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<meta name="DC.Identifier" content="urn:ietf:rfc:6455">
<meta name="DC.Date.Issued" content="December, 2011">
<meta name="DC.Creator" content="Ian Fette <[email protected]>">
<meta name="DC.Description.Abstract" content="The WebSocket Protocol enables two-way communication between a client
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<span class="pre noprint docinfo top">[<a href="https://tools.ietf.org/html/" title="Document search and retrieval page">Docs</a>] [<a href="https://tools.ietf.org/rfc/rfc6455.txt" title="Plaintext version of this document">txt</a>|<a href="https://tools.ietf.org/pdf/rfc6455" title="PDF version of this document">pdf</a>] [<a href="https://tools.ietf.org/html/draft-ietf-hybi-thewebsocketprotocol" title="draft-ietf-hybi-thewebsocketprotocol">draft-ietf-hybi-t...</a>] [<a href="https://tools.ietf.org/rfcdiff?difftype=--hwdiff&url2=rfc6455" title="Inline diff (wdiff)">Diff1</a>] [<a href="https://tools.ietf.org/rfcdiff?url2=rfc6455" title="Side-by-side diff">Diff2</a>] [<a href="https://www.rfc-editor.org/errata_search.php?rfc=6455">Errata</a>] </span><br>
<span class="pre noprint docinfo"> </span><br>
<span class="pre noprint docinfo">Updated by: <a href="https://tools.ietf.org/html/rfc7936">7936</a> PROPOSED STANDARD</span><br>
<span class="pre noprint docinfo"> <span style="color: #C00;">Errata Exist</span></span><br>
<pre>Internet Engineering Task Force (IETF) I. Fette
Request for Comments: 6455 Google, Inc.
Category: Standards Track A. Melnikov
ISSN: 2070-1721 Isode Ltd.
December 2011
<span class="h1"><h1>The WebSocket Protocol</h1></span>
Abstract
The WebSocket Protocol enables two-way communication between a client
running untrusted code in a controlled environment to a remote host
that has opted-in to communications from that code. The security
model used for this is the origin-based security model commonly used
by web browsers. The protocol consists of an opening handshake
followed by basic message framing, layered over TCP. The goal of
this technology is to provide a mechanism for browser-based
applications that need two-way communication with servers that does
not rely on opening multiple HTTP connections (e.g., using
XMLHttpRequest or <iframe>s and long polling).
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 <a href="https://tools.ietf.org/html/rfc5741#section-2">Section 2 of RFC 5741</a>.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
<a href="http://www.rfc-editor.org/info/rfc6455">http://www.rfc-editor.org/info/rfc6455</a>.
Copyright Notice
Copyright (c) 2011 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to <a href="https://tools.ietf.org/html/bcp78">BCP 78</a> and the IETF Trust's Legal
Provisions Relating to IETF Documents
(<a href="http://trustee.ietf.org/license-info">http://trustee.ietf.org/license-info</a>) 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
<span class="grey">Fette & Melnikov Standards Track [Page 1]</span></pre>
<hr class="noprint" style="width: 96ex;" align="left"><!--NewPage--><pre class="newpage"><a name="page-2" id="page-2" href="#page-2" class="invisible"> </a>
<span class="grey"><a href="https://tools.ietf.org/html/rfc6455">RFC 6455</a> The WebSocket Protocol December 2011</span>
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
<a href="#section-1">1</a>. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-4">4</a>
<a href="#section-1.1">1.1</a>. Background . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-4">4</a>
<a href="#section-1.2">1.2</a>. Protocol Overview . . . . . . . . . . . . . . . . . . . . <a href="#page-5">5</a>
<a href="#section-1.3">1.3</a>. Opening Handshake . . . . . . . . . . . . . . . . . . . . <a href="#page-6">6</a>
<a href="#section-1.4">1.4</a>. Closing Handshake . . . . . . . . . . . . . . . . . . . . <a href="#page-9">9</a>
<a href="#section-1.5">1.5</a>. Design Philosophy . . . . . . . . . . . . . . . . . . . . <a href="#page-9">9</a>
<a href="#section-1.6">1.6</a>. Security Model . . . . . . . . . . . . . . . . . . . . . . <a href="#page-10">10</a>
<a href="#section-1.7">1.7</a>. Relationship to TCP and HTTP . . . . . . . . . . . . . . . <a href="#page-11">11</a>
<a href="#section-1.8">1.8</a>. Establishing a Connection . . . . . . . . . . . . . . . . <a href="#page-11">11</a>
<a href="#section-1.9">1.9</a>. Subprotocols Using the WebSocket Protocol . . . . . . . . <a href="#page-12">12</a>
<a href="#section-2">2</a>. Conformance Requirements . . . . . . . . . . . . . . . . . . . <a href="#page-12">12</a>
<a href="#section-2.1">2.1</a>. Terminology and Other Conventions . . . . . . . . . . . . <a href="#page-13">13</a>
<a href="#section-3">3</a>. WebSocket URIs . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-14">14</a>
<a href="#section-4">4</a>. Opening Handshake . . . . . . . . . . . . . . . . . . . . . . <a href="#page-14">14</a>
<a href="#section-4.1">4.1</a>. Client Requirements . . . . . . . . . . . . . . . . . . . <a href="#page-14">14</a>
<a href="#section-4.2">4.2</a>. Server-Side Requirements . . . . . . . . . . . . . . . . . <a href="#page-20">20</a>
<a href="#section-4.2.1">4.2.1</a>. Reading the Client's Opening Handshake . . . . . . . . <a href="#page-21">21</a>
<a href="#section-4.2.2">4.2.2</a>. Sending the Server's Opening Handshake . . . . . . . . <a href="#page-22">22</a>
<a href="#section-4.3">4.3</a>. Collected ABNF for New Header Fields Used in Handshake . . <a href="#page-25">25</a>
<a href="#section-4.4">4.4</a>. Supporting Multiple Versions of WebSocket Protocol . . . . <a href="#page-26">26</a>
<a href="#section-5">5</a>. Data Framing . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-27">27</a>
<a href="#section-5.1">5.1</a>. Overview . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-27">27</a>
<a href="#section-5.2">5.2</a>. Base Framing Protocol . . . . . . . . . . . . . . . . . . <a href="#page-28">28</a>
<a href="#section-5.3">5.3</a>. Client-to-Server Masking . . . . . . . . . . . . . . . . . <a href="#page-32">32</a>
<a href="#section-5.4">5.4</a>. Fragmentation . . . . . . . . . . . . . . . . . . . . . . <a href="#page-33">33</a>
<a href="#section-5.5">5.5</a>. Control Frames . . . . . . . . . . . . . . . . . . . . . . <a href="#page-36">36</a>
<a href="#section-5.5.1">5.5.1</a>. Close . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-36">36</a>
<a href="#section-5.5.2">5.5.2</a>. Ping . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-37">37</a>
<a href="#section-5.5.3">5.5.3</a>. Pong . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-37">37</a>
<a href="#section-5.6">5.6</a>. Data Frames . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-38">38</a>
<a href="#section-5.7">5.7</a>. Examples . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-38">38</a>
<a href="#section-5.8">5.8</a>. Extensibility . . . . . . . . . . . . . . . . . . . . . . <a href="#page-39">39</a>
<a href="#section-6">6</a>. Sending and Receiving Data . . . . . . . . . . . . . . . . . . <a href="#page-39">39</a>
<a href="#section-6.1">6.1</a>. Sending Data . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-39">39</a>
<a href="#section-6.2">6.2</a>. Receiving Data . . . . . . . . . . . . . . . . . . . . . . <a href="#page-40">40</a>
<a href="#section-7">7</a>. Closing the Connection . . . . . . . . . . . . . . . . . . . . <a href="#page-41">41</a>
<a href="#section-7.1">7.1</a>. Definitions . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-41">41</a>
<a href="#section-7.1.1">7.1.1</a>. Close the WebSocket Connection . . . . . . . . . . . . <a href="#page-41">41</a>
<a href="#section-7.1.2">7.1.2</a>. Start the WebSocket Closing Handshake . . . . . . . . <a href="#page-42">42</a>
<a href="#section-7.1.3">7.1.3</a>. The WebSocket Closing Handshake is Started . . . . . . <a href="#page-42">42</a>
<a href="#section-7.1.4">7.1.4</a>. The WebSocket Connection is Closed . . . . . . . . . . <a href="#page-42">42</a>
<a href="#section-7.1.5">7.1.5</a>. The WebSocket Connection Close Code . . . . . . . . . <a href="#page-42">42</a>
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<a href="#section-7.1.6">7.1.6</a>. The WebSocket Connection Close Reason . . . . . . . . <a href="#page-43">43</a>
<a href="#section-7.1.7">7.1.7</a>. Fail the WebSocket Connection . . . . . . . . . . . . <a href="#page-43">43</a>
<a href="#section-7.2">7.2</a>. Abnormal Closures . . . . . . . . . . . . . . . . . . . . <a href="#page-44">44</a>
<a href="#section-7.2.1">7.2.1</a>. Client-Initiated Closure . . . . . . . . . . . . . . . <a href="#page-44">44</a>
<a href="#section-7.2.2">7.2.2</a>. Server-Initiated Closure . . . . . . . . . . . . . . . <a href="#page-44">44</a>
<a href="#section-7.2.3">7.2.3</a>. Recovering from Abnormal Closure . . . . . . . . . . . <a href="#page-44">44</a>
<a href="#section-7.3">7.3</a>. Normal Closure of Connections . . . . . . . . . . . . . . <a href="#page-45">45</a>
<a href="#section-7.4">7.4</a>. Status Codes . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-45">45</a>
<a href="#section-7.4.1">7.4.1</a>. Defined Status Codes . . . . . . . . . . . . . . . . . <a href="#page-45">45</a>
<a href="#section-7.4.2">7.4.2</a>. Reserved Status Code Ranges . . . . . . . . . . . . . <a href="#page-47">47</a>
<a href="#section-8">8</a>. Error Handling . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-48">48</a>
<a href="#section-8.1">8.1</a>. Handling Errors in UTF-8-Encoded Data . . . . . . . . . . <a href="#page-48">48</a>
<a href="#section-9">9</a>. Extensions . . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-48">48</a>
<a href="#section-9.1">9.1</a>. Negotiating Extensions . . . . . . . . . . . . . . . . . . <a href="#page-48">48</a>
<a href="#section-9.2">9.2</a>. Known Extensions . . . . . . . . . . . . . . . . . . . . . <a href="#page-50">50</a>
<a href="#section-10">10</a>. Security Considerations . . . . . . . . . . . . . . . . . . . <a href="#page-50">50</a>
<a href="#section-10.1">10.1</a>. Non-Browser Clients . . . . . . . . . . . . . . . . . . . <a href="#page-50">50</a>
<a href="#section-10.2">10.2</a>. Origin Considerations . . . . . . . . . . . . . . . . . . <a href="#page-50">50</a>
<a href="#section-10.3">10.3</a>. Attacks On Infrastructure (Masking) . . . . . . . . . . . <a href="#page-51">51</a>
<a href="#section-10.4">10.4</a>. Implementation-Specific Limits . . . . . . . . . . . . . . <a href="#page-52">52</a>
<a href="#section-10.5">10.5</a>. WebSocket Client Authentication . . . . . . . . . . . . . <a href="#page-53">53</a>
<a href="#section-10.6">10.6</a>. Connection Confidentiality and Integrity . . . . . . . . . <a href="#page-53">53</a>
<a href="#section-10.7">10.7</a>. Handling of Invalid Data . . . . . . . . . . . . . . . . . <a href="#page-53">53</a>
<a href="#section-10.8">10.8</a>. Use of SHA-1 by the WebSocket Handshake . . . . . . . . . <a href="#page-54">54</a>
<a href="#section-11">11</a>. IANA Considerations . . . . . . . . . . . . . . . . . . . . . <a href="#page-54">54</a>
<a href="#section-11.1">11.1</a>. Registration of New URI Schemes . . . . . . . . . . . . . <a href="#page-54">54</a>
<a href="#section-11.1.1">11.1.1</a>. Registration of "ws" Scheme . . . . . . . . . . . . . <a href="#page-54">54</a>
<a href="#section-11.1.2">11.1.2</a>. Registration of "wss" Scheme . . . . . . . . . . . . . <a href="#page-55">55</a>
<a href="#section-11.2">11.2</a>. Registration of the "WebSocket" HTTP Upgrade Keyword . . . <a href="#page-56">56</a>
<a href="#section-11.3">11.3</a>. Registration of New HTTP Header Fields . . . . . . . . . . <a href="#page-57">57</a>
<a href="#section-11.3.1">11.3.1</a>. Sec-WebSocket-Key . . . . . . . . . . . . . . . . . . <a href="#page-57">57</a>
<a href="#section-11.3.2">11.3.2</a>. Sec-WebSocket-Extensions . . . . . . . . . . . . . . . <a href="#page-58">58</a>
<a href="#section-11.3.3">11.3.3</a>. Sec-WebSocket-Accept . . . . . . . . . . . . . . . . . <a href="#page-58">58</a>
<a href="#section-11.3.4">11.3.4</a>. Sec-WebSocket-Protocol . . . . . . . . . . . . . . . . <a href="#page-59">59</a>
<a href="#section-11.3.5">11.3.5</a>. Sec-WebSocket-Version . . . . . . . . . . . . . . . . <a href="#page-60">60</a>
<a href="#section-11.4">11.4</a>. WebSocket Extension Name Registry . . . . . . . . . . . . <a href="#page-61">61</a>
<a href="#section-11.5">11.5</a>. WebSocket Subprotocol Name Registry . . . . . . . . . . . <a href="#page-61">61</a>
<a href="#section-11.6">11.6</a>. WebSocket Version Number Registry . . . . . . . . . . . . <a href="#page-62">62</a>
<a href="#section-11.7">11.7</a>. WebSocket Close Code Number Registry . . . . . . . . . . . <a href="#page-64">64</a>
<a href="#section-11.8">11.8</a>. WebSocket Opcode Registry . . . . . . . . . . . . . . . . <a href="#page-65">65</a>
<a href="#section-11.9">11.9</a>. WebSocket Framing Header Bits Registry . . . . . . . . . . <a href="#page-66">66</a>
<a href="#section-12">12</a>. Using the WebSocket Protocol from Other Specifications . . . . <a href="#page-66">66</a>
<a href="#section-13">13</a>. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-67">67</a>
<a href="#section-14">14</a>. References . . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-68">68</a>
<a href="#section-14.1">14.1</a>. Normative References . . . . . . . . . . . . . . . . . . . <a href="#page-68">68</a>
<a href="#section-14.2">14.2</a>. Informative References . . . . . . . . . . . . . . . . . . <a href="#page-69">69</a>
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<span class="h2"><h2><a class="selflink" name="section-1" href="#section-1">1</a>. Introduction</h2></span>
<span class="h3"><h3><a class="selflink" name="section-1.1" href="#section-1.1">1.1</a>. Background</h3></span>
_This section is non-normative._
Historically, creating web applications that need bidirectional
communication between a client and a server (e.g., instant messaging
and gaming applications) has required an abuse of HTTP to poll the
server for updates while sending upstream notifications as distinct
HTTP calls [<a href="https://tools.ietf.org/html/rfc6202" title=""Known Issues and Best Practices for the Use of Long Polling and Streaming in Bidirectional HTTP"">RFC6202</a>].
This results in a variety of problems:
o The server is forced to use a number of different underlying TCP
connections for each client: one for sending information to the
client and a new one for each incoming message.
o The wire protocol has a high overhead, with each client-to-server
message having an HTTP header.
o The client-side script is forced to maintain a mapping from the
outgoing connections to the incoming connection to track replies.
A simpler solution would be to use a single TCP connection for
traffic in both directions. This is what the WebSocket Protocol
provides. Combined with the WebSocket API [<a href="#ref-WSAPI" title=""The WebSocket API"">WSAPI</a>], it provides an
alternative to HTTP polling for two-way communication from a web page
to a remote server.
The same technique can be used for a variety of web applications:
games, stock tickers, multiuser applications with simultaneous
editing, user interfaces exposing server-side services in real time,
etc.
The WebSocket Protocol is designed to supersede existing
bidirectional communication technologies that use HTTP as a transport
layer to benefit from existing infrastructure (proxies, filtering,
authentication). Such technologies were implemented as trade-offs
between efficiency and reliability because HTTP was not initially
meant to be used for bidirectional communication (see [<a href="https://tools.ietf.org/html/rfc6202" title=""Known Issues and Best Practices for the Use of Long Polling and Streaming in Bidirectional HTTP"">RFC6202</a>] for
further discussion). The WebSocket Protocol attempts to address the
goals of existing bidirectional HTTP technologies in the context of
the existing HTTP infrastructure; as such, it is designed to work
over HTTP ports 80 and 443 as well as to support HTTP proxies and
intermediaries, even if this implies some complexity specific to the
current environment. However, the design does not limit WebSocket to
HTTP, and future implementations could use a simpler handshake over a
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dedicated port without reinventing the entire protocol. This last
point is important because the traffic patterns of interactive
messaging do not closely match standard HTTP traffic and can induce
unusual loads on some components.
<span class="h3"><h3><a class="selflink" name="section-1.2" href="#section-1.2">1.2</a>. Protocol Overview</h3></span>
_This section is non-normative._
The protocol has two parts: a handshake and the data transfer.
The handshake from the client looks as follows:
GET /chat HTTP/1.1
Host: server.example.com
Upgrade: websocket
Connection: Upgrade
Sec-WebSocket-Key: dGhlIHNhbXBsZSBub25jZQ==
Origin: http://example.com
Sec-WebSocket-Protocol: chat, superchat
Sec-WebSocket-Version: 13
The handshake from the server looks as follows:
HTTP/1.1 101 Switching Protocols
Upgrade: websocket
Connection: Upgrade
Sec-WebSocket-Accept: s3pPLMBiTxaQ9kYGzzhZRbK+xOo=
Sec-WebSocket-Protocol: chat
The leading line from the client follows the Request-Line format.
The leading line from the server follows the Status-Line format. The
Request-Line and Status-Line productions are defined in [<a href="https://tools.ietf.org/html/rfc2616" title=""Hypertext Transfer Protocol -- HTTP/1.1"">RFC2616</a>].
An unordered set of header fields comes after the leading line in
both cases. The meaning of these header fields is specified in
<a href="#section-4">Section 4</a> of this document. Additional header fields may also be
present, such as cookies [<a href="https://tools.ietf.org/html/rfc6265" title=""HTTP State Management Mechanism"">RFC6265</a>]. The format and parsing of
headers is as defined in [<a href="https://tools.ietf.org/html/rfc2616" title=""Hypertext Transfer Protocol -- HTTP/1.1"">RFC2616</a>].
Once the client and server have both sent their handshakes, and if
the handshake was successful, then the data transfer part starts.
This is a two-way communication channel where each side can,
independently from the other, send data at will.
After a successful handshake, clients and servers transfer data back
and forth in conceptual units referred to in this specification as
"messages". On the wire, a message is composed of one or more
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frames. The WebSocket message does not necessarily correspond to a
particular network layer framing, as a fragmented message may be
coalesced or split by an intermediary.
A frame has an associated type. Each frame belonging to the same
message contains the same type of data. Broadly speaking, there are
types for textual data (which is interpreted as UTF-8 [<a href="https://tools.ietf.org/html/rfc3629" title=""UTF-8, a transformation format of ISO 10646"">RFC3629</a>]
text), binary data (whose interpretation is left up to the
application), and control frames (which are not intended to carry
data for the application but instead for protocol-level signaling,
such as to signal that the connection should be closed). This
version of the protocol defines six frame types and leaves ten
reserved for future use.
<span class="h3"><h3><a class="selflink" name="section-1.3" href="#section-1.3">1.3</a>. Opening Handshake</h3></span>
_This section is non-normative._
The opening handshake is intended to be compatible with HTTP-based
server-side software and intermediaries, so that a single port can be
used by both HTTP clients talking to that server and WebSocket
clients talking to that server. To this end, the WebSocket client's
handshake is an HTTP Upgrade request:
GET /chat HTTP/1.1
Host: server.example.com
Upgrade: websocket
Connection: Upgrade
Sec-WebSocket-Key: dGhlIHNhbXBsZSBub25jZQ==
Origin: http://example.com
Sec-WebSocket-Protocol: chat, superchat
Sec-WebSocket-Version: 13
In compliance with [<a href="https://tools.ietf.org/html/rfc2616" title=""Hypertext Transfer Protocol -- HTTP/1.1"">RFC2616</a>], header fields in the handshake may be
sent by the client in any order, so the order in which different
header fields are received is not significant.
The "Request-URI" of the GET method [<a href="https://tools.ietf.org/html/rfc2616" title=""Hypertext Transfer Protocol -- HTTP/1.1"">RFC2616</a>] is used to identify the
endpoint of the WebSocket connection, both to allow multiple domains
to be served from one IP address and to allow multiple WebSocket
endpoints to be served by a single server.
The client includes the hostname in the |Host| header field of its
handshake as per [<a href="https://tools.ietf.org/html/rfc2616" title=""Hypertext Transfer Protocol -- HTTP/1.1"">RFC2616</a>], so that both the client and the server
can verify that they agree on which host is in use.
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Additional header fields are used to select options in the WebSocket
Protocol. Typical options available in this version are the
subprotocol selector (|Sec-WebSocket-Protocol|), list of extensions
support by the client (|Sec-WebSocket-Extensions|), |Origin| header
field, etc. The |Sec-WebSocket-Protocol| request-header field can be
used to indicate what subprotocols (application-level protocols
layered over the WebSocket Protocol) are acceptable to the client.
The server selects one or none of the acceptable protocols and echoes
that value in its handshake to indicate that it has selected that
protocol.
Sec-WebSocket-Protocol: chat
The |Origin| header field [<a href="https://tools.ietf.org/html/rfc6454" title=""The Web Origin Concept"">RFC6454</a>] is used to protect against
unauthorized cross-origin use of a WebSocket server by scripts using
the WebSocket API in a web browser. The server is informed of the
script origin generating the WebSocket connection request. If the
server does not wish to accept connections from this origin, it can
choose to reject the connection by sending an appropriate HTTP error
code. This header field is sent by browser clients; for non-browser
clients, this header field may be sent if it makes sense in the
context of those clients.
Finally, the server has to prove to the client that it received the
client's WebSocket handshake, so that the server doesn't accept
connections that are not WebSocket connections. This prevents an
attacker from tricking a WebSocket server by sending it carefully
crafted packets using XMLHttpRequest [<a href="#ref-XMLHttpRequest">XMLHttpRequest</a>] or a form
submission.
To prove that the handshake was received, the server has to take two
pieces of information and combine them to form a response. The first
piece of information comes from the |Sec-WebSocket-Key| header field
in the client handshake:
Sec-WebSocket-Key: dGhlIHNhbXBsZSBub25jZQ==
For this header field, the server has to take the value (as present
in the header field, e.g., the base64-encoded [<a href="https://tools.ietf.org/html/rfc4648" title=""The Base16, Base32, and Base64 Data Encodings"">RFC4648</a>] version minus
any leading and trailing whitespace) and concatenate this with the
Globally Unique Identifier (GUID, [<a href="https://tools.ietf.org/html/rfc4122" title=""A Universally Unique IDentifier (UUID) URN Namespace"">RFC4122</a>]) "258EAFA5-E914-47DA-
95CA-C5AB0DC85B11" in string form, which is unlikely to be used by
network endpoints that do not understand the WebSocket Protocol. A
SHA-1 hash (160 bits) [<a href="#ref-FIPS.180-3">FIPS.180-3</a>], base64-encoded (see <a href="https://tools.ietf.org/html/rfc4648#section-4">Section 4 of
[RFC4648]</a>), of this concatenation is then returned in the server's
handshake.
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Concretely, if as in the example above, the |Sec-WebSocket-Key|
header field had the value "dGhlIHNhbXBsZSBub25jZQ==", the server
would concatenate the string "258EAFA5-E914-47DA-95CA-C5AB0DC85B11"
to form the string "dGhlIHNhbXBsZSBub25jZQ==258EAFA5-E914-47DA-95CA-
C5AB0DC85B11". The server would then take the SHA-1 hash of this,
giving the value 0xb3 0x7a 0x4f 0x2c 0xc0 0x62 0x4f 0x16 0x90 0xf6
0x46 0x06 0xcf 0x38 0x59 0x45 0xb2 0xbe 0xc4 0xea. This value is
then base64-encoded (see <a href="https://tools.ietf.org/html/rfc4648#section-4">Section 4 of [RFC4648]</a>), to give the value
"s3pPLMBiTxaQ9kYGzzhZRbK+xOo=". This value would then be echoed in
the |Sec-WebSocket-Accept| header field.
The handshake from the server is much simpler than the client
handshake. The first line is an HTTP Status-Line, with the status
code 101:
HTTP/1.1 101 Switching Protocols
Any status code other than 101 indicates that the WebSocket handshake
has not completed and that the semantics of HTTP still apply. The
headers follow the status code.
The |Connection| and |Upgrade| header fields complete the HTTP
Upgrade. The |Sec-WebSocket-Accept| header field indicates whether
the server is willing to accept the connection. If present, this
header field must include a hash of the client's nonce sent in
|Sec-WebSocket-Key| along with a predefined GUID. Any other value
must not be interpreted as an acceptance of the connection by the
server.
HTTP/1.1 101 Switching Protocols
Upgrade: websocket
Connection: Upgrade
Sec-WebSocket-Accept: s3pPLMBiTxaQ9kYGzzhZRbK+xOo=
These fields are checked by the WebSocket client for scripted pages.
If the |Sec-WebSocket-Accept| value does not match the expected
value, if the header field is missing, or if the HTTP status code is
not 101, the connection will not be established, and WebSocket frames
will not be sent.
Option fields can also be included. In this version of the protocol,
the main option field is |Sec-WebSocket-Protocol|, which indicates
the subprotocol that the server has selected. WebSocket clients
verify that the server included one of the values that was specified
in the WebSocket client's handshake. A server that speaks multiple
subprotocols has to make sure it selects one based on the client's
handshake and specifies it in its handshake.
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<span class="grey"><a href="https://tools.ietf.org/html/rfc6455">RFC 6455</a> The WebSocket Protocol December 2011</span>
Sec-WebSocket-Protocol: chat
The server can also set cookie-related option fields to _set_
cookies, as described in [<a href="https://tools.ietf.org/html/rfc6265" title=""HTTP State Management Mechanism"">RFC6265</a>].
<span class="h3"><h3><a class="selflink" name="section-1.4" href="#section-1.4">1.4</a>. Closing Handshake</h3></span>
_This section is non-normative._
The closing handshake is far simpler than the opening handshake.
Either peer can send a control frame with data containing a specified
control sequence to begin the closing handshake (detailed in
<a href="#section-5.5.1">Section 5.5.1</a>). Upon receiving such a frame, the other peer sends a
Close frame in response, if it hasn't already sent one. Upon
receiving _that_ control frame, the first peer then closes the
connection, safe in the knowledge that no further data is
forthcoming.
After sending a control frame indicating the connection should be
closed, a peer does not send any further data; after receiving a
control frame indicating the connection should be closed, a peer
discards any further data received.
It is safe for both peers to initiate this handshake simultaneously.
The closing handshake is intended to complement the TCP closing
handshake (FIN/ACK), on the basis that the TCP closing handshake is
not always reliable end-to-end, especially in the presence of
intercepting proxies and other intermediaries.
By sending a Close frame and waiting for a Close frame in response,
certain cases are avoided where data may be unnecessarily lost. For
instance, on some platforms, if a socket is closed with data in the
receive queue, a RST packet is sent, which will then cause recv() to
fail for the party that received the RST, even if there was data
waiting to be read.
<span class="h3"><h3><a class="selflink" name="section-1.5" href="#section-1.5">1.5</a>. Design Philosophy</h3></span>
_This section is non-normative._
The WebSocket Protocol is designed on the principle that there should
be minimal framing (the only framing that exists is to make the
protocol frame-based instead of stream-based and to support a
distinction between Unicode text and binary frames). It is expected
that metadata would be layered on top of WebSocket by the application
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layer, in the same way that metadata is layered on top of TCP by the
application layer (e.g., HTTP).
Conceptually, WebSocket is really just a layer on top of TCP that
does the following:
o adds a web origin-based security model for browsers
o adds an addressing and protocol naming mechanism to support
multiple services on one port and multiple host names on one IP
address
o layers a framing mechanism on top of TCP to get back to the IP
packet mechanism that TCP is built on, but without length limits
o includes an additional closing handshake in-band that is designed
to work in the presence of proxies and other intermediaries
Other than that, WebSocket adds nothing. Basically it is intended to
be as close to just exposing raw TCP to script as possible given the
constraints of the Web. It's also designed in such a way that its
servers can share a port with HTTP servers, by having its handshake
be a valid HTTP Upgrade request. One could conceptually use other
protocols to establish client-server messaging, but the intent of
WebSockets is to provide a relatively simple protocol that can
coexist with HTTP and deployed HTTP infrastructure (such as proxies)
and that is as close to TCP as is safe for use with such
infrastructure given security considerations, with targeted additions
to simplify usage and keep simple things simple (such as the addition
of message semantics).
The protocol is intended to be extensible; future versions will
likely introduce additional concepts such as multiplexing.
<span class="h3"><h3><a class="selflink" name="section-1.6" href="#section-1.6">1.6</a>. Security Model</h3></span>
_This section is non-normative._
The WebSocket Protocol uses the origin model used by web browsers to
restrict which web pages can contact a WebSocket server when the
WebSocket Protocol is used from a web page. Naturally, when the
WebSocket Protocol is used by a dedicated client directly (i.e., not
from a web page through a web browser), the origin model is not
useful, as the client can provide any arbitrary origin string.
This protocol is intended to fail to establish a connection with
servers of pre-existing protocols like SMTP [<a href="https://tools.ietf.org/html/rfc5321" title=""Simple Mail Transfer Protocol"">RFC5321</a>] and HTTP, while
allowing HTTP servers to opt-in to supporting this protocol if
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desired. This is achieved by having a strict and elaborate handshake
and by limiting the data that can be inserted into the connection
before the handshake is finished (thus limiting how much the server
can be influenced).
It is similarly intended to fail to establish a connection when data
from other protocols, especially HTTP, is sent to a WebSocket server,
for example, as might happen if an HTML "form" were submitted to a
WebSocket server. This is primarily achieved by requiring that the
server prove that it read the handshake, which it can only do if the
handshake contains the appropriate parts, which can only be sent by a
WebSocket client. In particular, at the time of writing of this
specification, fields starting with |Sec-| cannot be set by an
attacker from a web browser using only HTML and JavaScript APIs such
as XMLHttpRequest [<a href="#ref-XMLHttpRequest">XMLHttpRequest</a>].
<span class="h3"><h3><a class="selflink" name="section-1.7" href="#section-1.7">1.7</a>. Relationship to TCP and HTTP</h3></span>
_This section is non-normative._
The WebSocket Protocol is an independent TCP-based protocol. Its
only relationship to HTTP is that its handshake is interpreted by
HTTP servers as an Upgrade request.
By default, the WebSocket Protocol uses port 80 for regular WebSocket
connections and port 443 for WebSocket connections tunneled over
Transport Layer Security (TLS) [<a href="https://tools.ietf.org/html/rfc2818" title=""HTTP Over TLS"">RFC2818</a>].
<span class="h3"><h3><a class="selflink" name="section-1.8" href="#section-1.8">1.8</a>. Establishing a Connection</h3></span>
_This section is non-normative._
When a connection is to be made to a port that is shared by an HTTP
server (a situation that is quite likely to occur with traffic to
ports 80 and 443), the connection will appear to the HTTP server to
be a regular GET request with an Upgrade offer. In relatively simple
setups with just one IP address and a single server for all traffic
to a single hostname, this might allow a practical way for systems
based on the WebSocket Protocol to be deployed. In more elaborate
setups (e.g., with load balancers and multiple servers), a dedicated
set of hosts for WebSocket connections separate from the HTTP servers
is probably easier to manage. At the time of writing of this
specification, it should be noted that connections on ports 80 and
443 have significantly different success rates, with connections on
port 443 being significantly more likely to succeed, though this may
change with time.
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<span class="h3"><h3><a class="selflink" name="section-1.9" href="#section-1.9">1.9</a>. Subprotocols Using the WebSocket Protocol</h3></span>
_This section is non-normative._
The client can request that the server use a specific subprotocol by
including the |Sec-WebSocket-Protocol| field in its handshake. If it
is specified, the server needs to include the same field and one of
the selected subprotocol values in its response for the connection to
be established.
These subprotocol names should be registered as per <a href="#section-11.5">Section 11.5</a>. To
avoid potential collisions, it is recommended to use names that
contain the ASCII version of the domain name of the subprotocol's
originator. For example, if Example Corporation were to create a
Chat subprotocol to be implemented by many servers around the Web,
they could name it "chat.example.com". If the Example Organization
called their competing subprotocol "chat.example.org", then the two
subprotocols could be implemented by servers simultaneously, with the
server dynamically selecting which subprotocol to use based on the
value sent by the client.
Subprotocols can be versioned in backward-incompatible ways by
changing the subprotocol name, e.g., going from
"bookings.example.net" to "v2.bookings.example.net". These
subprotocols would be considered completely separate by WebSocket
clients. Backward-compatible versioning can be implemented by
reusing the same subprotocol string but carefully designing the
actual subprotocol to support this kind of extensibility.
<span class="h2"><h2><a class="selflink" name="section-2" href="#section-2">2</a>. Conformance Requirements</h2></span>
All diagrams, examples, and notes in this specification are non-
normative, as are all sections explicitly marked non-normative.
Everything else in this specification is normative.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [<a href="https://tools.ietf.org/html/rfc2119" title=""Key words for use in RFCs to Indicate Requirement Levels"">RFC2119</a>].
Requirements phrased in the imperative as part of algorithms (such as
"strip any leading space characters" or "return false and abort these
steps") are to be interpreted with the meaning of the key word
("MUST", "SHOULD", "MAY", etc.) used in introducing the algorithm.
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Conformance requirements phrased as algorithms or specific steps MAY
be implemented in any manner, so long as the end result is
equivalent. (In particular, the algorithms defined in this
specification are intended to be easy to follow and not intended to
be performant.)
<span class="h3"><h3><a class="selflink" name="section-2.1" href="#section-2.1">2.1</a>. Terminology and Other Conventions</h3></span>
_ASCII_ shall mean the character-encoding scheme defined in
[<a href="#ref-ANSI.X3-4.1986">ANSI.X3-4.1986</a>].
This document makes reference to UTF-8 values and uses UTF-8
notational formats as defined in STD 63 [<a href="https://tools.ietf.org/html/rfc3629" title=""UTF-8, a transformation format of ISO 10646"">RFC3629</a>].
Key terms such as named algorithms or definitions are indicated like
_this_.
Names of header fields or variables are indicated like |this|.
Variable values are indicated like /this/.
This document references the procedure to _Fail the WebSocket
Connection_. This procedure is defined in <a href="#section-7.1.7">Section 7.1.7</a>.
_Converting a string to ASCII lowercase_ means replacing all
characters in the range U+0041 to U+005A (i.e., LATIN CAPITAL LETTER
A to LATIN CAPITAL LETTER Z) with the corresponding characters in the
range U+0061 to U+007A (i.e., LATIN SMALL LETTER A to LATIN SMALL
LETTER Z).
Comparing two strings in an _ASCII case-insensitive_ manner means
comparing them exactly, code point for code point, except that the
characters in the range U+0041 to U+005A (i.e., LATIN CAPITAL LETTER
A to LATIN CAPITAL LETTER Z) and the corresponding characters in the
range U+0061 to U+007A (i.e., LATIN SMALL LETTER A to LATIN SMALL
LETTER Z) are considered to also match.
The term "URI" is used in this document as defined in [<a href="https://tools.ietf.org/html/rfc3986" title=""Uniform Resource Identifier (URI): Generic Syntax"">RFC3986</a>].
When an implementation is required to _send_ data as part of the
WebSocket Protocol, the implementation MAY delay the actual
transmission arbitrarily, e.g., buffering data so as to send fewer IP
packets.
Note that this document uses both [<a href="https://tools.ietf.org/html/rfc5234" title=""Augmented BNF for Syntax Specifications: ABNF"">RFC5234</a>] and [<a href="https://tools.ietf.org/html/rfc2616" title=""Hypertext Transfer Protocol -- HTTP/1.1"">RFC2616</a>] variants of
ABNF in different sections.
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<span class="h2"><h2><a class="selflink" name="section-3" href="#section-3">3</a>. WebSocket URIs</h2></span>
This specification defines two URI schemes, using the ABNF syntax
defined in <a href="https://tools.ietf.org/html/rfc5234">RFC 5234</a> [<a href="https://tools.ietf.org/html/rfc5234" title=""Augmented BNF for Syntax Specifications: ABNF"">RFC5234</a>], and terminology and ABNF productions
defined by the URI specification <a href="https://tools.ietf.org/html/rfc3986">RFC 3986</a> [<a href="https://tools.ietf.org/html/rfc3986" title=""Uniform Resource Identifier (URI): Generic Syntax"">RFC3986</a>].
ws-URI = "ws:" "//" host [ ":" port ] path [ "?" query ]
wss-URI = "wss:" "//" host [ ":" port ] path [ "?" query ]
host = <host, defined in <a href="https://tools.ietf.org/html/rfc3986#section-3.2.2">[RFC3986], Section 3.2.2</a>>
port = <port, defined in <a href="https://tools.ietf.org/html/rfc3986#section-3.2.3">[RFC3986], Section 3.2.3</a>>
path = <path-abempty, defined in <a href="https://tools.ietf.org/html/rfc3986#section-3.3">[RFC3986], Section 3.3</a>>
query = <query, defined in <a href="https://tools.ietf.org/html/rfc3986#section-3.4">[RFC3986], Section 3.4</a>>
The port component is OPTIONAL; the default for "ws" is port 80,
while the default for "wss" is port 443.
The URI is called "secure" (and it is said that "the secure flag is
set") if the scheme component matches "wss" case-insensitively.
The "resource-name" (also known as /resource name/ in <a href="#section-4.1">Section 4.1</a>)
can be constructed by concatenating the following:
o "/" if the path component is empty
o the path component
o "?" if the query component is non-empty
o the query component
Fragment identifiers are meaningless in the context of WebSocket URIs
and MUST NOT be used on these URIs. As with any URI scheme, the
character "#", when not indicating the start of a fragment, MUST be
escaped as %23.
<span class="h2"><h2><a class="selflink" name="section-4" href="#section-4">4</a>. Opening Handshake</h2></span>
<span class="h3"><h3><a class="selflink" name="section-4.1" href="#section-4.1">4.1</a>. Client Requirements</h3></span>
To _Establish a WebSocket Connection_, a client opens a connection
and sends a handshake as defined in this section. A connection is
defined to initially be in a CONNECTING state. A client will need to
supply a /host/, /port/, /resource name/, and a /secure/ flag, which
are the components of a WebSocket URI as discussed in <a href="#section-3">Section 3</a>,
along with a list of /protocols/ and /extensions/ to be used.
Additionally, if the client is a web browser, it supplies /origin/.
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Clients running in controlled environments, e.g., browsers on mobile
handsets tied to specific carriers, MAY offload the management of the
connection to another agent on the network. In such a situation, the
client for the purposes of this specification is considered to
include both the handset software and any such agents.
When the client is to _Establish a WebSocket Connection_ given a set
of (/host/, /port/, /resource name/, and /secure/ flag), along with a
list of /protocols/ and /extensions/ to be used, and an /origin/ in
the case of web browsers, it MUST open a connection, send an opening
handshake, and read the server's handshake in response. The exact
requirements of how the connection should be opened, what should be
sent in the opening handshake, and how the server's response should
be interpreted are as follows in this section. In the following
text, we will use terms from <a href="#section-3">Section 3</a>, such as "/host/" and
"/secure/ flag" as defined in that section.
1. The components of the WebSocket URI passed into this algorithm
(/host/, /port/, /resource name/, and /secure/ flag) MUST be
valid according to the specification of WebSocket URIs specified
in <a href="#section-3">Section 3</a>. If any of the components are invalid, the client
MUST _Fail the WebSocket Connection_ and abort these steps.
2. If the client already has a WebSocket connection to the remote
host (IP address) identified by /host/ and port /port/ pair, even
if the remote host is known by another name, the client MUST wait
until that connection has been established or for that connection
to have failed. There MUST be no more than one connection in a
CONNECTING state. If multiple connections to the same IP address
are attempted simultaneously, the client MUST serialize them so
that there is no more than one connection at a time running
through the following steps.