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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
Intended status: Standards Track                              Consultant
Expires: November 6, 2012                                     V. Pascual
                                                             Acme Packet
                                                             May 5, 2012


    The WebSocket Protocol as a Transport for the Session Initiation
                             Protocol (SIP)
                  draft-ietf-sipcore-sip-websocket-00

Abstract

   The WebSocket protocol enables two-way realtime communication between
   clients and servers.  This document specifies a new WebSocket sub-
   protocol as a reliable transport mechanism between SIP (Session
   Initiation Protocol) entities and enables usage of the SIP protocol
   in new scenarios.

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 November 6, 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



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   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 . . . . . . . . . . . . . . . . . . . .  3
   4.  The WebSocket SIP Sub-Protocol . . . . . . . . . . . . . . . .  4
     4.1.  Handshake  . . . . . . . . . . . . . . . . . . . . . . . .  4
     4.2.  SIP encoding . . . . . . . . . . . . . . . . . . . . . . .  5
   5.  SIP WebSocket Transport  . . . . . . . . . . . . . . . . . . .  5
     5.1.  General  . . . . . . . . . . . . . . . . . . . . . . . . .  5
     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.  Sending Responses  . . . . . . . . . . . . . . . . . .  6
     5.3.  Locating a SIP Server  . . . . . . . . . . . . . . . . . .  7
   6.  Connection Keep Alive  . . . . . . . . . . . . . . . . . . . .  7
   7.  Authentication . . . . . . . . . . . . . . . . . . . . . . . .  8
   8.  Examples . . . . . . . . . . . . . . . . . . . . . . . . . . .  8
     8.1.  Registration . . . . . . . . . . . . . . . . . . . . . . .  8
     8.2.  INVITE dialog through a proxy  . . . . . . . . . . . . . . 10
   9.  Security Considerations  . . . . . . . . . . . . . . . . . . . 14
     9.1.  Secure WebSocket Connection  . . . . . . . . . . . . . . . 14
     9.2.  Usage of SIPS Scheme . . . . . . . . . . . . . . . . . . . 15
   10. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 15
     10.1. Registration of the WebSocket SIP Sub-Protocol . . . . . . 15
     10.2. Registration of new Via transports . . . . . . . . . . . . 15
     10.3. Registration of new SIP URI transport  . . . . . . . . . . 15
     10.4. Registration of new NAPTR service field values . . . . . . 15
   11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 16
   12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 16
     12.1. Normative References . . . . . . . . . . . . . . . . . . . 16
     12.2. Informative References . . . . . . . . . . . . . . . . . . 17
   Appendix A.  Implementation Guidelines . . . . . . . . . . . . . . 18
     A.1.  SIP WebSocket Client Considerations  . . . . . . . . . . . 19
     A.2.  SIP WebSocket Server Considerations  . . . . . . . . . . . 19
   Appendix B.  HTTP Topology Hiding  . . . . . . . . . . . . . . . . 19
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 20








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1.  Introduction

   The WebSocket [RFC6455] protocol enables messages 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 run a WebSocket client
   stack.  The specification in this document enables usage of the SIP
   protocol in those new scenarios.

   This specification defines a new WebSocket sub-protocol (section 1.9
   in [RFC6455]) for transporting SIP messages between a WebSocket
   client and server, a new reliable and message boundary transport for
   the SIP protocol, new DNS NAPTR [RFC3403] service values and
   procedures for SIP entities implementing the WebSocket transport.
   Media transport is out of the scope of this document.


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 with WebSocket servers and speaking 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 speaking the WebSocket
         SIP Sub-Protocol as defined by this document.


3.  The WebSocket Protocol

   _This section is non-normative._




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   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]
   protocol by means of a specific HTTP GET method with Upgrade request
   sent by the client which is answered by the server (if the
   negotiation succeeded) with HTTP 101 status code.  Once the handshake
   is done 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 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.  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 underlying TCP connection for sending WebSocket
   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
   split into various WebSocket text/binary frames.

      However, the WebSocket API [WS-API] for web browsers just includes
      callbacks that are invoked upon receipt of an entire message,
      regardless of whether it was received in a single or multiple
      WebSocket frames.


4.  The WebSocket SIP Sub-Protocol

   The term WebSocket sub-protocol refers to the 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.

4.1.  Handshake

   The SIP WebSocket Client and SIP WebSocket Server need to agree on
   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



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   handshake request.  The 101 reply from the server MUST contain "sip"
   in its corresponding Sec-WebSocket-Protocol header.

   Below is an example of the 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 supporting 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 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
   application protocol.

4.2.  SIP encoding

   WebSocket messages are carried on top of WebSocket UTF-8 text frames
   or binary frames.  The SIP protocol [RFC3261] allows both text and
   binary bodies in SIP messages.  Therefore SIP WebSocket Clients and
   SIP WebSocket Servers MUST accept both WebSocket text and binary
   frames.


5.  SIP WebSocket Transport

5.1.  General

   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].




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   Each complete SIP message MUST be carried within a single WebSocket
   message, and a WebSocket message MUST NOT contain more than one SIP
   message.  Therefore the usage of the Content-Length header field is
   optional.

      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 such as UDP or SCTP [RFC4168].

5.2.  Updates to RFC 3261

5.2.1.  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 using TLS
   [RFC5246] with TCP transport).

   The updated RFC 3261 augmented BNF (Backus-Naur Form) [RFC5234] for
   this parameter reads as follows:

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

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 WebSocket protocol as
   transport.

   The updated RFC 3261 augmented BNF (Backus-Naur Form) for this
   parameter reads as follows:

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

5.2.3.  Sending Responses

   This specification updates the section 18.2.2 "Sending Responses" in
   [RFC3261] by adding the following:





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   o  If the Via "sent-protocol" is "WS" or "WSS" the response MUST be
      sent using the existing WebSocket connection to the source of the
      original request that created the transaction, if that connection
      is still open.  If that connection is no longer open, the server
      SHOULD NOT attempt to open a WebSocket connection for sending the
      response.

      This is due to the nature of the WebSocket protocol in which just
      the WebSocket client can establish a connection with the WebSocket
      server.  Typically a WebSocket client does not listen for inbound
      connections and WebSocket servers do not open outbound
      connections.

5.3.  Locating a SIP Server

   RFC 3263 [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 transport and "SIPS+D2W" for SIP WebSocket
   Servers that support secure WebSocket transport.

      Unfortunately neither JavaScript stacks nor WebSocket stacks
      running in current web browsers are capable of performing DNS
      NAPTR/SRV queries.

   In the absence of an explicit port and DNS SRV resource records, the
   default port for a SIP URI with "ws" transport parameter is 80 in
   case of SIP scheme and 443 in case of SIPS scheme.


6.  Connection Keep Alive

   _This section is non-normative._

   It is RECOMMENDED that the SIP WebSocket Client or Server keeps the
   WebSocket connection open by sending periodic WebSocket Ping frames
   as described in [RFC6455] section 5.5.2.

      Note however that The WebSocket API [WS-API] does not provide a
      mechanism for web applications running in a web browser to decide
      whether or not to send periodic WebSocket Ping frames to the
      server.  The usage of such a keep alive feature is a decision of
      each web browser vendor and may depend on the web browser
      configuration.

   Any future WebSocket protocol extension providing a keep alive
   mechanism could also be used.




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   The SIP stack in the SIP WebSocket Client MAY also use Network
   Address Translation (NAT) keep-alive mechanisms defined for SIP
   connection-oriented transports, such as the CRLF Keep-Alive Technique
   mechanism described in [RFC5626] section 3.5.1 or [RFC6223].

      Implementing these techniques would involve sending a WebSocket
      message to the SIP WebSocket Server whose content is a double
      CRLF, and expecting a WebSocket message from the server containing
      a single CRLF as response.


7.  Authentication

   _This section is non-normative._

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

   In order to authorize the WebSocket connection, the SIP WebSocket
   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 usually provided by the web server once the user has
   authenticated itself with the web server by following any of the
   multiple existing mechanisms.  As an alternative method, the SIP
   WebSocket Server could request HTTP authentication by replying 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 client
      handles the response per HTTP [RFC2616] procedures, in particular
      the client might perform authentication if it receives 401 status
      code.

   Regardless whether the SIP WebSocket Server requires authentication
   during the WebSocket handshake or not, authentication MAY be
   requested at SIP protocol level.  Therefore it is RECOMMENDED for a
   SIP WebSocket Client to implement HTTP Digest [RFC2617]
   authentication as stated in [RFC3261].


8.  Examples

8.1.  Registration






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   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 (a SIP WebSocket Client)
   establishes a secure WebSocket connection with a SIP proxy/registrar
   (a SIP WebSocket Server) 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 a random ".invalid"
   domain name for the Via sent-by and for the URI hostpart in the
   Contact header (see Appendix A.1).

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





























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   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
   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

8.2.  INVITE dialog through a proxy



















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   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.  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
   CSeq: 1 INVITE



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   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
   Max-Forwards: 69



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   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
   Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001



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   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


9.  Security Considerations

9.1.  Secure WebSocket Connection

   It is recommended to protect the privacy of the SIP traffic through
   the WebSocket communication by using a secure WebSocket connection



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   (tunneled over TLS [RFC5246]).

9.2.  Usage of SIPS Scheme

   SIPS scheme within a SIP request dictates that the entire request
   path to the target be secured.  If such a path includes a WebSocket
   node it MUST be a secure WebSocket connection.


10.  IANA Considerations

10.1.  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:  WebSocket Transport for SIP (Session
      Initiation Protocol)

   Subprotocol Definition:  TBD, it should point to this document

10.2.  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.

10.3.  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.

10.4.  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



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   resulting entries are as follows:

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


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, Adam Roach,
   Ranjit Avasarala, Xavier Marjou, Kevin P. Fleming, Nataraju A.

   Special thanks to Alan Johnston, Christer Holmberg and Salvatore
   Loreto for their full reviews, and also to Saul Ibarra Corretge for
   his contribution and suggestions.

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


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.

   [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",



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              RFC 6455, December 2011.

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.

   [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.

   [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.

   [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]   Hickson, I., "The Web Sockets API", May 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) to an intranet, perform web login by
   entering their user identifier and credentials, and retrieve a
   JavaScript code (along with the HTML code itself) 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 request sent by the web browser for
   the WebSocket handshake includes a Cookie [RFC6265] header with the
   value previously retrieved after the successful web login procedure.
   The Cookie value is then inspected by the WebSocket server for
   authorizing the connection.  Once the WebSocket connection is
   established, the SIP WebSocket Client performs a SIP registration and
   common SIP stuf begins.  The SIP registrar server is located behind
   the SIP outbound proxy.

   This scenario is quite similar to the one in which SIP UAs behind NAT
   connect to an outbound proxy and need to reuse the same TCP
   connection for incoming requests.  In both cases, the SIP clients are
   just reachable through the outbound proxy they are connected to.

   Outbound [RFC5626] seems an appropriate solution for this scenario.
   Therefore these SIP WebSocket Clients and the SIP registrar implement
   both Outbound and Path [RFC3327], and the SIP outbound proxy becomes
   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 thirdy SIP user agent indicating
      the Contact URI of a SIP WebSocket Client as the target in the
      Refer-To header field, such a URI will be reachable by the thirdy
      SIP UA just in the case it is a globally routable URI.  GRUU
      (Globally Routable User Agent URI) is a solution for those
      scenarios, and would enforce the incoming request from the thirdy
      SIP user agent to reach the SIP registrar which would route the
      request 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 which the WebSocket connection
   is originated from.  Therefore the SIP WebSocket Client creates a
   domain consisting of a random token followed by .invalid top domain
   name, as stated in [RFC2606], and uses it within the Via and Contact
   header.

      The Contact URI provided by the SIP clients requesting Outbound
      support is not later used for routing purposes, thus it is safe to
      set a random domain in the Contact URI hostpart.

   Both Outbound and GRUU specifications require the SIP 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 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 dialogs path as
   specified in [RFC3261].  Otherwise in-dialog requests would fail
   since SIP WebSocket Clients make use of their SIP WebSocket Server in
   order to send and receive SIP requests and responses.


Appendix B.  HTTP Topology Hiding

   _This section is non-normative._

   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



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      "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 SIP
   WebSocket Client and 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 could reveal sensitive information
   about the internal topology of the provider network to the 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 SIP WebSocket Server is of little use.
   Therefore, in order to allow hiding possible sensitive information
   about the provider infrastructure, the implementer could decide not
   to satisfy the requirement in RFC 3261 [RFC3261] section 18.2.1
   "Receiving Requests" and not add the "received" parameter to the Via
   header.

      However, keep in mind that this would involve a violation of the
      RFC 3261.


Authors' Addresses

   Inaki Baz Castillo
   Consultant
   Barakaldo, Basque Country
   Spain

   Email: ibc@aliax.net


   Jose Luis Millan Villegas
   Consultant
   Bilbao, Basque Country
   Spain

   Email: jmillan@aliax.net









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

   Email: vpascual@acmepacket.com












































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