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Versions: (draft-ibc-sipcore-sip-websocket) 00 01 02 03 04 05 06 08 09 10 RFC 7118

SIPCORE Working Group                                    I. Baz Castillo
Internet-Draft                                        J. Millan Villegas
Updates: 3261 (if approved)                                    Versatica
Intended status: Standards Track                              V. Pascual
Expires: September 12, 2013                                  Acme Packet
                                                          March 11, 2013


    The WebSocket Protocol as a Transport for the Session Initiation
                             Protocol (SIP)
                  draft-ietf-sipcore-sip-websocket-08

Abstract

   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.

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 September 12, 2013.

Copyright Notice

   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



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   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  3
     2.1.  Definitions  . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  The WebSocket Protocol . . . . . . . . . . . . . . . . . . . .  4
   4.  The WebSocket SIP Sub-Protocol . . . . . . . . . . . . . . . .  4
     4.1.  Handshake  . . . . . . . . . . . . . . . . . . . . . . . .  5
     4.2.  SIP encoding . . . . . . . . . . . . . . . . . . . . . . .  5
   5.  SIP WebSocket Transport  . . . . . . . . . . . . . . . . . . .  5
     5.1.  General  . . . . . . . . . . . . . . . . . . . . . . . . .  6
     5.2.  Updates to RFC 3261  . . . . . . . . . . . . . . . . . . .  6
       5.2.1.  Via Transport Parameter  . . . . . . . . . . . . . . .  6
       5.2.2.  SIP URI Transport Parameter  . . . . . . . . . . . . .  6
       5.2.3.  Via received parameter . . . . . . . . . . . . . . . .  7
       5.2.4.  SIP transport implementation requirements  . . . . . .  7
     5.3.  Locating a SIP Server  . . . . . . . . . . . . . . . . . .  8
   6.  Connection Keep-Alive  . . . . . . . . . . . . . . . . . . . .  8
   7.  Authentication . . . . . . . . . . . . . . . . . . . . . . . .  8
   8.  Examples . . . . . . . . . . . . . . . . . . . . . . . . . . .  9
     8.1.  Registration . . . . . . . . . . . . . . . . . . . . . . .  9
     8.2.  INVITE dialog through a proxy  . . . . . . . . . . . . . . 11
   9.  Security Considerations  . . . . . . . . . . . . . . . . . . . 15
     9.1.  Secure WebSocket Connection  . . . . . . . . . . . . . . . 15
     9.2.  Usage of SIPS Scheme . . . . . . . . . . . . . . . . . . . 16
   10. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 16
     10.1. Registration of the WebSocket SIP Sub-Protocol . . . . . . 16
     10.2. Registration of new NAPTR service field values . . . . . . 16
     10.3. SIP/SIPS URI Parameters Sub-Registry . . . . . . . . . . . 17
     10.4. Header Fields Sub-Registry . . . . . . . . . . . . . . . . 17
     10.5. Header Field Parameters and Parameter Values
           Sub-Registry . . . . . . . . . . . . . . . . . . . . . . . 17
     10.6. SIP Transport Sub-Registry . . . . . . . . . . . . . . . . 17
   11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 18
   12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
     12.1. Normative References . . . . . . . . . . . . . . . . . . . 18
     12.2. Informative References . . . . . . . . . . . . . . . . . . 19
   Appendix A.  Implementation Guidelines . . . . . . . . . . . . . . 20
     A.1.  SIP WebSocket Client Considerations  . . . . . . . . . . . 21
     A.2.  SIP WebSocket Server Considerations  . . . . . . . . . . . 21
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 21




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1.  Introduction

   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.


2.  Terminology

   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].

2.1.  Definitions

   SIP WebSocket Client:  A SIP entity capable of opening outbound
         connections to WebSocket servers and communicating using the
         WebSocket SIP sub-protocol as defined by this document.

   SIP WebSocket Server:  A SIP entity capable of listening for inbound
         connections from WebSocket clients and communicating using the
         WebSocket SIP sub-protocol as defined by this document.





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3.  The WebSocket Protocol

   _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 WebSocket API [WS-API] for web browsers only defines callbacks
      to be invoked upon receipt of an entire message unit, regardless
      of whether it was received in a single Websocket frame or split
      across multiple frames.


4.  The WebSocket SIP Sub-Protocol

   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.



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4.1.  Handshake

   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.

   Below is an example of a WebSocket handshake in which the Client
   requests the WebSocket 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 accepting the WebSocket SIP
   sub-protocol would look as follows:

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

   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.

4.2.  SIP encoding

   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.


5.  SIP WebSocket Transport







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5.1.  General

   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.

      This simplifies parsing of SIP messages for both clients and
      servers.  There is no need to establish message boundaries using
      Content-Length headers between messages.  Other SIP transports,
      such as UDP and SCTP [RFC4168] also provide this benefit.

5.2.  Updates to RFC 3261

5.2.1.  Via Transport Parameter

   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).

   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]):

     transport  =/  "WS" / "WSS"

5.2.2.  SIP URI Transport Parameter

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

   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]):

     transport-param  =/  "transport=" "ws"





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5.2.3.  Via received parameter

   [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"
      field 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 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:

      When a SIP WebSocket Server receives a request it MAY decide not
      to add a "received" parameter to the top Via header.  Therefore
      SIP WebSocket Clients MUST accept responses without such a
      parameter in the top Via header regardless of whether the Via
      "sent-by" field contains a domain name.

5.2.4.  SIP transport implementation requirements

   [RFC3261] section 18 "Transport" states the following:

      All SIP elements MUST implement UDP and TCP.  SIP elements MAY
      implement other protocols.

   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



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   implement the WebSocket transport defined by this specification.

5.3.  Locating a SIP Server

   [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.

      At the time this document was written, DNS NAPTR/SRV queries could
      not be performed by commonly available WebSocket client stacks (in
      JavaScript engines and web browsers).

   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.


6.  Connection Keep-Alive

   _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 WebSocket API [WS-API] does not provide a mechanism for
      applications running in a web browser to control whether or not
      periodic WebSocket "Ping" frames are sent to the server.  The
      implementation of such a keep-alive feature is the decision of
      each web browser manufacturer and may also depend on the
      configuration of the web browser.

   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.


7.  Authentication

   _This section is non-normative._

   This section describes how authentication is achieved through the
   requirements in [RFC6455], [RFC6265] and [RFC3261].




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   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:

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

   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).


8.  Examples

8.1.  Registration


   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



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   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



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     ;+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

8.2.  INVITE dialog through a proxy































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   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



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   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



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   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



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   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


9.  Security Considerations

9.1.  Secure WebSocket Connection

   It is recommended that the SIP traffic transported over a WebSocket
   communication be protected by using a secure WebSocket connection



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   (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.

9.2.  Usage of SIPS Scheme

   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.


10.  IANA Considerations

   RFC Editor Note: Please set the RFC number assigned for this document
   in the sub-sections below and remove this note.

10.1.  Registration of the WebSocket SIP Sub-Protocol

   This specification requests IANA to register the WebSocket SIP sub-
   protocol under the "WebSocket Subprotocol Name" Registry with the
   following data:

   Subprotocol Identifier:  sip

   Subprotocol Common Name:  WebSocket Transport for SIP (Session
      Initiation Protocol)

   Subprotocol Definition:  TBD: this document

10.2.  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 Session Initiation Protocol (SIP) NAPTR Resource
   Record Services Field".  The resulting entries are as follows:

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






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10.3.  SIP/SIPS URI Parameters Sub-Registry

   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:

   Parameter Name   Predefined Values   Reference
   --------------   -----------------   ---------
   transport        Yes                 [RFC3261][TBD: this document]

10.4.  Header Fields Sub-Registry

   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:

   Header Name   compact   Reference
   -----------   -------   ---------
   Via           v         [RFC3261][TBD: this document]

10.5.  Header Field Parameters and Parameter Values Sub-Registry

   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:

                                 Predefined
   Header Field  Parameter Name  Values  Reference
   ------------  --------------  ------  ---------
   Via           received        No      [RFC3261][TBD: this document]

10.6.  SIP Transport Sub-Registry

   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:

   +------------+------------------------+
   | 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]   |
   +------------+------------------------+



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   The policy for registration of values in this registry is "Standards
   Action", as that term is defined by [RFC5226].


11.  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, 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.


12.  References

12.1.  Normative References

   [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.

   [RFC5226]  Narten, T. and H. Alvestrand, "Guidelines for Writing an
              IANA Considerations Section in RFCs", BCP 26, RFC 5226,
              May 2008.

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

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






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12.2.  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.

   [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.

   [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.

   [RFC5922]  Gurbani, V., Lawrence, S., and A. Jeffrey, "Domain
              Certificates in the Session Initiation Protocol (SIP)",
              RFC 5922, June 2010.

   [RFC6223]  Holmberg, C., "Indication of Support for Keep-Alive",
              RFC 6223, April 2011.

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

   [WS-API]   W3C and I. Hickson, Ed., "The WebSocket API",
              September 2012.






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Appendix A.  Implementation Guidelines

   _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.

      If a REFER request is sent to a third SIP user agent including the
      Contact URI of a SIP WebSocket Client as the target in its
      Refer-To header field, such a URI will be reachable by the third
      SIP UA only if it is a globally routable URI.  GRUU (Globally
      Routable User Agent URI) is a solution for those scenarios, and
      would cause the incoming request from the third SIP user agent to
      be sent to the SIP registrar, which would route the request to the
      SIP WebSocket Client via the Outbound Edge Proxy.



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A.1.  SIP WebSocket Client Considerations

   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.

      The Contact URI provided by SIP UAs requesting (and receiving)
      Outbound support is not used for routing requests to those UAs,
      thus it is safe to set a random domain in the Contact URI
      hostport.

   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.

      In web browsers it is difficult to generate or collect a suitable
      value to be used as a URN value from the browser itself.  This
      scenario 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 JavaScript SIP stack code, and then it is
      stored as a Cookie within the browser.

A.2.  SIP WebSocket Server Considerations

   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.


Authors' Addresses

   Inaki Baz Castillo
   Versatica
   Barakaldo, Basque Country
   Spain

   Email: ibc@aliax.net




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   Jose Luis Millan Villegas
   Versatica
   Bilbao, Basque Country
   Spain

   Email: jmillan@aliax.net


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

   Email: vpascual@acmepacket.com




































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