replied with HTTP 101 status
   code by the server.

   In order to authorize the WebSocket connection the server MAY inspect
   the Cookie [RFC6265] header in the HTTP GET request (if present).  In
   case of web applications the value of such a Cookie is typically
   provided by the web server once the user has authenticated itself
   against the web application by following any of the multiple existing
   mechanisms.  As an alternative method, the WebSocket server could
   request Digest [RFC2617] authentication by replying a HTTP 401 status
   code.  The WebSocket protocol [RFC6455] covers this usage in section
   4.1:

      If the status code received from the server is not 101, the client
      handles the response per HTTP [RFC2616] procedures, in particular
      the client might perform authentication if it receives 401 status
      code.

   Regardless the WebSocket server requires authentication during the
   WebSocket handshake or not, authentication MAY be requested at SIP
   protocol level.  Therefore a SIP client using the WebSocket transport
   MUST implement Digest [RFC2617] authentication as stated in
   [RFC3261].























Baz Castillo, et al.      Expires July 18, 2012                [Page 15]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


12.  Examples

12.1.  Registration


   Alice    (SIP WSS)    proxy.atlanta.com
   |                             |
   |REGISTER F1                  |
   |---------------------------->|
   |200 OK F2                    |
   |<----------------------------|
   |                             |

   Alice loads a web page using her web browser and retrieves a
   JavaScript code implementing the WebSocket SIP Sub-Protocol defined
   in this document.  The JavaScript code obtained from the web server
   establishes a secure WebSocket connection with a SIP proxy/registrar
   at proxy.atlanta.com.  Upon WebSocket connection, Alice constructs
   and sends a SIP REGISTER by requesting Outbound and GRUU support.
   Since the JavaScript stack in a browser has no way to determine the
   local address from which the WebSocket connection is made, this
   implementation uses df7jal23ls0d.invalid for the Via sent-by and for
   the URI hostpart in the Contact header.

   Message details (authentication and SDP bodies are omitted for
   simplicity):

























Baz Castillo, et al.      Expires July 18, 2012                [Page 16]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


   F1 REGISTER  Alice -> proxy.atlanta.com (transport WSS)

   REGISTER sip:proxy.atlanta.com SIP/2.0
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKasudf
   From: sip:alice@atlanta.com;tag=65bnmj.34asd
   To: sip:alice@atlanta.com
   Call-ID: aiuy7k9njasd
   CSeq: 1 REGISTER
   Max-Forwards: 70
   Supported: path, outbound, gruu
   Route: <sip:proxy.atlanta.com:443;transport=ws;lr>
   Contact: <sip:alice@df7jal23ls0d.invalid;transport=ws>
     ;reg-id=1
     ;+sip.instance="<urn:uuid:f81-7dec-14a06cf1>"


   F2 200 OK  proxy.atlanta.com -> Alice (transport WSS)

   SIP/2.0 200 OK
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKasudf
   From: sip:alice@atlanta.com;tag=65bnmj.34asd
   To: sip:alice@atlanta.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@atlanta.com;gr=urn:uuid:f81-7dec-14a06cf1"
     ;temp-gruu="sip:87ash54=3dd.98a@atlanta.com;gr"
     ;expires=3600

12.2.  INVITE dialog through a proxy


















Baz Castillo, et al.      Expires July 18, 2012                [Page 17]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


   Alice    (SIP WSS)    proxy.atlanta.com    (SIP UDP)       Bob
   |                             |                             |
   |INVITE F1                    |                             |
   |---------------------------->|                             |
   |100 Trying F2                |                             |
   |<----------------------------|                             |
   |                             |INVITE F3                    |
   |                             |---------------------------->|
   |                             |200 OK F4                    |
   |                             |<----------------------------|
   |200 OK F5                    |                             |
   |<----------------------------|                             |
   |                             |                             |
   |ACK F6                       |                             |
   |---------------------------->|                             |
   |                             |ACK F7                       |
   |                             |---------------------------->|
   |                             |                             |
   |                    Both Way RTP Media                     |
   |<=========================================================>|
   |                             |                             |
   |                             |BYE F8                       |
   |                             |<----------------------------|
   |BYE F9                       |                             |
   |<----------------------------|                             |
   |200 OK F10                   |                             |
   |---------------------------->|                             |
   |                             |200 OK F11                   |
   |                             |---------------------------->|
   |                             |                             |

   In the same scenario Alice places a call to Bob's AoR by using the
   public GRUU retrieved from the registrar as Contact URI of the
   INVITE.  The WebSocket SIP server at proxy.atlanta.com acts as a SIP
   proxy routing the INVITE to the UDP location of Bob, who answers the
   call and terminates it later.

   Message details (authentication and SDP bodies are omitted for
   simplicity):


   F1 INVITE  Alice -> proxy.atlanta.com (transport WSS)

   INVITE sip:bob@atlanta.com SIP/2.0
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
   From: sip:alice@atlanta.com;tag=asdyka899
   To: sip:bob@atlanta.com
   Call-ID: asidkj3ss



Baz Castillo, et al.      Expires July 18, 2012                [Page 18]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


   CSeq: 1 INVITE
   Max-Forwards: 70
   Supported: path, outbound, gruu
   Route: <sip:proxy.atlanta.com:443;transport=ws;lr>
   Contact: <sip:alice@atlanta.com
    ;gr=urn:uuid:f81-7dec-14a06cf1;ob>"
   Content-Type: application/sdp


   F2 100 Trying  proxy.atlanta.com -> Alice (transport WSS)

   SIP/2.0 100 Trying
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
   From: sip:alice@atlanta.com;tag=asdyka899
   To: sip:bob@atlanta.com
   Call-ID: asidkj3ss
   CSeq: 1 INVITE


   F3 INVITE  proxy.atlanta.com -> Bob (transport UDP)

   INVITE sip:bob@203.0.113.22:5060 SIP/2.0
   Via: SIP/2.0/UDP proxy.atlanta.com;branch=z9hG4bKhjhjqw32c
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
   Record-Route: <sip:proxy.atlanta.com;transport=udp;lr>,
     <sip:h7kjh12s@proxy.atlanta.com:443;transport=ws;lr>
   From: sip:alice@atlanta.com;tag=asdyka899
   To: sip:bob@atlanta.com
   Call-ID: asidkj3ss
   CSeq: 1 INVITE
   Max-Forwards: 69
   Supported: path, outbound, gruu
   Contact: <sip:alice@atlanta.com
     ;gr=urn:uuid:f81-7dec-14a06cf1;ob>"
   Content-Type: application/sdp


   F4 200 OK  Bob -> proxy.atlanta.com (transport UDP)

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP proxy.atlanta.com;branch=z9hG4bKhjhjqw32c
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
   Record-Route: <sip:proxy.atlanta.com;transport=udp;lr>,
     <sip:h7kjh12s@proxy.atlanta.com:443;transport=ws;lr>
   From: sip:alice@atlanta.com;tag=asdyka899
   To: sip:bob@atlanta.com;tag=bmqkjhsd
   Call-ID: asidkj3ss
   CSeq: 1 INVITE



Baz Castillo, et al.      Expires July 18, 2012                [Page 19]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


   Max-Forwards: 69
   Contact: <sip:bob@203.0.113.22:5060;transport=udp>
   Content-Type: application/sdp


   F5 200 OK  proxy.atlanta.com -> Alice (transport WSS)

   SIP/2.0 200 OK
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
   Record-Route: <sip:proxy.atlanta.com;transport=udp;lr>,
     <sip:h7kjh12s@proxy.atlanta.com:443;transport=ws;lr>
   From: sip:alice@atlanta.com;tag=asdyka899
   To: sip:bob@atlanta.com;tag=bmqkjhsd
   Call-ID: asidkj3ss
   CSeq: 1 INVITE
   Max-Forwards: 69
   Contact: <sip:bob@203.0.113.22:5060;transport=udp>
   Content-Type: application/sdp


   F6 ACK  Alice -> proxy.atlanta.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.atlanta.com:443;transport=ws;lr>,
     <sip:proxy.atlanta.com;transport=udp;lr>,
   From: sip:alice@atlanta.com;tag=asdyka899
   To: sip:bob@atlanta.com;tag=bmqkjhsd
   Call-ID: asidkj3ss
   CSeq: 1 ACK
   Max-Forwards: 70


   F7 ACK  proxy.atlanta.com -> Bob (transport UDP)

   ACK sip:bob@203.0.113.22:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP proxy.atlanta.com;branch=z9hG4bKhwpoc80zzx
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKhgqqp090
   From: sip:alice@atlanta.com;tag=asdyka899
   To: sip:bob@atlanta.com;tag=bmqkjhsd
   Call-ID: asidkj3ss
   CSeq: 1 ACK
   Max-Forwards: 69


   F8 BYE  Bob -> proxy.atlanta.com (transport UDP)

   BYE sip:alice@atlanta.com;gr=urn:uuid:f81-7dec-14a06cf1;ob SIP/2.0



Baz Castillo, et al.      Expires July 18, 2012                [Page 20]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


   Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001
   Route: <sip:proxy.atlanta.com;transport=udp;lr>,
     <sip:h7kjh12s@proxy.atlanta.com:443;transport=ws;lr>
   From: sip:bob@atlanta.com;tag=bmqkjhsd
   To: sip:alice@atlanta.com;tag=asdyka899
   Call-ID: asidkj3ss
   CSeq: 1201 BYE
   Max-Forwards: 70


   F9 BYE  proxy.atlanta.com -> Alice (transport WSS)

   BYE sip:alice@atlanta.com;gr=urn:uuid:f81-7dec-14a06cf1;ob SIP/2.0
   Via: SIP/2.0/WSS proxy.atlanta.com:443;branch=z9hG4bKmma01m3r5
   Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001
   From: sip:bob@atlanta.com;tag=bmqkjhsd
   To: sip:alice@atlanta.com;tag=asdyka899
   Call-ID: asidkj3ss
   CSeq: 1201 BYE
   Max-Forwards: 69


   F10 200 OK  Alice -> proxy.atlanta.com (transport WSS)

   SIP/2.0 200 OK
   Via: SIP/2.0/WSS proxy.atlanta.com:443;branch=z9hG4bKmma01m3r5
   Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001
   From: sip:bob@atlanta.com;tag=bmqkjhsd
   To: sip:alice@atlanta.com;tag=asdyka899
   Call-ID: asidkj3ss
   CSeq: 1201 BYE


   F11 200 OK  proxy.atlanta.com -> Bob (transport UDP)

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001
   From: sip:bob@atlanta.com;tag=bmqkjhsd
   To: sip:alice@atlanta.com;tag=asdyka899
   Call-ID: asidkj3ss
   CSeq: 1201 BYE










Baz Castillo, et al.      Expires July 18, 2012                [Page 21]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


13.  Security Considerations

13.1.  Secure WebSocket Connection

   It is recommended to protect the privacy of the SIP traffic through
   the WebSocket communication by using a secure WebSocket connection
   (tunneled over TLS [RFC5246]).  For this, the client application MUST
   be provided with a secure "wss" WebSocket URI.

13.2.  Usage of SIPS Schema

   SIPS schema within a SIP request dictates that the entire request
   path to the target be secured.  If such a path includes a WebSocket
   connection it MUST be a secure WebSocket connection (tunneled over
   TLS [RFC5246]) opened with a "wss" WebSocket URI.

13.3.  WebSocket Topology Hiding

   RFC 3261 [RFC3261] section 18.2.1 "Receiving Requests" states the
   following:

      When the server transport receives a request over any transport,
      it MUST examine the value of the "sent-by" parameter in the top
      Via header field value.  If the host portion of the "sent-by"
      parameter contains a domain name, or if it contains an IP address
      that differs from the packet source address, the server MUST add a
      "received" parameter to that Via header field value.  This
      parameter MUST contain the source address from which the packet
      was received.

   The requirement of adding the "received" parameter does not fit well
   into WebSocket protocol nature.  The WebSocket handshake connection
   reuses existing HTTP infrastructure in which there could be certain
   number of HTTP proxies and/or TCP load balancers between the client
   and the WebSocket server, so the source IP the server would write
   into the Via "received" parameter would be the IP of the HTTP/TCP
   intermediary in front of it.  This would reveal sensitive information
   about the internal topology of the provider network to the WebSocket
   client.

   Thus, given the fact that SIP responses can only be sent over the
   existing WebSocket connection, the meaning of the Via "received"
   parameter added by the server is of little use.  Therefore, in order
   to allow hiding possible sensitive information about the provider
   infrastructure, this specification relaxes the requirement in RFC
   3261 [RFC3261] section 18.2.1 "Receiving Requests" by stating that a
   WebSocket server receiving a SIP request from a WebSocket client MAY
   choose not to add the Via "received" parameter nor honor the Via



Baz Castillo, et al.      Expires July 18, 2012                [Page 22]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


   "rport" [RFC3581] parameter.  A SIP client implementing the WebSocket
   transport MUST be ready to receive SIP responses in which the topmost
   Via header field does not contain the "received" and "rport"
   parameters.















































Baz Castillo, et al.      Expires July 18, 2012                [Page 23]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


14.  IANA Considerations

14.1.  Registration of the WebSocket SIP Sub-Protocol

   This specification requests IANA to create the WebSocket SIP Sub-
   Protocol in the registry of WebSocket sub-protocols with the
   following data:

   Subprotocol Identifier:  sip

   Subprotocol Common Name:  SIP over WebSocket

   Subprotocol Definition:  TBD, it should point to this document

14.2.  Registration of new Via transports

   This specification registers two new transport identifiers for Via
   headers:

   WS:   MUST be used when constructing a SIP request to be sent over a
         plain WebSocket connection.

   WSS:  MUST be used when constructing a SIP request to be sent over a
         secure WebSocket connection (tunneled over TLS [RFC5246]).

14.3.  Registration of new SIP URI transport

   This specification registers a new value for the "transport"
   parameter in a SIP URI:

   ws:   Identifies a SIP URI to be contacted using a WebSocket
         connection.

14.4.  Registration of new NAPTR service field values

   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 SIP SRV Resource Record Services Field".  The
   resulting entries are as follows:

    Services Field        Protocol  Reference
    --------------------  --------  ---------
    SIP+D2W               WS        TBD: this document
    SIPS+D2W              WSS       TBD: this document







Baz Castillo, et al.      Expires July 18, 2012                [Page 24]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


15.  Acknowledgements

   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, Ranjit
   Avasarala.

   Special thanks to Saul Ibarra Corretge for his detailed review and
   provided suggestions.

   Special thanks also to Aranzazu Ruiz for her valuable collaboration
   in the whole document.






































Baz Castillo, et al.      Expires July 18, 2012                [Page 25]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


16.  References

16.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC2617]  Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
              Leach, P., Luotonen, A., and L. Stewart, "HTTP
              Authentication: Basic and Digest Access Authentication",
              RFC 2617, June 1999.

   [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.

   [RFC5234]  Crocker, D. and P. Overell, "Augmented BNF for Syntax
              Specifications: ABNF", STD 68, RFC 5234, January 2008.

   [RFC5626]  Jennings, C., Mahy, R., and F. Audet, "Managing Client-
              Initiated Connections in the Session Initiation Protocol
              (SIP)", RFC 5626, October 2009.

   [RFC5627]  Rosenberg, J., "Obtaining and Using Globally Routable User
              Agent URIs (GRUUs) in the Session Initiation Protocol
              (SIP)", RFC 5627, October 2009.

   [RFC6455]  Fette, I. and A. Melnikov, "The WebSocket Protocol",
              RFC 6455, December 2011.

16.2.  Informative References

   [RFC2606]  Eastlake, D. and A. Panitz, "Reserved Top Level DNS
              Names", BCP 32, RFC 2606, June 1999.

   [RFC2616]  Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
              Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext
              Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999.

   [RFC3327]  Willis, D. and B. Hoeneisen, "Session Initiation Protocol
              (SIP) Extension Header Field for Registering Non-Adjacent
              Contacts", RFC 3327, December 2002.




Baz Castillo, et al.      Expires July 18, 2012                [Page 26]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


   [RFC3581]  Rosenberg, J. and H. Schulzrinne, "An Extension to the
              Session Initiation Protocol (SIP) for Symmetric Response
              Routing", RFC 3581, August 2003.

   [RFC3986]  Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
              Resource Identifier (URI): Generic Syntax", STD 66,
              RFC 3986, January 2005.

   [RFC4168]  Rosenberg, J., Schulzrinne, H., and G. Camarillo, "The
              Stream Control Transmission Protocol (SCTP) as a Transport
              for the Session Initiation Protocol (SIP)", RFC 4168,
              October 2005.

   [RFC5246]  Dierks, T. and E. Rescorla, "The Transport Layer Security
              (TLS) Protocol Version 1.2", RFC 5246, August 2008.

   [RFC5658]  Froment, T., Lebel, C., and B. Bonnaerens, "Addressing
              Record-Route Issues in the Session Initiation Protocol
              (SIP)", RFC 5658, October 2009.

   [RFC6265]  Barth, A., "HTTP State Management Mechanism", RFC 6265,
              April 2011.

   [WS-API]   Hickson, I., "The Web Sockets API", September 2010.



























Baz Castillo, et al.      Expires July 18, 2012                [Page 27]


Internet-Draft      WebSocket as a Transport for SIP        January 2012


Authors' Addresses

   Inaki Baz Castillo
   XtraTelecom S.A.
   Barakaldo, Basque Country
   Spain

   Email: ibc@aliax.net


   Jose Luis Millan
   XtraTelecom S.A.
   Bilbao, Basque Country
   Spain

   Email: jmillan@aliax.net


   Victor Pascual
   Acme Packet
   Anabel Segura 10
   Madrid, Madrid  28108
   Spain

   Email: vpascual@acmepacket.com


























Baz Castillo, et al.      Expires July 18, 2012                [Page 28]


Html markup produced by rfcmarkup 1.122, available from https://tools.ietf.org/tools/rfcmarkup/