SIPCORE Working Group I. Baz Castillo
Internet-Draft J. Luis Millan
Intended status: Standards Track XtraTelecom S.A.
Expires: July 18, 2012 V. Pascual
Acme Packet
January 15, 2012
The WebSocket Protocol as a Transport for the Session Initiation
Protocol (SIP)
draft-ibc-sipcore-sip-websocket-01
Abstract
This document specifies a WebSocket Sub-Protocol for a new transport
in SIP (Session Initiation Protocol). The WebSocket protocol enables
two-way realtime communication between clients and servers.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on July 18, 2012.
Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
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described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. The WebSocket Protocol . . . . . . . . . . . . . . . . . . . . 5
4. The WebSocket SIP Sub-Protocol . . . . . . . . . . . . . . . . 6
5. SIP WebSocket Transport . . . . . . . . . . . . . . . . . . . 8
5.1. Via Transport Parameter . . . . . . . . . . . . . . . . . 8
5.2. SIP URI Transport Parameter . . . . . . . . . . . . . . . 8
5.3. Sending Responses . . . . . . . . . . . . . . . . . . . . 8
6. Outbound and GRUU Usage . . . . . . . . . . . . . . . . . . . 10
7. Locating a SIP Server . . . . . . . . . . . . . . . . . . . . 11
8. WebSocket Client Usage . . . . . . . . . . . . . . . . . . . . 12
8.1. WebSocket Disconnection . . . . . . . . . . . . . . . . . 12
9. WebSocket Server Usage . . . . . . . . . . . . . . . . . . . . 13
9.1. SIP Proxy Considerations . . . . . . . . . . . . . . . . . 13
10. Connection Keep Alive . . . . . . . . . . . . . . . . . . . . 14
11. Authentication . . . . . . . . . . . . . . . . . . . . . . . . 15
12. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
12.1. Registration . . . . . . . . . . . . . . . . . . . . . . . 16
12.2. INVITE dialog through a proxy . . . . . . . . . . . . . . 17
13. Security Considerations . . . . . . . . . . . . . . . . . . . 22
13.1. Secure WebSocket Connection . . . . . . . . . . . . . . . 22
13.2. Usage of SIPS Schema . . . . . . . . . . . . . . . . . . . 22
13.3. WebSocket Topology Hiding . . . . . . . . . . . . . . . . 22
14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 24
14.1. Registration of the WebSocket SIP Sub-Protocol . . . . . . 24
14.2. Registration of new Via transports . . . . . . . . . . . . 24
14.3. Registration of new SIP URI transport . . . . . . . . . . 24
14.4. Registration of new NAPTR service field values . . . . . . 24
15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 25
16. References . . . . . . . . . . . . . . . . . . . . . . . . . . 26
16.1. Normative References . . . . . . . . . . . . . . . . . . . 26
16.2. Informative References . . . . . . . . . . . . . . . . . . 26
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 28
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1. Introduction
This specification defines a new WebSocket Sub-Protocol for
transporting SIP messages between a WebSocket client and server, a
new transport for the SIP protocol and procedures for SIP servers
when bridging WebSocket and other SIP transports.
This specification is focused on integrating the SIP protocol within
client applications running a WebSocket stack. Other aspects such as
the usage of WebSocket as a transport between SIP servers are not
fully covered by this specification.
This is because WebSocket client agents are expected to be mostly
implemented in client applications running in personal computers
and devices as smartphones, being applications that typically are
not able to manage TCP or UDP connections directly. Therefore
using WebSocket as a SIP transport between two proxies or servers
is of little use given the fact that those servers can typically
access to the UDP/TCP layer rather than having to use an extra
layer on top of it.
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2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
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3. The WebSocket Protocol
WebSocket protocol [RFC6455] is a transport layer on top of TCP in
which both client and server exchange message units in both
directions. The protocol defines a connection handshake, WebSocket
Sub-Protocol and extensions negotiation, a frame format for sending
application and control data, a masking mechanism, and status codes
for indicating disconnection causes.
The WebSocket connection handshake is based on HTTP [RFC2616]
protocol by means of a specific HTTP GET request sent by the client,
typically a web browser, which is answered by the server (if the
negotiation succeeded) with HTTP 101 status code. This handshake
procedure is designed to reuse the existing HTTP infrastructure.
During the connection handshake, client and server agree in the
application protocol to use on top of the WebSocket transport. Such
application protocol (also known as the "WebSocket Sub-Protocol")
defines the format and semantics of the messages exchanged between
both endpoints. The WebSocket Sub-Protocol to be used is up to the
application developer. It may be a custom protocol or a standarized
one (as the WebSocket SIP Sub-Protocol proposed in this document).
Once the HTTP 101 response is processed both client and server reuse
the existing TCP connection for sending application messages and
control frames to each other in a persistent way.
WebSocket defines message units as application data exchange for
communication endpoints, becoming a message boundary transport layer.
These messages can contain UTF-8 text or binary data, and can be
splitted into various WebSocket text/binary frames. However, the
WebSocket API [WS-API] for web browsers just includes JavaScript
callbacks that are invoked upon receipt of an entire message,
regardless it has been received in a single or multiple WebSocket
frames.
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4. The WebSocket SIP Sub-Protocol
The term WebSocket Sub-Protocol refers to the application-level
protocol layered over a WebSocket connection. This document
specifies the WebSocket SIP Sub-Protocol for carrying SIP requests
and responses through a WebSocket connection.
The WebSocket client and server need to agree on this protocol during
the WebSocket handshake procedure as defined in section 1.3 of
[RFC6455]. The client MUST include the value "sip" in the Sec-
WebSocket-Protocol header in its handshake request. The 101 reply
from the WebSocket server MUST contain "sip" in its own Sec-
WebSocket-Protocol header.
Below is an example of the WebSocket handshake in which the client
requests SIP Sub-Protocol support from the server:
GET / HTTP/1.1
Host: sip-ws.example.com
Upgrade: websocket
Connection: Upgrade
Sec-WebSocket-Key: dGhlIHNhbXBsZSBub25jZQ==
Origin: http://www.example.com
Sec-WebSocket-Protocol: sip
Sec-WebSocket-Version: 13
The handshake response from the server supporting the WebSocket SIP
Sub-Protocol would look like:
HTTP/1.1 101 Switching Protocols
Upgrade: websocket
Connection: Upgrade
Sec-WebSocket-Accept: s3pPLMBiTxaQ9kYGzzhZRbK+xOo=
Sec-WebSocket-Protocol: sip
Once the negotiation is done, the WebSocket connection is established
with SIP as the WebSocket Sub-Protocol. The WebSocket messages to be
transmitted over this connection MUST conform to the established
signaling protocol.
WebSocket messages are carried on top of WebSocket UTF-8 text frames
or binary frames. SIP protocol [RFC3261] allows both text and binary
bodies in SIP messages. Therefore a client and server implementing
the WebSocket SIP Sub-Protocol MUST accept both WebSocket text and
binary frames.
Each SIP message MUST be carried within a single WebSocket message
and MUST be a complete SIP message, so a Content-Length header field
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is not mandatory. Sending more than one SIP message within a single
WebSocket message is not allowed, neither sending an incomplete SIP
message.
This makes parsing of SIP messages easier on client side
(typically web-based applications with an strict and simple API
for receiving WebSocket messages). There is no need to establish
boundaries (using Content-Length headers) between different
messages. Same advantage is present in other message-based SIP
transports as UDP or SCTP [RFC4168].
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5. SIP WebSocket Transport
WebSocket [RFC6455] is a reliable protocol and therefore the
WebSocket sub-protocol for a SIP transport defined by this document
is also a reliable transport. Thus, client and server transactions
using WebSocket transport MUST follow the procedures and timer values
for reliable transports as defined in [RFC3261].
5.1. Via Transport Parameter
Via header fields carry the transport protocol identifier. This
document defines the value "WS" to be used for requests over plain
WebSocket protocol and "WSS" for requests over secure WebSocket
protocol (in which the WebSocket connection is established on top of
TLS [RFC5246] over TCP transport).
The updated augmented BNF (Backus-Naur Form) [RFC5234] for this
parameter is the following:
transport = "UDP" / "TCP" / "TLS" / "SCTP" / "WS" / "WSS"
/ other-transport
5.2. SIP URI Transport Parameter
This document defines the value "ws" as the transport parameter value
for a SIP URI [RFC3986] to be contacted using WebSocket protocol as
transport.
The updated augmented BNF (Backus-Naur Form) [RFC5234] for this
parameter is the following:
transport-param = "transport="
( "udp" / "tcp" / "sctp" / "tls" / "sctp"
/ "ws"
/ other-transport )
5.3. Sending Responses
The SIP server transport uses the value of the top Via header field
in order to determine where to send a response. If the "sent-
protocol" is "WS" or "WSS" the response MUST be sent using the
existing WebSocket connection to the source of the original request,
if that connection is still open. This requires the server transport
to maintain an association between server transactions and transport
connections. If that connection is no longer open, the server MUST
NOT attempt to open a WebSocket connection to the Via "sent-
by"/"received"/"rport". In that case the SIP server transport SHOULD
inform the transport user of a failure in sending.
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This is due to the nature of the WebSocket protocol in which just
the WebSocket client can establish a connection with the WebSocket
server. A WebSocket client does not listen for incoming
connections.
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6. Outbound and GRUU Usage
WebSocket requires the client to open a TCP connection with the
server and perform the WebSocket handshake. A WebSocket client does
not listen for incoming connections so it can only receive SIP
requests from the WebSocket server it is connected to. WebSocket
clients may use either public or private addressing but it is
expected that many of them will run the latter. Unfortunately, some
implementations may not have the ability to discover the local
transport address which the WebSocket connection is originated from
(e.g. a JavaScript stack within a web browser). Those
implementations are encouraged to create a domain consisting of a
random token followed by .invalid top domain name, as stated in
[RFC2606], and use it within the Via and Contact header.
Therefore clients and servers implementing SIP over the WebSocket
transport MUST implement the Outbound mechanism [RFC5626], being this
the most suitable solution for SIP clients behind Network Address
Translation (NAT) using reliable transports for contacting SIP
servers.
A client implementing SIP over the WebSocket transport SHOULD also
implement GRUU [RFC5627]. The registrar responsible for the
registration of SIP clients using the WebSocket transport SHOULD
implement GRUU as well.
If a REFER request is sent to a SIP User Agent indicating the
Contact URI of a WebSocket client as the target in the Refer-To
header field, such a URI will be reachable by the SIP UA just in
the case it is a globally routable URI obtained from a SIP
registrar implementing GRUU.
Both Outbound and GRUU require the client to indicate a Uniform
Resource Name (URN) in the "+sip.instance" parameter of the Contact
header during the registration. The client device is responsible for
getting such a constant and unique value.
In the case of web browsers it is hard to get a URN value from the
browser itself. This specification suggests that value is
generated according to [RFC5626] section 4.1 by the web
application running in the browser the first time it loads the web
page, and then it is stored as a Cookie [RFC6265] within the
browser data and loaded every time the same web page is visited.
The application developer could choose any other mechanism which
accomplishes the requirements of a URN.
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7. Locating a SIP Server
SIP entities follow normal SIP procedures in [RFC3263] to discover a
SIP server. This specification defines the NAPTR service value "SIP+
D2W" for servers that support plain WebSocket transport and "SIPS+
D2W" for servers that support secure WebSocket transport.
A SIP entity using the WebSocket transport SHOULD perform procedures
in [RFC3263] for the given WebSocket URI it will connect to. If the
WebSocket URI has "wss" schema the SIP entity MUST only consider
"SIPS+D2W" resource records. If the WebSocket URI does not contain a
domain in the host part or does include a port, the SIP entity MUST
follow procedures in [RFC6455] section 3 instead.
Unfortunately the JavaScript stack running in web browsers cannot
perform DNS NAPTR/SRV queries, neither the WebSocket stack running
in web browsers can do it. Thus, a WebSocket URI given within a
web application needs to have a numeric network address or a
hostname with associated DNS A/AAAA resource record(s) in its host
part.
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8. WebSocket Client Usage
The WebSocket connection MUST be established in order to allow the
client application to send and receive SIP requests.
Based on local policy, this might occur once the JavaScript SIP
application has been downloaded from the web server, or when the
SIP user using the web browser application registers itself to a
SIP registrar (assuming that SIP requests cannot be sent or
received before then).
In case the client application decides to close the WebSocket
connection (for example when performing "logout" in a web
application) it is recommended to remove the existing SIP
registration binding (if present) by means of a REGISTER with
expiration value of 0 and the associated "+sip.instance" Contact
header parameter as per [RFC5626].
8.1. WebSocket Disconnection
In some circumstances the WebSocket connection could be terminated by
the WebSocket server (for example when the server is restarted). If
the client application wants to become reachable again it SHOULD
reconnect to the WebSocket server and perform a new SIP registration
with same "+sip.instance" and "reg-id" Contact header parameters (as
stated in [RFC5626]).
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9. WebSocket Server Usage
How a SIP server authorizes WebSocket connection attemps from clients
is out of the scope of this specification. However some
informational guidelines are provided in Section 11. Once the
WebSocket SIP Sub-Protocol is agreed, both client and server can send
SIP messages to each other.
9.1. SIP Proxy Considerations
When a SIP proxy bridges WebSocket and any other SIP transport
(including WebSocket transport) it MUST perform Loose Routing as
specified in [RFC3261]. Otherwise in-dialog requests would fail
since WebSocket clients cannot contact destinations other than their
WebSocket server, and non-WebSocket SIP entities cannot establish a
connection to WebSocket clients. It is also recommended that SIP
proxy implementations use double Record-Route techniques (as
specified in [RFC5658]).
In the same way, if the SIP proxy implementing the WebSocket server
behaves as an outbound proxy for REGISTER requests, it MUST add a
Path header field as described in [RFC3327]. Otherwise the WebSocket
client would never receive incoming requests from the SIP registrar
server after the lookup procedures in the SIP location service.
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10. Connection Keep Alive
It is recommended that the WebSocket client or server keeps the
WebSocket connection open by sending periodic WebSocket Ping frames
as described in [RFC6455] section 5.5.2. The decision for a
WebSocket endpoint to maintain, or not, the connection over time is
out of scope of this document.
The client application MAY also use Network Address Translation (NAT)
keep-alive mechanisms defined for the SIP protocol, such as the CRLF
Keep-Alive Technique mechanism described in [RFC5626] section 3.5.1.
Therefore, a SIP server implementing the WebSocket transport MUST
support the CRLF Keep-Alive Technique.
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11. Authentication
Prior to sending SIP requests, the WebSocket client implementing the
SIP protocol connects to the WebSocket server and performs the
connection handshake. As described in Section 3 the handshake
procedure involves an HTTP GET request replied with HTTP 101 status
code by the server.
In order to authorize the WebSocket connection the server MAY inspect
the Cookie [RFC6265] header in the HTTP GET request (if present). In
case of web applications the value of such a Cookie is typically
provided by the web server once the user has authenticated itself
against the web application by following any of the multiple existing
mechanisms. As an alternative method, the WebSocket server could
request Digest [RFC2617] authentication by replying a HTTP 401 status
code. The WebSocket protocol [RFC6455] covers this usage in section
4.1:
If the status code received from the server is not 101, the client
handles the response per HTTP [RFC2616] procedures, in particular
the client might perform authentication if it receives 401 status
code.
Regardless the WebSocket server requires authentication during the
WebSocket handshake or not, authentication MAY be requested at SIP
protocol level. Therefore a SIP client using the WebSocket transport
MUST implement Digest [RFC2617] authentication as stated in
[RFC3261].
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12. Examples
12.1. Registration
Alice (SIP WSS) proxy.atlanta.com
| |
|REGISTER F1 |
|---------------------------->|
|200 OK F2 |
|<----------------------------|
| |
Alice loads a web page using her web browser and retrieves a
JavaScript code implementing the WebSocket SIP Sub-Protocol defined
in this document. The JavaScript code obtained from the web server
establishes a secure WebSocket connection with a SIP proxy/registrar
at proxy.atlanta.com. Upon WebSocket connection, Alice constructs
and sends a SIP REGISTER by requesting Outbound and GRUU support.
Since the JavaScript stack in a browser has no way to determine the
local address from which the WebSocket connection is made, this
implementation uses df7jal23ls0d.invalid for the Via sent-by and for
the URI hostpart in the Contact header.
Message details (authentication and SDP bodies are omitted for
simplicity):
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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
Route: <sip:proxy.atlanta.com:443;transport=ws;lr>
Contact: <sip:alice@df7jal23ls0d.invalid;transport=ws>
;reg-id=1
;+sip.instance="<urn:uuid:f81-7dec-14a06cf1>"
F2 200 OK proxy.atlanta.com -> Alice (transport WSS)
SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKasudf
From: sip:alice@atlanta.com;tag=65bnmj.34asd
To: sip:alice@atlanta.com;tag=12isjljn8
Call-ID: aiuy7k9njasd
CSeq: 1 REGISTER
Supported: outbound, gruu
Contact: <sip:alice@df7jal23ls0d.invalid;transport=ws>
;reg-id=1
;+sip.instance="<urn:uuid:f81-7dec-14a06cf1>"
;pub-gruu="sip:alice@atlanta.com;gr=urn:uuid:f81-7dec-14a06cf1"
;temp-gruu="sip:87ash54=3dd.98a@atlanta.com;gr"
;expires=3600
12.2. INVITE dialog through a proxy
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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 by using the
public GRUU retrieved from the registrar as Contact URI of the
INVITE. The WebSocket SIP server at proxy.atlanta.com acts as a SIP
proxy routing the INVITE to the UDP location of Bob, who answers the
call and terminates it later.
Message details (authentication and SDP bodies are omitted for
simplicity):
F1 INVITE Alice -> proxy.atlanta.com (transport WSS)
INVITE sip:bob@atlanta.com SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
From: sip:alice@atlanta.com;tag=asdyka899
To: sip:bob@atlanta.com
Call-ID: asidkj3ss
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CSeq: 1 INVITE
Max-Forwards: 70
Supported: path, outbound, gruu
Route: <sip:proxy.atlanta.com:443;transport=ws;lr>
Contact: <sip:alice@atlanta.com
;gr=urn:uuid:f81-7dec-14a06cf1;ob>"
Content-Type: application/sdp
F2 100 Trying proxy.atlanta.com -> Alice (transport WSS)
SIP/2.0 100 Trying
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
From: sip:alice@atlanta.com;tag=asdyka899
To: sip:bob@atlanta.com
Call-ID: asidkj3ss
CSeq: 1 INVITE
F3 INVITE proxy.atlanta.com -> Bob (transport UDP)
INVITE sip:bob@203.0.113.22:5060 SIP/2.0
Via: SIP/2.0/UDP proxy.atlanta.com;branch=z9hG4bKhjhjqw32c
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
Record-Route: <sip:proxy.atlanta.com;transport=udp;lr>,
<sip:h7kjh12s@proxy.atlanta.com:443;transport=ws;lr>
From: sip:alice@atlanta.com;tag=asdyka899
To: sip:bob@atlanta.com
Call-ID: asidkj3ss
CSeq: 1 INVITE
Max-Forwards: 69
Supported: path, outbound, gruu
Contact: <sip:alice@atlanta.com
;gr=urn:uuid:f81-7dec-14a06cf1;ob>"
Content-Type: application/sdp
F4 200 OK Bob -> proxy.atlanta.com (transport UDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP proxy.atlanta.com;branch=z9hG4bKhjhjqw32c
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
Record-Route: <sip:proxy.atlanta.com;transport=udp;lr>,
<sip:h7kjh12s@proxy.atlanta.com:443;transport=ws;lr>
From: sip:alice@atlanta.com;tag=asdyka899
To: sip:bob@atlanta.com;tag=bmqkjhsd
Call-ID: asidkj3ss
CSeq: 1 INVITE
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Max-Forwards: 69
Contact: <sip:bob@203.0.113.22:5060;transport=udp>
Content-Type: application/sdp
F5 200 OK proxy.atlanta.com -> Alice (transport WSS)
SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
Record-Route: <sip:proxy.atlanta.com;transport=udp;lr>,
<sip:h7kjh12s@proxy.atlanta.com:443;transport=ws;lr>
From: sip:alice@atlanta.com;tag=asdyka899
To: sip:bob@atlanta.com;tag=bmqkjhsd
Call-ID: asidkj3ss
CSeq: 1 INVITE
Max-Forwards: 69
Contact: <sip:bob@203.0.113.22:5060;transport=udp>
Content-Type: application/sdp
F6 ACK Alice -> proxy.atlanta.com (transport WSS)
ACK sip:bob@203.0.113.22:5060;transport=udp SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKhgqqp090
Route: <sip:h7kjh12s@proxy.atlanta.com:443;transport=ws;lr>,
<sip:proxy.atlanta.com;transport=udp;lr>,
From: sip:alice@atlanta.com;tag=asdyka899
To: sip:bob@atlanta.com;tag=bmqkjhsd
Call-ID: asidkj3ss
CSeq: 1 ACK
Max-Forwards: 70
F7 ACK proxy.atlanta.com -> Bob (transport UDP)
ACK sip:bob@203.0.113.22:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP proxy.atlanta.com;branch=z9hG4bKhwpoc80zzx
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKhgqqp090
From: sip:alice@atlanta.com;tag=asdyka899
To: sip:bob@atlanta.com;tag=bmqkjhsd
Call-ID: asidkj3ss
CSeq: 1 ACK
Max-Forwards: 69
F8 BYE Bob -> proxy.atlanta.com (transport UDP)
BYE sip:alice@atlanta.com;gr=urn:uuid:f81-7dec-14a06cf1;ob SIP/2.0
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Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001
Route: <sip:proxy.atlanta.com;transport=udp;lr>,
<sip:h7kjh12s@proxy.atlanta.com:443;transport=ws;lr>
From: sip:bob@atlanta.com;tag=bmqkjhsd
To: sip:alice@atlanta.com;tag=asdyka899
Call-ID: asidkj3ss
CSeq: 1201 BYE
Max-Forwards: 70
F9 BYE proxy.atlanta.com -> Alice (transport WSS)
BYE sip:alice@atlanta.com;gr=urn:uuid:f81-7dec-14a06cf1;ob SIP/2.0
Via: SIP/2.0/WSS proxy.atlanta.com:443;branch=z9hG4bKmma01m3r5
Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001
From: sip:bob@atlanta.com;tag=bmqkjhsd
To: sip:alice@atlanta.com;tag=asdyka899
Call-ID: asidkj3ss
CSeq: 1201 BYE
Max-Forwards: 69
F10 200 OK Alice -> proxy.atlanta.com (transport WSS)
SIP/2.0 200 OK
Via: SIP/2.0/WSS proxy.atlanta.com:443;branch=z9hG4bKmma01m3r5
Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001
From: sip:bob@atlanta.com;tag=bmqkjhsd
To: sip:alice@atlanta.com;tag=asdyka899
Call-ID: asidkj3ss
CSeq: 1201 BYE
F11 200 OK proxy.atlanta.com -> Bob (transport UDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001
From: sip:bob@atlanta.com;tag=bmqkjhsd
To: sip:alice@atlanta.com;tag=asdyka899
Call-ID: asidkj3ss
CSeq: 1201 BYE
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13. Security Considerations
13.1. Secure WebSocket Connection
It is recommended to protect the privacy of the SIP traffic through
the WebSocket communication by using a secure WebSocket connection
(tunneled over TLS [RFC5246]). For this, the client application MUST
be provided with a secure "wss" WebSocket URI.
13.2. Usage of SIPS Schema
SIPS schema within a SIP request dictates that the entire request
path to the target be secured. If such a path includes a WebSocket
connection it MUST be a secure WebSocket connection (tunneled over
TLS [RFC5246]) opened with a "wss" WebSocket URI.
13.3. WebSocket Topology Hiding
RFC 3261 [RFC3261] section 18.2.1 "Receiving Requests" states the
following:
When the server transport receives a request over any transport,
it MUST examine the value of the "sent-by" parameter in the top
Via header field value. If the host portion of the "sent-by"
parameter contains a domain name, or if it contains an IP address
that differs from the packet source address, the server MUST add a
"received" parameter to that Via header field value. This
parameter MUST contain the source address from which the packet
was received.
The requirement of adding the "received" parameter does not fit well
into WebSocket protocol nature. The WebSocket handshake connection
reuses existing HTTP infrastructure in which there could be certain
number of HTTP proxies and/or TCP load balancers between the client
and the WebSocket server, so the source IP the server would write
into the Via "received" parameter would be the IP of the HTTP/TCP
intermediary in front of it. This would reveal sensitive information
about the internal topology of the provider network to the WebSocket
client.
Thus, given the fact that SIP responses can only be sent over the
existing WebSocket connection, the meaning of the Via "received"
parameter added by the server is of little use. Therefore, in order
to allow hiding possible sensitive information about the provider
infrastructure, this specification relaxes the requirement in RFC
3261 [RFC3261] section 18.2.1 "Receiving Requests" by stating that a
WebSocket server receiving a SIP request from a WebSocket client MAY
choose not to add the Via "received" parameter nor honor the Via
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"rport" [RFC3581] parameter. A SIP client implementing the WebSocket
transport MUST be ready to receive SIP responses in which the topmost
Via header field does not contain the "received" and "rport"
parameters.
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14. IANA Considerations
14.1. Registration of the WebSocket SIP Sub-Protocol
This specification requests IANA to create the WebSocket SIP Sub-
Protocol in the registry of WebSocket sub-protocols with the
following data:
Subprotocol Identifier: sip
Subprotocol Common Name: SIP over WebSocket
Subprotocol Definition: TBD, it should point to this document
14.2. Registration of new Via transports
This specification registers two new transport identifiers for Via
headers:
WS: MUST be used when constructing a SIP request to be sent over a
plain WebSocket connection.
WSS: MUST be used when constructing a SIP request to be sent over a
secure WebSocket connection (tunneled over TLS [RFC5246]).
14.3. Registration of new SIP URI transport
This specification registers a new value for the "transport"
parameter in a SIP URI:
ws: Identifies a SIP URI to be contacted using a WebSocket
connection.
14.4. Registration of new NAPTR service field values
This document defines two new NAPTR service field values (SIP+D2W and
SIPS+D2W) and requests IANA to register these values under the
"Registry for the SIP SRV Resource Record Services Field". The
resulting entries are as follows:
Services Field Protocol Reference
-------------------- -------- ---------
SIP+D2W WS TBD: this document
SIPS+D2W WSS TBD: this document
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15. Acknowledgements
Special thanks to the following people who participated in
discussions on the SIPCORE and RTCWEB WG mailing lists and
contributed ideas and/or provided detailed reviews (the list is
likely to be incomplete): Hadriel Kaplan, Paul Kyzivat, Ranjit
Avasarala.
Special thanks to Saul Ibarra Corretge for his detailed review and
provided suggestions.
Special thanks also to Aranzazu Ruiz for her valuable collaboration
in the whole document.
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16. References
16.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2617] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
Leach, P., Luotonen, A., and L. Stewart, "HTTP
Authentication: Basic and Digest Access Authentication",
RFC 2617, June 1999.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3263] Rosenberg, J. and H. Schulzrinne, "Session Initiation
Protocol (SIP): Locating SIP Servers", RFC 3263,
June 2002.
[RFC5234] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", STD 68, RFC 5234, January 2008.
[RFC5626] Jennings, C., Mahy, R., and F. Audet, "Managing Client-
Initiated Connections in the Session Initiation Protocol
(SIP)", RFC 5626, October 2009.
[RFC5627] Rosenberg, J., "Obtaining and Using Globally Routable User
Agent URIs (GRUUs) in the Session Initiation Protocol
(SIP)", RFC 5627, October 2009.
[RFC6455] Fette, I. and A. Melnikov, "The WebSocket Protocol",
RFC 6455, December 2011.
16.2. Informative References
[RFC2606] Eastlake, D. and A. Panitz, "Reserved Top Level DNS
Names", BCP 32, RFC 2606, June 1999.
[RFC2616] Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext
Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999.
[RFC3327] Willis, D. and B. Hoeneisen, "Session Initiation Protocol
(SIP) Extension Header Field for Registering Non-Adjacent
Contacts", RFC 3327, December 2002.
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[RFC3581] Rosenberg, J. and H. Schulzrinne, "An Extension to the
Session Initiation Protocol (SIP) for Symmetric Response
Routing", RFC 3581, August 2003.
[RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Syntax", STD 66,
RFC 3986, January 2005.
[RFC4168] Rosenberg, J., Schulzrinne, H., and G. Camarillo, "The
Stream Control Transmission Protocol (SCTP) as a Transport
for the Session Initiation Protocol (SIP)", RFC 4168,
October 2005.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246, August 2008.
[RFC5658] Froment, T., Lebel, C., and B. Bonnaerens, "Addressing
Record-Route Issues in the Session Initiation Protocol
(SIP)", RFC 5658, October 2009.
[RFC6265] Barth, A., "HTTP State Management Mechanism", RFC 6265,
April 2011.
[WS-API] Hickson, I., "The Web Sockets API", September 2010.
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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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