Network Working Group I. Baz Castillo
Internet-Draft J. Luis Millan
Intended status: Standards Track XtraTelecom S.A.
Expires: March 16, 2012 V. Pascual
Acme Packet
September 13, 2011
WebSocket Transport for Session Initiation Protocol (SIP)
draft-ibc-rtcweb-sip-websocket-00
Abstract
This document specifies a WebSocket subprotocol for a new transport
in SIP (Session Initiation Protocol). The WebSocket protocol enables
two-way realtime communication between clients (typically web-based
applications) and servers. The main goal of this specification is to
integrate the SIP protocol within web applications.
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 March 16, 2012.
Copyright Notice
Copyright (c) 2011 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
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Scope . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
4. SIP WebSocket Transport . . . . . . . . . . . . . . . . . . . 6
4.1. Via Transport Parameter . . . . . . . . . . . . . . . . . 6
4.2. SIP URI Transport Parameter . . . . . . . . . . . . . . . 6
4.3. Sending Responses . . . . . . . . . . . . . . . . . . . . 7
5. The WebSocket SIP Subprotocol . . . . . . . . . . . . . . . . 8
6. WebSocket Client Usage . . . . . . . . . . . . . . . . . . . . 9
6.1. WebSocket Disconnection . . . . . . . . . . . . . . . . . 10
7. WebSocket Server Usage . . . . . . . . . . . . . . . . . . . . 11
7.1. SIP Proxy Considerations . . . . . . . . . . . . . . . . . 11
8. WebSocket Connection Keep Alive . . . . . . . . . . . . . . . 12
9. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
9.1. Registration . . . . . . . . . . . . . . . . . . . . . . . 13
9.2. INVITE dialog through a proxy . . . . . . . . . . . . . . 15
9.3. INVITE dialog through two proxies . . . . . . . . . . . . 18
10. Security Considerations . . . . . . . . . . . . . . . . . . . 23
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 24
11.1. Registration of new Via transports . . . . . . . . . . . . 24
11.2. Registration of new SIP URI transport . . . . . . . . . . 24
11.3. Registration of the WebSocket SIP subprotocol . . . . . . 24
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 25
12.1. Normative References . . . . . . . . . . . . . . . . . . . 25
12.2. Informative References . . . . . . . . . . . . . . . . . . 25
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 27
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1. Introduction
Integrating the SIP protocol [RFC3261] within modern web-based
applications has been a hard task historically due to the
specification complexity and inherent limitations in web browsers and
HTTP protocol [RFC2616]. The arrival of WebSocket
[I-D.ietf-hybi-thewebsocketprotocol] and [RTC-Web] (Real Time
Collaboration on the World Wide Web) provides a two-way communication
technology for web-based applications along with multimedia
capabilities for audio and video sessions in web browsers, making
feasible the requeriments of the SIP protocol.
This specification defines a new WebSocket subprotocol for
transporting SIP messages between a WebSocket client and server, a
new transport for the SIP protocol and procedures for SIP proxies
when behaving as a bridge between WebSocket and other SIP transports.
No changes have been made to the SIP protocol [RFC3261].
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2. Conventions
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].
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3. Scope
The WebSocket protocol is mostly suitable for web-based applications
running in a web browser. Other applications running out of web
browsers do not have the constraints of web applications since
typically they can directly access to the transport layer.
In the same manner, the WebSocket protocol adds a network overhead
since it works as an intermediary layer between the transport and
application layers. There is no benefit on using SIP over WebSocket
transport between two SIP nodes when none of them runs within a web
browser. Even more, the WebSocket protocol is not symmetric since
just a WebSocket client can open a connection to a WebSocket server
(a WebSocket client does not listen for incoming connections).
Given these arguments, this specification is mostly focused on
integrating the SIP protocol within web-based applications or any
client application using the WebSocket protocol. Other aspects such
as DNS NAPTR/SRV resolution for SIP over WebSocket transport are not
covered by this specification since they are mainly useless given the
WebSocket protocol nature.
This document just covers SIP as a signalling protocol, leaving
multimedia capabilities integration for a separate document once
[RTC-Web] (Real Time Collaboration on the World Wide Web) becomes a
standard.
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4. SIP WebSocket Transport
WebSocket [I-D.ietf-hybi-thewebsocketprotocol] is a reliable protocol
and therefore the WebSocket subprotocol 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 a reliable transport as defined in
[RFC3261].
4.1. 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 session is established on top of TLS
[RFC5246] over 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]):
transport = "UDP" / "TCP" / "TLS" / "WS" / "WSS"
/ other-transport
The following are examples of Via header fields using "WS" and "WSS":
Via: SIP/2.0/WS 1.2.3.4:28456
Via: SIP/2.0/WSS [2001:0:63ba:74c:1806:7ea2:9aab:f892]:32802
4.2. 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.
Whether to select a plain or secure WebSocket connection depends on
the SIP URI schema ("sip" schema means plain WebSocket connection
while "sips" schema requires secure WebSocket connection).
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]):
transport-param = "transport="
( "udp" / "tcp" / "sctp" / "tls" / "ws"
/ other-transport )
The following are examples of SIP URI's containing a "ws" transport
parameter:
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sip:alice@1.2.3.4:28456;transport=ws
sips:bob@[2001:0:63ba:74c:1806:7ea2:9aab:f892]:32802;transport=ws
4.3. 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".
This is due 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.
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5. The WebSocket SIP Subprotocol
The term WebSocket subprotocol refers to the application-level
protocol layered over a WebSocket connection. This document
specifies the WebSocket SIP subprotocol for carrying SIP requests and
responses through a WebSocket connection.
WebSocket [I-D.ietf-hybi-thewebsocketprotocol] defines message units
as application data exchange for communication endpoints, becoming a
message boundary protocol. These messages can contain UTF-8 text or
binary data. The WebSocket SIP subprotocol specified in this
document mandates messages of type UTF-8 text.
The WebSocket client and WebSocket server send SIP messages to each
other. Each SIP message MUST be carried within a single WebSocket
message and MUST be a complete SIP message, so a Content-Length
header field 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 (typically using Content-Length headers) between
different messages. Same advantage is present in other message-
based SIP transports as UDP or SCTP [RFC4168].
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6. WebSocket Client Usage
As stated in [I-D.ietf-hybi-thewebsocketprotocol], a WebSocket URI
[RFC3986] is given to the WebSocket client (typically within a web-
based application) who resolves the URI destination and establishes a
WebSocket connection with the corresponding server (by performing the
handshake and negotiation procedures described in
[I-D.ietf-hybi-thewebsocketprotocol]).
The client application is supposed to be provided with SIP account
configuration values (as an AoR, outbound proxy and so on). Such
values are used by the client application when generating SIP
messages.
After establishing the WebSocket connection, the client SHOULD
discover the source IP and port from which the server has received
the TCP connection. Such IP and port are required for constructing
the client's SIP local URI (to be used in Contact header during SIP
registration and SIP dialogs).
The mechanism used by the client application in order to discover
its source IP and port is currently out of the scope of this
specification, although it might be defined in future revisions of
this document.
The client's SIP local URI MUST be constructed as follows:
o If the WebSocket connection is secure (given WebSocket URI has
"wss" schema) the URI MUST have "sips" schema, "sip" otherwise.
o The URI username is up to the application.
o The URI hostport is determined by the local IP and port previously
retrieved.
o A "transport" parameter with value "ws" MUST be added to the URI.
This SIP local URI MUST be used by the client as a registration
binding (Contact URI in a REGISTER) and as a local target for SIP
dialogs (Contact URI in a request or response) since this URI is the
only adddress in which the client can be contacted, and just through
the WebSocket server.
Any new request sent by the client MUST contain the discovered local
IP and port in the Via "sent-by" field. Via "sent-transport" field
MUST be set to "WSS" if the WebSocket connection is secure, to "WS"
otherwise.
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Due to the nature of the WebSocket protocol, the client sends all the
SIP requests to the WebSocket server it is connected to, so the
WebSocket server behaves as a de facto outbound SIP proxy.
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 specifying an expiration
interval of "0" for that contact address in a REGISTER request as
described in section 10.2.2 of [RFC3261].
6.1. 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 the SIP local URI
discovery process again followed by a new SIP registration.
The client MAY also remove the previous registration binding in the
registrar server, as such address is no longer reachable.
When the WebSocket server is also the SIP registrar server, it MAY
remove the SIP registration bindings associated to a WebSocket
connection after such connection has been closed. Such a decision
is out of the scope of this specification and depends on the SIP
network topology.
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7. WebSocket Server Usage
Here we assume that a SIP proxy or UAS (User Agent Server) is also
acting as a WebSocket server implementing the WebSocket subprotocol
described in this document. The server receives WebSocket connection
attempts from clients. How the server authorizes or not those
connections is out of the scope of this specification. Once the
WebSocket subprotocol defined in this document has been negotiated,
both client and server can send SIP messages to each other.
The server can only contact a SIP URI with the parameter
"transport=ws" in case the destination address belongs to an existing
WebSocket connection established from a WebSocket client. If not, a
local transport error MUST be generated (which involves a 500 or 503
SIP response code).
Such a case could happen when an existing SIP registration binding
points to an already closed WebSocket connection which was not
removed.
7.1. SIP Proxy Considerations
A SIP proxy implementing WebSocket transport can intercommunicate
clients using SIP over WebSocket with other SIP clients or nodes
using any other transport.
When the proxy bridges between WebSocket transport 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 nodes cannot
establish a connection to WebSocket clients. It is also recommended
that the proxy follows recommendations in [RFC5658] and uses double
Record-Route technique in these cases.
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 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.
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8. WebSocket Connection Keep Alive
It is recommended that the WebSocket client or server keeps the
WebSocket connection open by sending periodic Ping frames as
described in [I-D.ietf-hybi-thewebsocketprotocol] section 5.5.2. The
mechanisms of decision for a WebSocket endpoint to maintain, or not,
the connection over time is out of scope of this document.
In some cases due to transient network errors, the connection with
the WebSocket server could be lost without the WebSocket client being
aware of it. The WebSocket client would only realize of the network
failure when attempting to send new data over the WebSocket
connection.
The authors of this specification have requested the W3C (World
Wide Web Consortium) to include a mechanism in the WebSocket API
[WS-API] for instructing the WebSocket client to supervise the
connection by sending periodical Ping frames at the interval
requested by the API user.
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9. Examples
The flows depicted in this section describe the behavior of an
initial prototype which is currently under development.
9.1. Registration
Alice (SIP WS) WebSocket SIP Server
| |
|OPTIONS F1 |
|---------------------------->|
|200 OK F2 |
|<----------------------------|
| |
|REGISTER F3 |
|---------------------------->|
|200 OK F4 |
|<----------------------------|
| |
Alice is a WebSocket client running on a web browser. Alice
establishes a plain WebSocket connection with a WebSocket server
(also a SIP proxy/registrar) implementing the SIP subprotocol. Upon
connection, Alice sends a SIP OPTIONS request including an empty
"rport" parameter [RFC3581] in the Via header and obtains its source
IP and port from the Via "received" and "rport" parameters in the
response. Alice then forms its SIP local URI and constructs a
REGISTER request.
Message details (authentication and SDP bodies are omitted for
simplicity):
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F1 OPTIONS Alice -> WebSocket SIP Server
OPTIONS sip:ws-server.atlanta.com SIP/2.0
Via: SIP/2.0/WS 1.2.3.4;branch=z9hG4bKasudf;rport
From: sip:alice@atlanta.com;tag=ux8asodj
To: sip:ws-server.atlanta.com
Call-ID: 87djahs72kjsd
CSeq: 1 OPTIONS
Max-Forwards: 1
Accept: application/sdp
F2 200 OK WebSocket SIP Server -> Alice
SIP/2.0 200 OK
Via: SIP/2.0/WS 1.2.3.4;branch=z9hG4bKasudf;received=93.12.40.105;
rport=19465
From: sip:alice@atlanta.com;tag=ux8asodj
To: sip:ws-server.atlanta.com;tag=jcx67hjm
Call-ID: 87djahs72kjsd
CSeq: 1 OPTIONS
Content-Type: application/sdp
F3 REGISTER Alice -> WebSocket SIP Server
REGISTER sip:proxy.atlanta.com SIP/2.0
Via: SIP/2.0/WS 93.12.40.105:19465;branch=z9hG4bKasudf
From: sip:alice@atlanta.com;tag=65bnmj.34asd
To: sip:ws-server.atlanta.com
Call-ID: aiuy7k9njasd
CSeq: 1 REGISTER
Max-Forwards: 70
Contact: <sip:alice@93.12.40.105:19465;transport=ws>
F4 200 OK WebSocket SIP Server -> Alice
SIP/2.0 200 OK
Via: SIP/2.0/WS 93.12.40.105:19465;branch=z9hG4bKasudf
From: sip:alice@atlanta.com;tag=65bnmj.34asd
To: sip:ws-server.atlanta.com;tag=12isjljn8
Call-ID: aiuy7k9njasd
CSeq: 1 REGISTER
Contact: <sip:alice@93.12.40.105:19465;transport=ws>
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9.2. INVITE dialog through a proxy
Alice (SIP WSS) SIP Proxy (SIP UDP) Carol
| | |
|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 |
| |---------------------------->|
| | |
Here the WebSocket server is also a SIP proxy and registrar for the
domain atlanta.com. Alice, a WebSocket SIP client, calls Carol's AoR
through a secure WebSocket connection. The WebSocket SIP server acts
as a SIP proxy routing the INVITE to the UDP location of Carol. The
proxy does Loose-Routing. Carol answers the call and terminates it
later.
Message details (authentication and SDP bodies are omitted for
simplicity):
F1 INVITE Alice -> SIP Proxy (transport WSS)
INVITE sip:carol@atlanta.com SIP/2.0
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Via: SIP/2.0/WSS 93.12.40.105:20565;branch=z9hG4bK56sdasks
From: sip:alice@atlanta.com;tag=asdyka899
To: sip:carol@atlanta.com
Call-ID: asidkj3ss
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sips:alice@93.12.40.105:20565;transport=ws>
Content-Type: application/sdp
F2 100 Trying SIP Proxy -> Alice (transport WSS)
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 93.12.40.105:20565;branch=z9hG4bK56sdasks
From: sip:alice@atlanta.com;tag=asdyka899
To: sip:carol@atlanta.com
Call-ID: asidkj3ss
CSeq: 1 INVITE
F3 INVITE SIP Proxy -> Carol (transport UDP)
INVITE sip:carol@77.123.45.23:5060 SIP/2.0
Via: SIP/2.0/UDP 100.100.100.100;branch=z9hG4bKhjhjqw32c
Via: SIP/2.0/WSS 93.12.40.105:20565;branch=z9hG4bK56sdasks
Record-Route: <sip:100.100.100.100;transport=udp>,
<sips:100.100.100.100:9090;transport=ws>
From: sip:alice@atlanta.com;tag=asdyka899
To: sip:carol@atlanta.com
Call-ID: asidkj3ss
CSeq: 1 INVITE
Max-Forwards: 69
Contact: <sips:alice@93.12.40.105:20565;transport=ws>
Content-Type: application/sdp
F4 200 OK Carol -> SIP Proxy (transport UDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 100.100.100.100;branch=z9hG4bKhjhjqw32c
Via: SIP/2.0/WSS 93.12.40.105:20565;branch=z9hG4bK56sdasks
Record-Route: <sip:100.100.100.100;transport=udp>,
<sips:100.100.100.100:9090;transport=ws>
From: sip:alice@atlanta.com;tag=asdyka899
To: sip:carol@atlanta.com;tag=bmqkjhsd
Call-ID: asidkj3ss
CSeq: 1 INVITE
Max-Forwards: 69
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Contact: <sip:carol@77.123.45.23:5060;transport=udp>
Content-Type: application/sdp
F5 200 OK SIP Proxy -> Alice (transport WSS)
SIP/2.0 200 OK
Via: SIP/2.0/WSS 93.12.40.105:20565;branch=z9hG4bK56sdasks
Record-Route: <sip:100.100.100.100;transport=udp>,
<sips:100.100.100.100:9090;transport=ws>
From: sip:alice@atlanta.com;tag=asdyka899
To: sip:carol@atlanta.com;tag=bmqkjhsd
Call-ID: asidkj3ss
CSeq: 1 INVITE
Max-Forwards: 69
Contact: <sip:carol@77.123.45.23:5060;transport=udp>
Content-Type: application/sdp
F6 ACK Alice -> SIP Proxy (transport WSS)
ACK sip:carol@77.123.45.23:5060;transport=udp SIP/2.0
Via: SIP/2.0/WSS 93.12.40.105:20565;branch=z9hG4bKhgqqp090
Route: <sips:100.100.100.100:9090;transport=ws>,
<sip:100.100.100.100;transport=udp>
From: sip:alice@atlanta.com;tag=asdyka899
To: sip:carol@atlanta.com;tag=bmqkjhsd
Call-ID: asidkj3ss
CSeq: 1 ACK
Max-Forwards: 70
F7 ACK SIP Proxy -> Carol (transport UDP)
ACK sip:carol@77.123.45.23:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 100.100.100.100;branch=z9hG4bKhwpoc80zzx
Via: SIP/2.0/WSS 93.12.40.105:20565;branch=z9hG4bKhgqqp090
From: sip:alice@atlanta.com;tag=asdyka899
To: sip:carol@atlanta.com;tag=bmqkjhsd
Call-ID: asidkj3ss
CSeq: 1 ACK
Max-Forwards: 69
F8 BYE Carol -> SIP Proxy (transport UDP)
BYE sips:alice@93.12.40.105:20565;transport=ws SIP/2.0
Via: SIP/2.0/UDP 77.123.45.23;branch=z9hG4bKbiuiansd001
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Route: <sip:100.100.100.100;transport=udp>,
<sips:100.100.100.100:9090;transport=ws>
From: sip:carol@atlanta.com;tag=bmqkjhsd
To: sip:alice@atlanta.com;tag=asdyka899
Call-ID: asidkj3ss
CSeq: 1201 BYE
Max-Forwards: 70
F9 BYE SIP Proxy -> Alice (transport WSS)
BYE sips:alice@93.12.40.105:20565;transport=ws SIP/2.0
Via: SIP/2.0/WSS 100.100.100.100:9090;branch=z9hG4bKmma01m3r5
Via: SIP/2.0/UDP 77.123.45.23;branch=z9hG4bKbiuiansd001
From: sip:carol@atlanta.com;tag=bmqkjhsd
To: sip:alice@atlanta.com;tag=asdyka899
Call-ID: asidkj3ss
CSeq: 1201 BYE
Max-Forwards: 69
F10 200 OK Alice -> SIP Proxy (transport WSS)
SIP/2.0 200 OK
Via: SIP/2.0/WSS 100.100.100.100:9090;branch=z9hG4bKmma01m3r5
Via: SIP/2.0/UDP 77.123.45.23;branch=z9hG4bKbiuiansd001
From: sip:carol@atlanta.com;tag=bmqkjhsd
To: sip:alice@atlanta.com;tag=asdyka899
Call-ID: asidkj3ss
CSeq: 1201 BYE
F11 200 OK SIP Proxy -> Carol (transport UDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 77.123.45.23;branch=z9hG4bKbiuiansd001
From: sip:carol@atlanta.com;tag=bmqkjhsd
To: sip:alice@atlanta.com;tag=asdyka899
Call-ID: asidkj3ss
CSeq: 1201 BYE
9.3. INVITE dialog through two proxies
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Alice (SIP WS) Proxy 1 (SIP UDP) Proxy 2 (SIP WSS) Bob
| | | |
|INVITE F1 | | |
|-------------------->| | |
|100 Trying F2 | | |
|<--------------------| | |
| |INVITE F3 | |
| |-------------------->| |
| |100 Trying F4 | |
| |<--------------------| |
| | |INVITE F5 |
| | |-------------------->|
| | |486 Busy Here F6 |
| | |<--------------------|
| | |ACK F7 |
| | |-------------------->|
| |486 Busy Here F8 | |
| |<--------------------| |
| |ACK F9 | |
| |-------------------->| |
|486 Busy Here F10 | | |
|<--------------------| | |
|ACK F11 | | |
|-------------------->| | |
| | | |
Alice and Bob are WebSocket clients running on web browsers. Alice
belongs to atlanta.com SIP domain while Bob does to biloxi.com. Each
domain has its own SIP proxy. Both proxies are also WebSocket
servers. Alice calls Bob's AoR through a WebSocket connection. Bob
responds the INVITE with a 486 Busy Here. Communication through
proxies is made via UDP transport protocol.
Message details (authentication and SDP bodies are omitted for
simplicity):
F1 INVITE Alice -> Proxy 1 (transport WS)
INVITE sip:bob@biloxi.com SIP/2.0
Via: SIP/2.0/WS 93.12.40.105:21324;branch=z9hG4bKmmuuq
From: Alice <sip:alice@atlanta.com>;tag=lxtyr
To: Bob <sip:bob@biloxi.com>
Call-ID: aslke3dkj
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:alice@93.12.40.105:21324;transport=ws>
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Content-Type: application/sdp
F2 100 Trying Proxy 1 -> Alice (transport WS)
SIP/2.0 100 Trying
Via: SIP/2.0/WS 93.12.40.105:21324;branch=z9hG4bKmmuuq
From: Alice <sip:alice@atlanta.com>;tag=lxtyr
To: Bob <sip:bob@biloxi.com>
Call-ID: aslke3dkj
CSeq: 1 INVITE
F3 INVITE Proxy 1 -> Proxy 2 (transport UDP)
INVITE sip:bob@biloxi.com SIP/2.0
Via: SIP/2.0/UDP 101.101.101.101;branch=z9hG4bKdkej
Via: SIP/2.0/WS 93.12.40.105:21324;branch=z9hG4bKmmuuq
Record-Route: <sip:101.101.101.101;transport=udp>
Record-Route: <sip:101.101.101.101:80;transport=ws>
From: Alice <sip:alice@atlanta.com>;tag=lxtyr
To: Bob <sip:bob@biloxi.com>
Call-ID: aslke3dkj
CSeq: 1 INVITE
Max-Forwards: 69
Contact: <sip:alice@93.12.40.105:21324;transport=ws>
Content-Type: application/sdp
F4 100 Trying Proxy 2 -> Proxy 1 (transport UDP)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 101.101.101.101;branch=z9hG4bKdkej
Via: SIP/2.0/WS 93.12.40.105:21324;branch=z9hG4bKmmuuq
From: Alice <sip:alice@atlanta.com>;tag=lxtyr
To: Bob <sip:bob@biloxi.com>
Call-ID: aslke3dkj
CSeq: 1 INVITE
F5 INVITE Proxy 2 -> Bob (transport WSS)
INVITE sips:bob@85.84.123.222:30142;transport=ws SIP/2.0
Via: SIP/2.0/WSS 102.102.102.102:443;branch=z9hG4bKqowin
Via: SIP/2.0/UDP 101.101.101.101;branch=z9hG4bKdkej
Via: SIP/2.0/WS 93.12.40.105:21324;branch=z9hG4bKmmuuq
Record-Route: <sips:102.102.102.102:443;transport=ws>
Record-Route: <sip:102.102.102.102;transport=udp>
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Record-Route: <sip:101.101.101.101;transport=udp>
Record-Route: <sip:101.101.101.101:9090;transport=ws>
From: Alice <sip:alice@atlanta.com>;tag=lxtyr
To: Bob <sip:bob@biloxi.com>
Call-ID: aslke3dkj
CSeq: 1 INVITE
Max-Forwards: 68
Contact: <sip:alice@93.12.40.105:21324;transport=ws>
Content-Type: application/sdp
F6 486 Busy Here Bob -> Proxy 2 (transport WSS)
SIP/2.0 486 Busy Here
Via: SIP/2.0/WSS 102.102.102.102:443;branch=z9hG4bKqowin
Via: SIP/2.0/UDP 101.101.101.101;branch=z9hG4bKdkej
Via: SIP/2.0/WS 93.12.40.105:21324;branch=z9hG4bKmmuuq
From: Alice <sip:alice@atlanta.com>;tag=lxtyr
To: Bob <sip:bob@biloxi.com>;tag=dskfjd
Call-ID: aslke3dkj
CSeq: 1 INVITE
F7 ACK Proxy 2 -> Bob (transport WSS)
ACK sips:bob@85.84.123.222:30142;transport=ws SIP/2.0
Via: SIP/2.0/WSS 102.102.102.102:443;branch=z9hG4bKqowin
From: Alice <sip:alice@atlanta.com>;tag=lxtyr
To: Bob <sip:bob@biloxi.com>;tag=dskfjd
Call-ID: aslke3dkj
CSeq: 1 ACK
F8 486 Busy Here Proxy 2 -> Proxy 1 (transport UDP)
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 101.101.101.101;branch=z9hG4bKdkej
Via: SIP/2.0/WS 93.12.40.105:21324;branch=z9hG4bKmmuuq
From: Alice <sip:alice@atlanta.com>;tag=lxtyr
To: Bob <sip:bob@biloxi.com>;tag=dskfjd
Call-ID: aslke3dkj
CSeq: 1 INVITE
F9 ACK Proxy 1 -> Proxy 2 (transport UDP)
ACK sip:bob@biloxi.com SIP/2.0
Via: SIP/2.0/UDP 101.101.101.101;branch=z9hG4bKdkej
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From: Alice <sip:alice@atlanta.com>;tag=lxtyr
To: Bob <sip:bob@biloxi.com>;tag=dskfjd
Call-ID: aslke3dkj
CSeq: 1 ACK
F10 486 Busy Here Proxy 1 -> Alice (transport WS)
SIP/2.0 486 Busy Here
Via: SIP/2.0/WS 93.12.40.105:21324;branch=z9hG4bKmmuuq
From: Alice <sip:alice@atlanta.com>;tag=lxtyr
To: Bob <sip:bob@biloxi.com>;tag=dskfjd
Call-ID: aslke3dkj
CSeq: 1 INVITE
F11 ACK Alice -> Proxy 1 (transport WS)
ACK sip:bob@biloxi.com SIP/2.0
Via: SIP/2.0/WS 93.12.40.105:21324;branch=z9hG4bKmmuuq
From: Alice <sip:alice@atlanta.com>;tag=lxtyr
To: Bob <sip:bob@biloxi.com>;tag=dskfjd
Call-ID: aslke3dkj
CSeq: 1 ACK
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10. Security Considerations
If the client (typically a web-based application) needs to protect
the privacy of the SIP traffic through the WebSocket connection, it
is encouraged to use a secure WebSocket connection.
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11. IANA Considerations
11.1. 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]).
11.2. 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 (plain
or secure) connection.
11.3. Registration of the WebSocket SIP subprotocol
If a registry is created for WebSocket subprotocols, the SIP
subprotocol defined in this specification will be registered.
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12. References
12.1. Normative References
[I-D.ietf-hybi-thewebsocketprotocol]
Fette, I. and A. Melnikov, "The WebSocket protocol",
draft-ietf-hybi-thewebsocketprotocol-14 (work in
progress), September 2011.
[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.
[RFC5234] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", STD 68, RFC 5234, January 2008.
12.2. Informative References
[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.
[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
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(SIP)", RFC 5658, October 2009.
[RTC-Web] IETF and W3C, "Real Time Collaboration on the World Wide
Web", October 2010.
[WS-API] Hickson, I., "The Web Sockets API", September 2010.
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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
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