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Keywords: SIP, WebSocket
Internet Engineering Task Force (IETF) I. Baz Castillo
Request for Comments: 7118 J. Millan Villegas
Category: Standards Track Versatica
ISSN: 2070-1721 V. Pascual
Quobis
January 2014
The WebSocket Protocol as a Transport for the
Session Initiation Protocol (SIP)
Abstract
The WebSocket protocol enables two-way real-time communication
between clients and servers in web-based applications. This document
specifies a WebSocket subprotocol as a reliable transport mechanism
between Session Initiation Protocol (SIP) entities to enable use of
SIP in web-oriented deployments.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc7118.
Copyright Notice
Copyright (c) 2014 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
described in the Simplified BSD License.
Baz Castillo, et al. Standards Track [Page 1]
RFC 7118 WebSocket as a Transport for SIP January 2014
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
2.1. Definitions . . . . . . . . . . . . . . . . . . . . . . . 3
3. The WebSocket Protocol . . . . . . . . . . . . . . . . . . . 3
4. The WebSocket SIP Subprotocol . . . . . . . . . . . . . . . . 4
4.1. Handshake . . . . . . . . . . . . . . . . . . . . . . . . 4
4.2. SIP Encoding . . . . . . . . . . . . . . . . . . . . . . 5
5. SIP WebSocket Transport . . . . . . . . . . . . . . . . . . . 6
5.1. Via Transport Parameter . . . . . . . . . . . . . . . . . 6
5.2. SIP URI Transport Parameter . . . . . . . . . . . . . . . 6
5.3. Via "received" Parameter . . . . . . . . . . . . . . . . 7
5.4. SIP Transport Implementation Requirements . . . . . . . . 7
5.5. Locating a SIP Server . . . . . . . . . . . . . . . . . . 8
6. Connection Keep-Alive . . . . . . . . . . . . . . . . . . . . 8
7. Authentication . . . . . . . . . . . . . . . . . . . . . . . 8
8. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 10
8.1. Registration . . . . . . . . . . . . . . . . . . . . . . 10
8.2. INVITE Dialog through a Proxy . . . . . . . . . . . . . . 12
9. Security Considerations . . . . . . . . . . . . . . . . . . . 16
9.1. Secure WebSocket Connection . . . . . . . . . . . . . . . 16
9.2. Usage of "sips" Scheme . . . . . . . . . . . . . . . . . 16
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 16
10.1. Registration of the WebSocket SIP Subprotocol . . . . . 16
10.2. Registration of New NAPTR Service Field Values . . . . . 17
10.3. SIP/SIPS URI Parameters Subregistry . . . . . . . . . . 17
10.4. Header Fields Subregistry . . . . . . . . . . . . . . . 17
10.5. Header Field Parameters and Parameter Values Subregistry 17
10.6. SIP Transport Subregistry . . . . . . . . . . . . . . . 18
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 18
12. References . . . . . . . . . . . . . . . . . . . . . . . . . 18
12.1. Normative References . . . . . . . . . . . . . . . . . . 18
12.2. Informative References . . . . . . . . . . . . . . . . . 19
Appendix A. Authentication Use Cases . . . . . . . . . . . . . . 21
A.1. Just SIP Authentication . . . . . . . . . . . . . . . . . 21
A.2. Just Web Authentication . . . . . . . . . . . . . . . . . 21
A.3. Cookie-Based Authentication . . . . . . . . . . . . . . . 22
Appendix B. Implementation Guidelines . . . . . . . . . . . . . 22
B.1. SIP WebSocket Client Considerations . . . . . . . . . . . 23
B.2. SIP WebSocket Server Considerations . . . . . . . . . . . 24
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1. Introduction
The WebSocket protocol [RFC6455] enables message exchange between
clients and servers on top of a persistent TCP connection (optionally
secured with Transport Layer Security (TLS) [RFC5246]). The initial
protocol handshake makes use of HTTP [RFC2616] semantics, allowing
the WebSocket protocol to reuse existing HTTP infrastructure.
Modern web browsers include a WebSocket client stack complying with
the WebSocket API [WS-API] as specified by the W3C. It is expected
that other client applications (those running in personal computers
and devices such as smartphones) will also make a WebSocket client
stack available. The specification in this document enables use of
SIP in these scenarios.
This specification defines a WebSocket subprotocol (as defined in
Section 1.9 of [RFC6455]) for transporting SIP messages between a
WebSocket client and server, a reliable and message-boundary-
preserving transport for SIP, and DNS Naming Authority Pointer
(NAPTR) [RFC3403] service values and procedures for SIP entities
implementing the WebSocket transport. Media transport is out of the
scope of this document.
Section 3 in this specification relaxes the requirement in [RFC3261]
by which the SIP server transport MUST add a "received" parameter in
the top Via header in certain circumstances.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
2.1. Definitions
SIP WebSocket Client: A SIP entity capable of opening outbound
connections to WebSocket servers and communicating using the
WebSocket SIP subprotocol as defined by this document.
SIP WebSocket Server: A SIP entity capable of listening for inbound
connections from WebSocket clients and communicating using the
WebSocket SIP subprotocol as defined by this document.
3. The WebSocket Protocol
The WebSocket protocol [RFC6455] is a transport layer on top of TCP
(optionally secured with TLS [RFC5246]) in which both client and
server exchange message units in both directions. The protocol
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defines a connection handshake, WebSocket subprotocol and extensions
negotiation, a frame format for sending application and control data,
a masking mechanism, and status codes for indicating disconnection
causes.
The WebSocket connection handshake is based on HTTP [RFC2616] and
utilizes the HTTP GET method with an "Upgrade" request. This is sent
by the client and then answered by the server (if the negotiation
succeeded) with an HTTP 101 status code. Once the handshake is
completed, the connection upgrades from HTTP to the WebSocket
protocol. This handshake procedure is designed to reuse the existing
HTTP infrastructure. During the connection handshake, the client and
server agree on the application protocol to use on top of the
WebSocket transport. Such an application protocol (also known as a
"WebSocket subprotocol") defines the format and semantics of the
messages exchanged by the endpoints. This could be a custom protocol
or a standardized one (as defined by the WebSocket SIP subprotocol in
this document). Once the HTTP 101 response is processed, both the
client and server reuse the underlying TCP connection for sending
WebSocket messages and control frames to each other. Unlike plain
HTTP, this connection is persistent and can be used for multiple
message exchanges.
WebSocket defines message units to be used by applications for the
exchange of data, so it provides a message-boundary-preserving
transport layer. These message units can contain either UTF-8 text
or binary data and can be split into multiple WebSocket text/binary
transport frames as needed by the WebSocket stack.
The WebSocket API [WS-API] for web browsers only defines callbacks
to be invoked upon receipt of an entire message unit, regardless
of whether it was received in a single WebSocket frame or split
across multiple frames.
4. The WebSocket SIP Subprotocol
The term WebSocket subprotocol refers to an application-level
protocol layered on top of a WebSocket connection. This document
specifies the WebSocket SIP subprotocol for carrying SIP requests and
responses through a WebSocket connection.
4.1. Handshake
The SIP WebSocket Client and SIP WebSocket Server negotiate usage of
the WebSocket SIP subprotocol during the WebSocket handshake
procedure as defined in Section 1.3 of [RFC6455]. The client MUST
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include the value "sip" in the Sec-WebSocket-Protocol header in its
handshake request. The 101 reply from the server MUST contain "sip"
in its corresponding Sec-WebSocket-Protocol header.
The WebSocket client initiates a WebSocket connection when
attempting to send a SIP request (unless there is an already
established WebSocket connection for sending the SIP request). In
case there is no HTTP 101 response during the WebSocket handshake,
it is considered a transaction error as per [RFC3261],
Section 8.1.3.1., "Transaction Layer Errors".
Below is an example of a WebSocket handshake in which the client
requests the WebSocket SIP subprotocol support from the server:
GET / HTTP/1.1
Host: sip-ws.example.com
Upgrade: websocket
Connection: Upgrade
Sec-WebSocket-Key: dGhlIHNhbXBsZSBub25jZQ==
Origin: http://www.example.com
Sec-WebSocket-Protocol: sip
Sec-WebSocket-Version: 13
The handshake response from the server accepting the WebSocket SIP
subprotocol would look as follows:
HTTP/1.1 101 Switching Protocols
Upgrade: websocket
Connection: Upgrade
Sec-WebSocket-Accept: s3pPLMBiTxaQ9kYGzzhZRbK+xOo=
Sec-WebSocket-Protocol: sip
Once the negotiation has been completed, the WebSocket connection is
established and can be used for the transport of SIP requests and
responses. Messages other than SIP requests and responses MUST NOT
be transmitted over this connection.
4.2. SIP Encoding
WebSocket messages can be transported in either UTF-8 text frames or
binary frames. SIP [RFC3261] allows both text and binary bodies in
SIP requests and responses. Therefore, SIP WebSocket Clients and SIP
WebSocket Servers MUST accept both text and binary frames.
If there is at least one non-UTF-8 symbol in the whole SIP message
(including headers and the body), then the whole message MUST be
sent within a WebSocket binary message. Given the nature of
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JavaScript and the WebSocket API, it is RECOMMENDED to use UTF-8
encoding (or ASCII, which is a subset of UTF-8) for SIP messages
carried over a WebSocket connection.
5. SIP WebSocket Transport
WebSocket [RFC6455] is a reliable protocol; therefore, the SIP
WebSocket subprotocol defined by this document is a reliable SIP
transport. Thus, client and server transactions using WebSocket for
transport MUST follow the procedures and timer values for reliable
transports as defined in [RFC3261].
Each SIP message MUST be carried within a single WebSocket message,
and a WebSocket message MUST NOT contain more than one SIP message.
Because the WebSocket transport preserves message boundaries, the use
of the Content-Length header in SIP messages is not necessary when
they are transported using the WebSocket subprotocol.
This simplifies the parsing of SIP messages for both clients and
servers. There is no need to establish message boundaries using
Content-Length headers between messages. Other SIP transports,
such as UDP and the Stream Control Transmission Protocol (SCTP)
[RFC4168], also provide this benefit.
5.1. Via Transport Parameter
Via header fields in SIP messages carry a transport protocol
identifier. This document defines the value "WS" to be used for
requests over plain WebSocket connections and "WSS" for requests over
secure WebSocket connections (in which the WebSocket connection is
established using TLS [RFC5246] with TCP transport).
The updated augmented BNF (Backus-Naur Form) [RFC5234] for this
parameter is the following (the original BNF for this parameter can
be found in [RFC3261], which was then updated by [RFC4168]):
transport =/ "WS" / "WSS"
5.2. SIP URI Transport Parameter
This document defines the value "ws" as the transport parameter value
for a SIP URI [RFC3986] to be contacted using the SIP WebSocket
subprotocol as transport.
The updated augmented BNF for this parameter is the following (the
original BNF for this parameter can be found in [RFC3261]):
transport-param =/ "transport=" "ws"
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5.3. Via "received" Parameter
The following is stated in [RFC3261], Section 18.2.1, "Receiving
Requests":
When the server transport receives a request over any transport,
it MUST examine the value of the "sent-by" parameter in the top
Via header field value. If the host portion of the "sent-by"
field contains a domain name, or if it contains an IP address that
differs from the packet source address, the server MUST add a
"received" parameter to that Via header field value. This
parameter MUST contain the source address from which the packet
was received.
The requirement of adding the "received" parameter does not fit well
into the WebSocket protocol design. The WebSocket connection
handshake reuses the existing HTTP infrastructure in which there
could be an unknown number of HTTP proxies and/or TCP load balancers
between the SIP WebSocket Client and Server, so the source address
the server would write into the Via "received" parameter would be the
address of the HTTP/TCP intermediary in front of it. This could
reveal sensitive information about the internal topology of the
server's network to the client.
Given the fact that SIP responses can only be sent over the existing
WebSocket connection, the Via "received" parameter is of little use.
Therefore, in order to allow hiding possible sensitive information
about the SIP WebSocket Server's network, this document updates
[RFC3261], Section 18.2.1 by stating:
When a SIP WebSocket Server receives a request, it MAY decide not
to add a "received" parameter to the top Via header. Therefore,
SIP WebSocket Clients MUST accept responses without such a
parameter in the top Via header regardless of whether the Via
"sent-by" field contains a domain name.
5.4. SIP Transport Implementation Requirements
The following is stated in [RFC3261], Section 18, "Transport":
All SIP elements MUST implement UDP and TCP. SIP elements MAY
implement other protocols.
The specification of this transport enables SIP to be used as a
session establishment protocol in scenarios where none of the other
transport protocols defined for SIP can be used. Since some
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environments do not enable SIP elements to use UDP and TCP as SIP
transport protocols, a SIP element acting as a SIP WebSocket Client
is not mandated to implement support of UDP and TCP.
5.5. Locating a SIP Server
[RFC3263] specifies the procedures that should be followed by SIP
entities for locating SIP servers. This specification defines the
NAPTR service value "SIP+D2W" for SIP WebSocket Servers that support
plain WebSocket connections and "SIPS+D2W" for SIP WebSocket Servers
that support secure WebSocket connections.
At the time this document was written, DNS NAPTR/Service Record
(SRV) queries could not be performed by commonly available
WebSocket client stacks (in JavaScript engines and web browsers).
In the absence of DNS SRV resource records or an explicit port, the
default port for a SIP URI using the "sip" scheme and the "ws"
transport parameter is 80, and the default port for a SIP URI using
the "sips" scheme and the "ws" transport parameter is 443.
6. Connection Keep-Alive
SIP WebSocket Clients and Servers may keep their WebSocket
connections open by sending periodic WebSocket "Ping" frames as
described in [RFC6455], Section 5.5.2.
The WebSocket API [WS-API] does not provide a mechanism for
applications running in a web browser to control whether or not
periodic WebSocket "Ping" frames are sent to the server. The
implementation of such a keep-alive feature is the decision of
each web browser manufacturer and may also depend on the
configuration of the web browser.
The indication and use of the CRLF NAT keep-alive mechanism defined
for SIP connection-oriented transports in [RFC5626], Section 3.5.1 or
[RFC6223] are, of course, usable over the transport defined in this
specification.
7. Authentication
This section describes how authentication is achieved through the
requirements in [RFC6455], [RFC6265], [RFC2617], and [RFC3261].
The WebSocket protocol [RFC6455] does not define an authentication
mechanism; instead, it exposes the following text in Section 10.5,
"WebSocket Client Authentication":
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RFC 7118 WebSocket as a Transport for SIP January 2014
This protocol doesn't prescribe any particular way that servers
can authenticate clients during the WebSocket handshake. The
WebSocket server can use any client authentication mechanism
available to a generic HTTP server, such as cookies, HTTP
authentication, or TLS authentication.
The following list exposes mandatory-to-implement and optional
mechanisms for SIP WebSocket Clients and Servers in order to get
interoperability at the WebSocket authentication level:
o A SIP WebSocket Client MUST be ready to add a session cookie when
it runs in a web browser (or behaves like a browser navigating a
website) and has previously retrieved a session cookie from the
web server whose URL domain matches the domain in the WebSocket
URI. This mechanism is defined by [RFC6265].
o A SIP WebSocket Client MUST be ready to be challenged with an HTTP
401 status code [RFC2617] by the SIP WebSocket Server when
performing the WebSocket handshake.
o A SIP WebSocket Client MAY use TLS client authentication (when in
a secure WebSocket connection) as an optional authentication
mechanism.
Note, however, that TLS client authentication in the WebSocket
protocol is governed by the rules of the HTTP protocol rather
than the rules of SIP.
o A SIP WebSocket Server MUST be ready to read session cookies when
present in the WebSocket handshake request and use such a cookie
value for determining whether the WebSocket connection has been
initiated by an HTTP client navigating a website in the same
domain (or subdomain) as the SIP WebSocket Server.
o A SIP WebSocket Server SHOULD be able to reject a WebSocket
handshake request with an HTTP 401 status code by providing a
Basic/Digest challenge as defined for the HTTP protocol.
Regardless of whether or not the SIP WebSocket Server requires
authentication during the WebSocket handshake, authentication MAY be
requested at the SIP level.
Some authentication use cases are exposed in Appendix A.
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8. Examples
8.1. Registration
Alice (SIP WSS) proxy.example.com
| |
|HTTP GET (WS handshake) F1 |
|---------------------------->|
|101 Switching Protocols F2 |
|<----------------------------|
| |
|REGISTER F3 |
|---------------------------->|
|200 OK F4 |
|<----------------------------|
| |
Alice loads a web page using her web browser and retrieves JavaScript
code implementing the WebSocket SIP subprotocol defined in this
document. The JavaScript code (a SIP WebSocket Client) establishes a
secure WebSocket connection with a SIP proxy/registrar (a SIP
WebSocket Server) at proxy.example.com. Upon WebSocket connection,
Alice constructs and sends a SIP REGISTER request, including Outbound
[RFC5626] and Globally Routable User Agent URI (GRUU) [RFC5627]
support. Since the JavaScript stack in a browser has no way to
determine the local address from which the WebSocket connection was
made, this implementation uses a random ".invalid" domain name for
the Via header "sent-by" parameter and for the hostport of the URI in
the Contact header (see Appendix B.1).
Message details (authentication and Session Description Protocol
(SDP) bodies are omitted for simplicity):
F1 HTTP GET (WS handshake) Alice -> proxy.example.com (TLS)
GET / HTTP/1.1
Host: proxy.example.com
Upgrade: websocket
Connection: Upgrade
Sec-WebSocket-Key: dGhlIHNhbXBsZSBub25jZQ==
Origin: https://www.example.com
Sec-WebSocket-Protocol: sip
Sec-WebSocket-Version: 13
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F2 101 Switching Protocols proxy.example.com -> Alice (TLS)
HTTP/1.1 101 Switching Protocols
Upgrade: websocket
Connection: Upgrade
Sec-WebSocket-Accept: s3pPLMBiTxaQ9kYGzzhZRbK+xOo=
Sec-WebSocket-Protocol: sip
F3 REGISTER Alice -> proxy.example.com (transport WSS)
REGISTER sip:proxy.example.com SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKasudf
From: sip:alice@example.com;tag=65bnmj.34asd
To: sip:alice@example.com
Call-ID: aiuy7k9njasd
CSeq: 1 REGISTER
Max-Forwards: 70
Supported: path, outbound, gruu
Contact: <sip:alice@df7jal23ls0d.invalid;transport=ws>
;reg-id=1
;+sip.instance="<urn:uuid:f81-7dec-14a06cf1>"
F4 200 OK proxy.example.com -> Alice (transport WSS)
SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKasudf
From: sip:alice@example.com;tag=65bnmj.34asd
To: sip:alice@example.com;tag=12isjljn8
Call-ID: aiuy7k9njasd
CSeq: 1 REGISTER
Supported: outbound, gruu
Contact: <sip:alice@df7jal23ls0d.invalid;transport=ws>
;reg-id=1
;+sip.instance="<urn:uuid:f81-7dec-14a06cf1>"
;pub-gruu="sip:alice@example.com;gr=urn:uuid:f81-7dec-14a06cf1"
;temp-gruu="sip:87ash54=3dd.98a@example.com;gr"
;expires=3600
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8.2. INVITE Dialog through a Proxy
Alice (SIP WSS) proxy.example.com (SIP UDP) Bob
| | |
|INVITE F1 | |
|---------------------------->| |
|100 Trying F2 | |
|<----------------------------| |
| |INVITE F3 |
| |---------------------------->|
| |200 OK F4 |
| |<----------------------------|
|200 OK F5 | |
|<----------------------------| |
| | |
|ACK F6 | |
|---------------------------->| |
| |ACK F7 |
| |---------------------------->|
| | |
| Bidirectional RTP Media |
|<=========================================================>|
| | |
| |BYE F8 |
| |<----------------------------|
|BYE F9 | |
|<----------------------------| |
|200 OK F10 | |
|---------------------------->| |
| |200 OK F11 |
| |---------------------------->|
| | |
In the same scenario, Alice places a call to Bob's Address of Record
(AOR). The SIP WebSocket Server at proxy.example.com acts as a SIP
proxy, routing the INVITE to Bob's contact address (which happens to
be using SIP transported over UDP). Bob answers the call and then
terminates it.
Message details (authentication and SDP bodies are omitted for
simplicity):
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F1 INVITE Alice -> proxy.example.com (transport WSS)
INVITE sip:bob@example.com SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
From: sip:alice@example.com;tag=asdyka899
To: sip:bob@example.com
Call-ID: asidkj3ss
CSeq: 1 INVITE
Max-Forwards: 70
Supported: path, outbound, gruu
Route: <sip:proxy.example.com:443;transport=ws;lr>
Contact: <sip:alice@example.com
;gr=urn:uuid:f81-7dec-14a06cf1;ob>
Content-Type: application/sdp
F2 100 Trying proxy.example.com -> Alice (transport WSS)
SIP/2.0 100 Trying
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
From: sip:alice@example.com;tag=asdyka899
To: sip:bob@example.com
Call-ID: asidkj3ss
CSeq: 1 INVITE
F3 INVITE proxy.example.com -> Bob (transport UDP)
INVITE sip:bob@203.0.113.22:5060 SIP/2.0
Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bKhjhjqw32c
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
Record-Route: <sip:proxy.example.com;transport=udp;lr>,
<sip:h7kjh12s@proxy.example.com:443;transport=ws;lr>
From: sip:alice@example.com;tag=asdyka899
To: sip:bob@example.com
Call-ID: asidkj3ss
CSeq: 1 INVITE
Max-Forwards: 69
Supported: path, outbound, gruu
Contact: <sip:alice@example.com
;gr=urn:uuid:f81-7dec-14a06cf1;ob>
Content-Type: application/sdp
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F4 200 OK Bob -> proxy.example.com (transport UDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bKhjhjqw32c
;received=192.0.2.10
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
Record-Route: <sip:proxy.example.com;transport=udp;lr>,
<sip:h7kjh12s@proxy.example.com:443;transport=ws;lr>
From: sip:alice@example.com;tag=asdyka899
To: sip:bob@example.com;tag=bmqkjhsd
Call-ID: asidkj3ss
CSeq: 1 INVITE
Contact: <sip:bob@203.0.113.22:5060;transport=udp>
Content-Type: application/sdp
F5 200 OK proxy.example.com -> Alice (transport WSS)
SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
Record-Route: <sip:proxy.example.com;transport=udp;lr>,
<sip:h7kjh12s@proxy.example.com:443;transport=ws;lr>
From: sip:alice@example.com;tag=asdyka899
To: sip:bob@example.com;tag=bmqkjhsd
Call-ID: asidkj3ss
CSeq: 1 INVITE
Contact: <sip:bob@203.0.113.22:5060;transport=udp>
Content-Type: application/sdp
F6 ACK Alice -> proxy.example.com (transport WSS)
ACK sip:bob@203.0.113.22:5060;transport=udp SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKhgqqp090
Route: <sip:h7kjh12s@proxy.example.com:443;transport=ws;lr>,
<sip:proxy.example.com;transport=udp;lr>,
From: sip:alice@example.com;tag=asdyka899
To: sip:bob@example.com;tag=bmqkjhsd
Call-ID: asidkj3ss
CSeq: 1 ACK
Max-Forwards: 70
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F7 ACK proxy.example.com -> Bob (transport UDP)
ACK sip:bob@203.0.113.22:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bKhwpoc80zzx
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKhgqqp090
From: sip:alice@example.com;tag=asdyka899
To: sip:bob@example.com;tag=bmqkjhsd
Call-ID: asidkj3ss
CSeq: 1 ACK
Max-Forwards: 69
F8 BYE Bob -> proxy.example.com (transport UDP)
BYE sip:alice@example.com;gr=urn:uuid:f81-7dec-14a06cf1;ob SIP/2.0
Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001
Route: <sip:proxy.example.com;transport=udp;lr>,
<sip:h7kjh12s@proxy.example.com:443;transport=ws;lr>
From: sip:bob@example.com;tag=bmqkjhsd
To: sip:alice@example.com;tag=asdyka899
Call-ID: asidkj3ss
CSeq: 1201 BYE
Max-Forwards: 70
F9 BYE proxy.example.com -> Alice (transport WSS)
BYE sip:alice@example.com;gr=urn:uuid:f81-7dec-14a06cf1;ob SIP/2.0
Via: SIP/2.0/WSS proxy.example.com:443;branch=z9hG4bKmma01m3r5
Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001
From: sip:bob@example.com;tag=bmqkjhsd
To: sip:alice@example.com;tag=asdyka899
Call-ID: asidkj3ss
CSeq: 1201 BYE
Max-Forwards: 69
F10 200 OK Alice -> proxy.example.com (transport WSS)
SIP/2.0 200 OK
Via: SIP/2.0/WSS proxy.example.com:443;branch=z9hG4bKmma01m3r5
Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001
From: sip:bob@example.com;tag=bmqkjhsd
To: sip:alice@example.com;tag=asdyka899
Call-ID: asidkj3ss
CSeq: 1201 BYE
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F11 200 OK proxy.example.com -> Bob (transport UDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.0.113.22;branch=z9hG4bKbiuiansd001
From: sip:bob@example.com;tag=bmqkjhsd
To: sip:alice@example.com;tag=asdyka899
Call-ID: asidkj3ss
CSeq: 1201 BYE
9. Security Considerations
9.1. Secure WebSocket Connection
It is RECOMMENDED that the SIP traffic transported over a WebSocket
communication be protected by using a secure WebSocket connection
(using TLS [RFC5246] over TCP).
When establishing a connection using SIP over secure WebSocket
transport, the client MUST authenticate the server using the server's
certificate according to the WebSocket validation procedure in
[RFC6455].
Server operators should note that this authentication procedure is
different from the procedure for SIP domain certificates defined
in [RFC5922]. Certificates that are appropriate for SIP over TLS
over TCP will probably not be appropriate for SIP over secure
WebSocket connections.
9.2. Usage of "sips" Scheme
The "sips" scheme in a SIP URI dictates that the entire request path
to the target be secure. If such a path includes a WebSocket
connection, it MUST be a secure WebSocket connection.
10. IANA Considerations
10.1. Registration of the WebSocket SIP Subprotocol
IANA has registered the WebSocket SIP subprotocol under the
"WebSocket Subprotocol Name" registry with the following data:
Subprotocol Identifier: sip
Subprotocol Common Name: WebSocket Transport for SIP (Session
Initiation Protocol)
Subprotocol Definition: [RFC7118]
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10.2. Registration of New NAPTR Service Field Values
This document defines two new NAPTR service field values (SIP+D2W and
SIPS+D2W) and IANA has registered these values under the "Registry
for the Session Initiation Protocol (SIP) NAPTR Resource Record
Services Field". The entries are as follows:
Services Field Protocol Reference
-------------- -------- ---------
SIP+D2W WS [RFC7118]
SIPS+D2W WS [RFC7118]
10.3. SIP/SIPS URI Parameters Subregistry
IANA has added a reference to this document under the "SIP/SIPS URI
Parameters" subregistry within the "Session Initiation Protocol (SIP)
Parameters" registry:
Parameter Name Predefined Values Reference
-------------- ----------------- ---------
transport Yes [RFC3261][RFC7118]
10.4. Header Fields Subregistry
IANA has added a reference to this document under the "Header Fields"
subregistry within the "Session Initiation Protocol (SIP) Parameters"
registry:
Header Name compact Reference
----------- ------- ---------
Via v [RFC3261][RFC7118]
10.5. Header Field Parameters and Parameter Values Subregistry
IANA has added a reference to this document under the "Header Field
Parameters and Parameter Values" subregistry within the "Session
Initiation Protocol (SIP) Parameters" registry:
Predefined
Header Field Parameter Name Values Reference
------------ -------------- ------ ---------
Via received No [RFC3261][RFC7118]
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10.6. SIP Transport Subregistry
This document adds a new subregistry, "SIP Transport", to the
"Session Initiation Protocol (SIP) Parameters" registry. Its format
and initial values are as shown in the following table:
+------------+------------------------+
| Transport | Reference |
+------------+------------------------+
| UDP | [RFC3261] |
| TCP | [RFC3261] |
| TLS | [RFC3261] |
| SCTP | [RFC3261], [RFC4168] |
| TLS-SCTP | [RFC4168] |
| WS | [RFC7118] |
| WSS | [RFC7118] |
+------------+------------------------+
The policy for registration of values in this registry is "Standards
Action" [RFC5226].
11. Acknowledgements
Special thanks to the following people who participated in
discussions on the SIPCORE and RTCWEB WG mailing lists and
contributed ideas and/or provided detailed reviews (the list is
likely to be incomplete): Hadriel Kaplan, Paul Kyzivat, Robert
Sparks, Adam Roach, Ranjit Avasarala, Xavier Marjou, Nataraju A. B.,
Martin Vopatek, Alexey Melnikov, Alan Johnston, Christer Holmberg,
Salvatore Loreto, Kevin P. Fleming, Suresh Krishnan, Yaron Sheffer,
Richard Barnes, Barry Leiba, Stephen Farrell, Ted Lemon, Benoit
Claise, Pete Resnick, Binod P.G., and Saul Ibarra Corretge.
12. References
12.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[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.
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[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3263] Rosenberg, J. and H. Schulzrinne, "Session Initiation
Protocol (SIP): Locating SIP Servers", RFC 3263, June
2002.
[RFC3403] Mealling, M., "Dynamic Delegation Discovery System (DDDS)
Part Three: The Domain Name System (DNS) Database", RFC
3403, October 2002.
[RFC5226] Narten, T. and H. Alvestrand, "Guidelines for Writing an
IANA Considerations Section in RFCs", BCP 26, RFC 5226,
May 2008.
[RFC5234] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", STD 68, RFC 5234, January 2008.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246, August 2008.
[RFC6265] Barth, A., "HTTP State Management Mechanism", RFC 6265,
April 2011.
[RFC6455] Fette, I. and A. Melnikov, "The WebSocket Protocol", 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.
[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.
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[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.
[RFC5626] Jennings, C., Mahy, R., and F. Audet, "Managing Client-
Initiated Connections in the Session Initiation Protocol
(SIP)", RFC 5626, October 2009.
[RFC5627] Rosenberg, J., "Obtaining and Using Globally Routable User
Agent URIs (GRUUs) in the Session Initiation Protocol
(SIP)", RFC 5627, October 2009.
[RFC5922] Gurbani, V., Lawrence, S., and A. Jeffrey, "Domain
Certificates in the Session Initiation Protocol (SIP)",
RFC 5922, June 2010.
[RFC6223] Holmberg, C., "Indication of Support for Keep-Alive", RFC
6223, April 2011.
[WS-API] W3C and I. Hickson, Ed., "The WebSocket API", September
2012.
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Appendix A. Authentication Use Cases
The sections below briefly describe some SIP over WebSocket scenarios
in which authentication takes place in different ways.
A.1. Just SIP Authentication
SIP Private Branch Exchange (PBX) model A implements the SIP
WebSocket transport defined by this specification. Its
implementation is 100% website agnostic as it does not share
information with the web server providing the HTML code to browsers,
meaning that the SIP WebSocket Server (here, PBX model A) has no
knowledge about web login activity within the website.
In this simple scenario, the SIP WebSocket Server does not inspect
fields in the WebSocket handshake HTTP GET request such as the
request URL, the Origin header value, the Host header value, or the
Cookie header value (if present). However, some of those fields
could be inspected for a minimal validation (i.e., PBX model A could
require that the Origin header value contains a specific URL so just
users navigating such a website would be able to establish a
WebSocket connection with PBX model A).
Once the WebSocket connection has been established, SIP
authentication is requested by PBX model A for each SIP request
coming over that connection. Therefore, SIP WebSocket Clients must
be provisioned with their corresponding SIP password.
A.2. Just Web Authentication
A SIP-to-PSTN (Public Switched Telephone Network) provider offers
telephony service for clients logged into its website. The provider
does not want to expose SIP passwords into the web for security/
privacy reasons.
Once the user is logged into the web, the web server provides him
with a SIP identity (SIP URI) and a session temporary token string
(along with the SIP WebSocket Client JavaScript application and SIP
settings). The web server stores the SIP identity and session token
into a database.
The web application adds the SIP identity and session token as URL
query parameters in the WebSocket handshake request and attempts the
connection. The SIP WebSocket Server inspects the handshake request
and validates that the session token matches the value stored in the
database for the given SIP identity. In case the value matches, the
WebSocket connection gets "authenticated" for that SIP identity. The
SIP WebSocket Client can then register and make calls. The SIP
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WebSocket Server would, however, verify that the identity in those
SIP requests (i.e., the From URI value) matches the SIP identity the
WebSocket connection is associated to (otherwise, the SIP request is
rejected).
When the user performs a logout action in the web, the web server
removes the SIP identity and session token tuple from the database
and notifies the SIP WebSocket Server, which revokes and closes the
WebSocket connection.
No SIP authentication takes place in this scenario.
A.3. Cookie-Based Authentication
The Apache web server comes with a new module: mod_sip_websocket. In
port 80, the web server is configured to listen for both HTTP common
requests and WebSocket handshake requests. Therefore, both the web
server and the SIP WebSocket Server are co-located within the same
host and same domain.
Once the user is logged into the web, he is provided with the SIP
WebSocket Client JavaScript application and SIP settings. The HTTP
200 response after the login procedure also contains a session cookie
[RFC6265]. The web application then attempts a WebSocket connection
against the same URL/domain of the website, and thus the session
cookie is automatically added by the browser into the WebSocket
handshake request (as the WebSocket protocol [RFC6455] states).
The web server inspects the cookie value (as it would do for a common
HTTP request containing a session cookie so that the login procedure
is not required again). If the cookie is valid, the WebSocket
connection is authorized. And, as in the previous use case, the
connection is also associated with a specific SIP identity that must
be satisfied by every SIP request coming over that connection.
No SIP authentication takes place in this scenario but just common
cookie usage as widely deployed in the World Wide Web.
Appendix B. Implementation Guidelines
Let us assume a scenario in which the users access with their web
browsers (probably behind NAT) an application provided by a server on
an intranet, login by entering their user identifier and credentials,
and retrieve a JavaScript application (along with the HTML)
implementing a SIP WebSocket Client.
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Such a SIP stack connects to a given SIP WebSocket Server (an
outbound SIP proxy that also implements classic SIP transports such
as UDP and TCP). The HTTP GET method request sent by the web browser
for the WebSocket handshake includes a Cookie [RFC6265] header with
the value previously provided by the server after the successful
login procedure. The cookie value is then inspected by the WebSocket
server to authorize the connection. Once the WebSocket connection is
established, the SIP WebSocket Client performs a SIP registration to
a SIP registrar server that is reachable through the proxy. After
registration, the SIP WebSocket Client and Server exchange SIP
messages as would normally be expected.
This scenario is quite similar to ones in which SIP user agents (UAs)
behind NATs connect to a proxy and must reuse the same TCP connection
for incoming requests (because they are not directly reachable by the
proxy otherwise). In both cases, the SIP UAs are only reachable
through the proxy to which they are connected.
The SIP Outbound extension [RFC5626] seems an appropriate solution
for this scenario. Therefore, these SIP WebSocket Clients and the
SIP registrar implement both the Outbound and Path [RFC3327]
extensions, and the SIP proxy acts as an Outbound Edge Proxy (as
defined in [RFC5626], Section 3.4).
SIP WebSocket Clients in this scenario receive incoming SIP requests
via the SIP WebSocket Server to which they are connected. Therefore,
in some call transfer cases, the use of GRUU [RFC5627] (which should
be implemented in both the SIP WebSocket Clients and SIP registrar)
is valuable.
If a REFER request is sent to a third SIP user agent including the
Contact URI of a SIP WebSocket Client as the target in its
Refer-To header field, such a URI will be reachable by the third
SIP UA only if it is a globally routable URI. GRUU (Globally
Routable User Agent URI) is a solution for those scenarios and
would cause the incoming request from the third SIP user agent to
be sent to the SIP registrar, which would route the request to the
SIP WebSocket Client via the Outbound Edge Proxy.
B.1. SIP WebSocket Client Considerations
The JavaScript stack in web browsers does not have the ability to
discover the local transport address used for originating WebSocket
connections. A SIP WebSocket Client running in such an environment
can construct a domain name consisting of a random token followed by
the ".invalid" top-level domain name, as stated in [RFC2606], and
uses it within its Via and Contact headers.
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The Contact URI provided by SIP UAs requesting (and receiving)
Outbound support is not used for routing requests to those UAs,
thus it is safe to set a random domain in the Contact URI
hostport.
Both the Outbound and GRUU specifications require a SIP UA to include
a Uniform Resource Name (URN) in a "+sip.instance" parameter of the
Contact header in which they include their SIP REGISTER requests.
The client device is responsible for generating or collecting a
suitable value for this purpose.
In web browsers, it is difficult to generate or collect a suitable
value to be used as an 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.
B.2. SIP WebSocket Server Considerations
The SIP WebSocket Server in this scenario behaves as a SIP Outbound
Edge Proxy, which involves support for Outbound [RFC5626] and Path
[RFC3327].
The proxy performs loose routing and remains in the path of dialogs
as specified in [RFC3261]. If it did not do this, in-dialog requests
would fail since SIP WebSocket Clients make use of their SIP
WebSocket Server in order to send and receive SIP messages.
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Authors' Addresses
Inaki Baz Castillo
Versatica
Barakaldo, Basque Country
Spain
EMail: ibc@aliax.net
Jose Luis Millan Villegas
Versatica
Bilbao, Basque Country
Spain
EMail: jmillan@aliax.net
Victor Pascual
Quobis
Spain
EMail: victor.pascual@quobis.com
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