weixin_34221276

The WebSocket Protocol

[Docs] [txt|pdf] [draft-ietf-hybi-t...] [Diff1] [Diff2] [Errata]

Updated by: 7936 PROPOSED STANDARD
Errata Exist

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

The WebSocket Protocol



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 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 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/rfc6455.

Copyright Notice

   Copyright (c) 2011 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must




<span class="grey">Fette & Melnikov             Standards Track                    [Page 1]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
     1.1.  Background . . . . . . . . . . . . . . . . . . . . . . . .  4
     1.2.  Protocol Overview  . . . . . . . . . . . . . . . . . . . .  5
     1.3.  Opening Handshake  . . . . . . . . . . . . . . . . . . . .  6
     1.4.  Closing Handshake  . . . . . . . . . . . . . . . . . . . .  9
     1.5.  Design Philosophy  . . . . . . . . . . . . . . . . . . . .  9
     1.6.  Security Model . . . . . . . . . . . . . . . . . . . . . . 10
     1.7.  Relationship to TCP and HTTP . . . . . . . . . . . . . . . 11
     1.8.  Establishing a Connection  . . . . . . . . . . . . . . . . 11
     1.9.  Subprotocols Using the WebSocket Protocol  . . . . . . . . 12
   2.  Conformance Requirements . . . . . . . . . . . . . . . . . . . 12
     2.1.  Terminology and Other Conventions  . . . . . . . . . . . . 13
   3.  WebSocket URIs . . . . . . . . . . . . . . . . . . . . . . . . 14
   4.  Opening Handshake  . . . . . . . . . . . . . . . . . . . . . . 14
     4.1.  Client Requirements  . . . . . . . . . . . . . . . . . . . 14
     4.2.  Server-Side Requirements . . . . . . . . . . . . . . . . . 20
       4.2.1.  Reading the Client's Opening Handshake . . . . . . . . 21
       4.2.2.  Sending the Server's Opening Handshake . . . . . . . . 22
     4.3.  Collected ABNF for New Header Fields Used in Handshake . . 25
     4.4.  Supporting Multiple Versions of WebSocket Protocol . . . . 26
   5.  Data Framing . . . . . . . . . . . . . . . . . . . . . . . . . 27
     5.1.  Overview . . . . . . . . . . . . . . . . . . . . . . . . . 27
     5.2.  Base Framing Protocol  . . . . . . . . . . . . . . . . . . 28
     5.3.  Client-to-Server Masking . . . . . . . . . . . . . . . . . 32
     5.4.  Fragmentation  . . . . . . . . . . . . . . . . . . . . . . 33
     5.5.  Control Frames . . . . . . . . . . . . . . . . . . . . . . 36
       5.5.1.  Close  . . . . . . . . . . . . . . . . . . . . . . . . 36
       5.5.2.  Ping . . . . . . . . . . . . . . . . . . . . . . . . . 37
       5.5.3.  Pong . . . . . . . . . . . . . . . . . . . . . . . . . 37
     5.6.  Data Frames  . . . . . . . . . . . . . . . . . . . . . . . 38
     5.7.  Examples . . . . . . . . . . . . . . . . . . . . . . . . . 38
     5.8.  Extensibility  . . . . . . . . . . . . . . . . . . . . . . 39
   6.  Sending and Receiving Data . . . . . . . . . . . . . . . . . . 39
     6.1.  Sending Data . . . . . . . . . . . . . . . . . . . . . . . 39
     6.2.  Receiving Data . . . . . . . . . . . . . . . . . . . . . . 40
   7.  Closing the Connection . . . . . . . . . . . . . . . . . . . . 41
     7.1.  Definitions  . . . . . . . . . . . . . . . . . . . . . . . 41
       7.1.1.  Close the WebSocket Connection . . . . . . . . . . . . 41
       7.1.2.  Start the WebSocket Closing Handshake  . . . . . . . . 42
       7.1.3.  The WebSocket Closing Handshake is Started . . . . . . 42
       7.1.4.  The WebSocket Connection is Closed . . . . . . . . . . 42
       7.1.5.  The WebSocket Connection Close Code  . . . . . . . . . 42



<span class="grey">Fette & Melnikov             Standards Track                    [Page 2]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


       7.1.6.  The WebSocket Connection Close Reason  . . . . . . . . 43
       7.1.7.  Fail the WebSocket Connection  . . . . . . . . . . . . 43
     7.2.  Abnormal Closures  . . . . . . . . . . . . . . . . . . . . 44
       7.2.1.  Client-Initiated Closure . . . . . . . . . . . . . . . 44
       7.2.2.  Server-Initiated Closure . . . . . . . . . . . . . . . 44
       7.2.3.  Recovering from Abnormal Closure . . . . . . . . . . . 44
     7.3.  Normal Closure of Connections  . . . . . . . . . . . . . . 45
     7.4.  Status Codes . . . . . . . . . . . . . . . . . . . . . . . 45
       7.4.1.  Defined Status Codes . . . . . . . . . . . . . . . . . 45
       7.4.2.  Reserved Status Code Ranges  . . . . . . . . . . . . . 47
   8.  Error Handling . . . . . . . . . . . . . . . . . . . . . . . . 48
     8.1.  Handling Errors in UTF-8-Encoded Data  . . . . . . . . . . 48
   9.  Extensions . . . . . . . . . . . . . . . . . . . . . . . . . . 48
     9.1.  Negotiating Extensions . . . . . . . . . . . . . . . . . . 48
     9.2.  Known Extensions . . . . . . . . . . . . . . . . . . . . . 50
   10. Security Considerations  . . . . . . . . . . . . . . . . . . . 50
     10.1. Non-Browser Clients  . . . . . . . . . . . . . . . . . . . 50
     10.2. Origin Considerations  . . . . . . . . . . . . . . . . . . 50
     10.3. Attacks On Infrastructure (Masking)  . . . . . . . . . . . 51
     10.4. Implementation-Specific Limits . . . . . . . . . . . . . . 52
     10.5. WebSocket Client Authentication  . . . . . . . . . . . . . 53
     10.6. Connection Confidentiality and Integrity . . . . . . . . . 53
     10.7. Handling of Invalid Data . . . . . . . . . . . . . . . . . 53
     10.8. Use of SHA-1 by the WebSocket Handshake  . . . . . . . . . 54
   11. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 54
     11.1. Registration of New URI Schemes  . . . . . . . . . . . . . 54
       11.1.1. Registration of "ws" Scheme  . . . . . . . . . . . . . 54
       11.1.2. Registration of "wss" Scheme . . . . . . . . . . . . . 55
     11.2. Registration of the "WebSocket" HTTP Upgrade Keyword . . . 56
     11.3. Registration of New HTTP Header Fields . . . . . . . . . . 57
       11.3.1. Sec-WebSocket-Key  . . . . . . . . . . . . . . . . . . 57
       11.3.2. Sec-WebSocket-Extensions . . . . . . . . . . . . . . . 58
       11.3.3. Sec-WebSocket-Accept . . . . . . . . . . . . . . . . . 58
       11.3.4. Sec-WebSocket-Protocol . . . . . . . . . . . . . . . . 59
       11.3.5. Sec-WebSocket-Version  . . . . . . . . . . . . . . . . 60
     11.4. WebSocket Extension Name Registry  . . . . . . . . . . . . 61
     11.5. WebSocket Subprotocol Name Registry  . . . . . . . . . . . 61
     11.6. WebSocket Version Number Registry  . . . . . . . . . . . . 62
     11.7. WebSocket Close Code Number Registry . . . . . . . . . . . 64
     11.8. WebSocket Opcode Registry  . . . . . . . . . . . . . . . . 65
     11.9. WebSocket Framing Header Bits Registry . . . . . . . . . . 66
   12. Using the WebSocket Protocol from Other Specifications . . . . 66
   13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 67
   14. References . . . . . . . . . . . . . . . . . . . . . . . . . . 68
     14.1. Normative References . . . . . . . . . . . . . . . . . . . 68
     14.2. Informative References . . . . . . . . . . . . . . . . . . 69





<span class="grey">Fette & Melnikov             Standards Track                    [Page 3]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


</span></pre> 
   <h2>1. Introduction</h2> 
   <h3>1.1. Background</h3> 
   <pre class="newpage"><span class="grey"><span class="h2"><span class="h3">

   _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 [RFC6202].

   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 [WSAPI], 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 [RFC6202] 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



<span class="grey">Fette & Melnikov             Standards Track                    [Page 4]</span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   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></pre> 
   <h3>1.2. Protocol Overview</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   _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 [RFC2616].

   An unordered set of header fields comes after the leading line in
   both cases.  The meaning of these header fields is specified in
   Section 4 of this document.  Additional header fields may also be
   present, such as cookies [RFC6265].  The format and parsing of
   headers is as defined in [RFC2616].

   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



<span class="grey">Fette & Melnikov             Standards Track                    [Page 5]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   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 [RFC3629]
   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></pre> 
   <h3>1.3. Opening Handshake</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   _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 [RFC2616], 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 [RFC2616] 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 [RFC2616], so that both the client and the server
   can verify that they agree on which host is in use.






<span class="grey">Fette & Melnikov             Standards Track                    [Page 6]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   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 [RFC6454] 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 [XMLHttpRequest] 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 [RFC4648] version minus
   any leading and trailing whitespace) and concatenate this with the
   Globally Unique Identifier (GUID, [RFC4122]) "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) [FIPS.180-3], base64-encoded (see Section 4 of
   [RFC4648]), of this concatenation is then returned in the server's
   handshake.





<span class="grey">Fette & Melnikov             Standards Track                    [Page 7]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   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 Section 4 of [RFC4648]), 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.




<span class="grey">Fette & Melnikov             Standards Track                    [Page 8]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


        Sec-WebSocket-Protocol: chat

   The server can also set cookie-related option fields to _set_
   cookies, as described in [RFC6265].

</span></pre> 
   <h3>1.4. Closing Handshake</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   _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
   Section 5.5.1).  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></span></pre> 
   <h3>1.5. Design Philosophy</h3> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h3">

   _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




<span class="grey">Fette & Melnikov             Standards Track                    [Page 9]</span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   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></pre> 
   <h3>1.6. Security Model</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   _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 [RFC5321] and HTTP, while
   allowing HTTP servers to opt-in to supporting this protocol if



<span class="grey">Fette & Melnikov             Standards Track                   [Page 10]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   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 [XMLHttpRequest].

</span></pre> 
   <h3>1.7. Relationship to TCP and HTTP</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   _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) [RFC2818].

</span></span></pre> 
   <h3>1.8. Establishing a Connection</h3> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h3">

   _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.





<span class="grey">Fette & Melnikov             Standards Track                   [Page 11]</span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


</span></pre> 
   <h3>1.9. Subprotocols Using the WebSocket Protocol</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   _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 Section 11.5.  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></span></pre> 
   <h2>2. Conformance Requirements</h2> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h2">

   All diagrams, examples, and notes in this specification are non-
   normative, as are all sections explicitly marked non-normative.
   Everything else in this specification is normative.

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

   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.








<span class="grey">Fette & Melnikov             Standards Track                   [Page 12]</span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   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></pre> 
   <h3>2.1. Terminology and Other Conventions</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   _ASCII_ shall mean the character-encoding scheme defined in
   [ANSI.X3-4.1986].

   This document makes reference to UTF-8 values and uses UTF-8
   notational formats as defined in STD 63 [RFC3629].

   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 Section 7.1.7.

   _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 [RFC3986].

   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 [RFC5234] and [RFC2616] variants of
   ABNF in different sections.





<span class="grey">Fette & Melnikov             Standards Track                   [Page 13]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


</span></pre> 
   <h2>3. WebSocket URIs</h2> 
   <pre class="newpage"><span class="grey"><span class="h2">

   This specification defines two URI schemes, using the ABNF syntax
   defined in RFC 5234 [RFC5234], and terminology and ABNF productions
   defined by the URI specification RFC 3986 [RFC3986].

          ws-URI = "ws:" "//" host [ ":" port ] path [ "?" query ]
          wss-URI = "wss:" "//" host [ ":" port ] path [ "?" query ]

          host = <host, defined in [RFC3986], Section 3.2.2>
          port = <port, defined in [RFC3986], Section 3.2.3>
          path = <path-abempty, defined in [RFC3986], Section 3.3>
          query = <query, defined in [RFC3986], Section 3.4>

   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 Section 4.1)
   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></span></pre> 
   <h2>4. Opening Handshake</h2> 
   <h3>4.1. Client Requirements</h3> 
   <pre class="newpage"><span class="grey"><span class="h2"><span class="h2"><span class="h3">

   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 Section 3,
   along with a list of /protocols/ and /extensions/ to be used.
   Additionally, if the client is a web browser, it supplies /origin/.




<span class="grey">Fette & Melnikov             Standards Track                   [Page 14]</span></span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   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 Section 3, 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 Section 3.  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.

       If the client cannot determine the IP address of the remote host
       (for example, because all communication is being done through a
       proxy server that performs DNS queries itself), then the client
       MUST assume for the purposes of this step that each host name
       refers to a distinct remote host, and instead the client SHOULD
       limit the total number of simultaneous pending connections to a
       reasonably low number (e.g., the client might allow simultaneous
       pending connections to a.example.com and b.example.com, but if
       thirty simultaneous connections to a single host are requested,
       that may not be allowed).  For example, in a web browser context,
       the client needs to consider the number of tabs the user has open
       in setting a limit to the number of simultaneous pending
       connections.





<span class="grey">Fette & Melnikov             Standards Track                   [Page 15]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


       NOTE: This makes it harder for a script to perform a denial-of-
       service attack by just opening a large number of WebSocket
       connections to a remote host.  A server can further reduce the
       load on itself when attacked by pausing before closing the
       connection, as that will reduce the rate at which the client
       reconnects.

       NOTE: There is no limit to the number of established WebSocket
       connections a client can have with a single remote host.  Servers
       can refuse to accept connections from hosts/IP addresses with an
       excessive number of existing connections or disconnect resource-
       hogging connections when suffering high load.

   3.  _Proxy Usage_: If the client is configured to use a proxy when
       using the WebSocket Protocol to connect to host /host/ and port
       /port/, then the client SHOULD connect to that proxy and ask it
       to open a TCP connection to the host given by /host/ and the port
       given by /port/.

          EXAMPLE: For example, if the client uses an HTTP proxy for all
          traffic, then if it was to try to connect to port 80 on server
          example.com, it might send the following lines to the proxy
          server:

              CONNECT example.com:80 HTTP/1.1
              Host: example.com

          If there was a password, the connection might look like:

              CONNECT example.com:80 HTTP/1.1
              Host: example.com
              Proxy-authorization: Basic ZWRuYW1vZGU6bm9jYXBlcyE=

       If the client is not configured to use a proxy, then a direct TCP
       connection SHOULD be opened to the host given by /host/ and the
       port given by /port/.

       NOTE: Implementations that do not expose explicit UI for
       selecting a proxy for WebSocket connections separate from other
       proxies are encouraged to use a SOCKS5 [RFC1928] proxy for
       WebSocket connections, if available, or failing that, to prefer
       the proxy configured for HTTPS connections over the proxy
       configured for HTTP connections.

       For the purpose of proxy autoconfiguration scripts, the URI to
       pass the function MUST be constructed from /host/, /port/,
       /resource name/, and the /secure/ flag using the definition of a
       WebSocket URI as given in Section 3.



<span class="grey">Fette & Melnikov             Standards Track                   [Page 16]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


       NOTE: The WebSocket Protocol can be identified in proxy
       autoconfiguration scripts from the scheme ("ws" for unencrypted
       connections and "wss" for encrypted connections).

   4.  If the connection could not be opened, either because a direct
       connection failed or because any proxy used returned an error,
       then the client MUST _Fail the WebSocket Connection_ and abort
       the connection attempt.

   5.  If /secure/ is true, the client MUST perform a TLS handshake over
       the connection after opening the connection and before sending
       the handshake data [RFC2818].  If this fails (e.g., the server's
       certificate could not be verified), then the client MUST _Fail
       the WebSocket Connection_ and abort the connection.  Otherwise,
       all further communication on this channel MUST run through the
       encrypted tunnel [RFC5246].

       Clients MUST use the Server Name Indication extension in the TLS
       handshake [RFC6066].

   Once a connection to the server has been established (including a
   connection via a proxy or over a TLS-encrypted tunnel), the client
   MUST send an opening handshake to the server.  The handshake consists
   of an HTTP Upgrade request, along with a list of required and
   optional header fields.  The requirements for this handshake are as
   follows.

   1.   The handshake MUST be a valid HTTP request as specified by
        [RFC2616].

   2.   The method of the request MUST be GET, and the HTTP version MUST
        be at least 1.1.

        For example, if the WebSocket URI is "ws://example.com/chat",
        the first line sent should be "GET /chat HTTP/1.1".

   3.   The "Request-URI" part of the request MUST match the /resource
        name/ defined in Section 3 (a relative URI) or be an absolute
        http/https URI that, when parsed, has a /resource name/, /host/,
        and /port/ that match the corresponding ws/wss URI.

   4.   The request MUST contain a |Host| header field whose value
        contains /host/ plus optionally ":" followed by /port/ (when not
        using the default port).

   5.   The request MUST contain an |Upgrade| header field whose value
        MUST include the "websocket" keyword.




<span class="grey">Fette & Melnikov             Standards Track                   [Page 17]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   6.   The request MUST contain a |Connection| header field whose value
        MUST include the "Upgrade" token.

   7.   The request MUST include a header field with the name
        |Sec-WebSocket-Key|.  The value of this header field MUST be a
        nonce consisting of a randomly selected 16-byte value that has
        been base64-encoded (see Section 4 of [RFC4648]).  The nonce
        MUST be selected randomly for each connection.

        NOTE: As an example, if the randomly selected value was the
        sequence of bytes 0x01 0x02 0x03 0x04 0x05 0x06 0x07 0x08 0x09
        0x0a 0x0b 0x0c 0x0d 0x0e 0x0f 0x10, the value of the header
        field would be "AQIDBAUGBwgJCgsMDQ4PEC=="

   8.   The request MUST include a header field with the name |Origin|
        [RFC6454] if the request is coming from a browser client.  If
        the connection is from a non-browser client, the request MAY
        include this header field if the semantics of that client match
        the use-case described here for browser clients.  The value of
        this header field is the ASCII serialization of origin of the
        context in which the code establishing the connection is
        running.  See [RFC6454] for the details of how this header field
        value is constructed.

        As an example, if code downloaded from www.example.com attempts
        to establish a connection to ww2.example.com, the value of the
        header field would be "http://www.example.com".

   9.   The request MUST include a header field with the name
        |Sec-WebSocket-Version|.  The value of this header field MUST be
        13.

        NOTE: Although draft versions of this document (-09, -10, -11,
        and -12) were posted (they were mostly comprised of editorial
        changes and clarifications and not changes to the wire
        protocol), values 9, 10, 11, and 12 were not used as valid
        values for Sec-WebSocket-Version.  These values were reserved in
        the IANA registry but were not and will not be used.

   10.  The request MAY include a header field with the name
        |Sec-WebSocket-Protocol|.  If present, this value indicates one
        or more comma-separated subprotocol the client wishes to speak,
        ordered by preference.  The elements that comprise this value
        MUST be non-empty strings with characters in the range U+0021 to
        U+007E not including separator characters as defined in
        [RFC2616] and MUST all be unique strings.  The ABNF for the
        value of this header field is 1#token, where the definitions of
        constructs and rules are as given in [RFC2616].



<span class="grey">Fette & Melnikov             Standards Track                   [Page 18]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   11.  The request MAY include a header field with the name
        |Sec-WebSocket-Extensions|.  If present, this value indicates
        the protocol-level extension(s) the client wishes to speak.  The
        interpretation and format of this header field is described in
        Section 9.1.

   12.  The request MAY include any other header fields, for example,
        cookies [RFC6265] and/or authentication-related header fields
        such as the |Authorization| header field [RFC2616], which are
        processed according to documents that define them.

   Once the client's opening handshake has been sent, the client MUST
   wait for a response from the server before sending any further data.
   The client MUST validate the server's response as follows:

   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 a 401 status code; the server might redirect the client
       using a 3xx status code (but clients are not required to follow
       them), etc.  Otherwise, proceed as follows.

   2.  If the response lacks an |Upgrade| header field or the |Upgrade|
       header field contains a value that is not an ASCII case-
       insensitive match for the value "websocket", the client MUST
       _Fail the WebSocket Connection_.

   3.  If the response lacks a |Connection| header field or the
       |Connection| header field doesn't contain a token that is an
       ASCII case-insensitive match for the value "Upgrade", the client
       MUST _Fail the WebSocket Connection_.

   4.  If the response lacks a |Sec-WebSocket-Accept| header field or
       the |Sec-WebSocket-Accept| contains a value other than the
       base64-encoded SHA-1 of the concatenation of the |Sec-WebSocket-
       Key| (as a string, not base64-decoded) with the string "258EAFA5-
       E914-47DA-95CA-C5AB0DC85B11" but ignoring any leading and
       trailing whitespace, the client MUST _Fail the WebSocket
       Connection_.

   5.  If the response includes a |Sec-WebSocket-Extensions| header
       field and this header field indicates the use of an extension
       that was not present in the client's handshake (the server has
       indicated an extension not requested by the client), the client
       MUST _Fail the WebSocket Connection_.  (The parsing of this
       header field to determine which extensions are requested is
       discussed in Section 9.1.)




<span class="grey">Fette & Melnikov             Standards Track                   [Page 19]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   6.  If the response includes a |Sec-WebSocket-Protocol| header field
       and this header field indicates the use of a subprotocol that was
       not present in the client's handshake (the server has indicated a
       subprotocol not requested by the client), the client MUST _Fail
       the WebSocket Connection_.

   If the server's response does not conform to the requirements for the
   server's handshake as defined in this section and in Section 4.2.2,
   the client MUST _Fail the WebSocket Connection_.

   Please note that according to [RFC2616], all header field names in
   both HTTP requests and HTTP responses are case-insensitive.

   If the server's response is validated as provided for above, it is
   said that _The WebSocket Connection is Established_ and that the
   WebSocket Connection is in the OPEN state.  The _Extensions In Use_
   is defined to be a (possibly empty) string, the value of which is
   equal to the value of the |Sec-WebSocket-Extensions| header field
   supplied by the server's handshake or the null value if that header
   field was not present in the server's handshake.  The _Subprotocol In
   Use_ is defined to be the value of the |Sec-WebSocket-Protocol|
   header field in the server's handshake or the null value if that
   header field was not present in the server's handshake.
   Additionally, if any header fields in the server's handshake indicate
   that cookies should be set (as defined by [RFC6265]), these cookies
   are referred to as _Cookies Set During the Server's Opening
   Handshake_.

</span></pre> 
   <h3>4.2. Server-Side Requirements</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   Servers MAY offload the management of the connection to other agents
   on the network, for example, load balancers and reverse proxies.  In
   such a situation, the server for the purposes of this specification
   is considered to include all parts of the server-side infrastructure
   from the first device to terminate the TCP connection all the way to
   the server that processes requests and sends responses.

   EXAMPLE: A data center might have a server that responds to WebSocket
   requests with an appropriate handshake and then passes the connection
   to another server to actually process the data frames.  For the
   purposes of this specification, the "server" is the combination of
   both computers.









<span class="grey">Fette & Melnikov             Standards Track                   [Page 20]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


</span></pre> 
   <h4>4.2.1. Reading the Client's Opening Handshake</h4> 
   <pre class="newpage"><span class="grey"><span class="h4">

   When a client starts a WebSocket connection, it sends its part of the
   opening handshake.  The server must parse at least part of this
   handshake in order to obtain the necessary information to generate
   the server part of the handshake.

   The client's opening handshake consists of the following parts.  If
   the server, while reading the handshake, finds that the client did
   not send a handshake that matches the description below (note that as
   per [RFC2616], the order of the header fields is not important),
   including but not limited to any violations of the ABNF grammar
   specified for the components of the handshake, the server MUST stop
   processing the client's handshake and return an HTTP response with an
   appropriate error code (such as 400 Bad Request).

   1.   An HTTP/1.1 or higher GET request, including a "Request-URI"
        [RFC2616] that should be interpreted as a /resource name/
        defined in Section 3 (or an absolute HTTP/HTTPS URI containing
        the /resource name/).

   2.   A |Host| header field containing the server's authority.

   3.   An |Upgrade| header field containing the value "websocket",
        treated as an ASCII case-insensitive value.

   4.   A |Connection| header field that includes the token "Upgrade",
        treated as an ASCII case-insensitive value.

   5.   A |Sec-WebSocket-Key| header field with a base64-encoded (see
        Section 4 of [RFC4648]) value that, when decoded, is 16 bytes in
        length.

   6.   A |Sec-WebSocket-Version| header field, with a value of 13.

   7.   Optionally, an |Origin| header field.  This header field is sent
        by all browser clients.  A connection attempt lacking this
        header field SHOULD NOT be interpreted as coming from a browser
        client.

   8.   Optionally, a |Sec-WebSocket-Protocol| header field, with a list
        of values indicating which protocols the client would like to
        speak, ordered by preference.

   9.   Optionally, a |Sec-WebSocket-Extensions| header field, with a
        list of values indicating which extensions the client would like
        to speak.  The interpretation of this header field is discussed
        in Section 9.1.



<span class="grey">Fette & Melnikov             Standards Track                   [Page 21]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   10.  Optionally, other header fields, such as those used to send
        cookies or request authentication to a server.  Unknown header
        fields are ignored, as per [RFC2616].

</span></pre> 
   <h4>4.2.2. Sending the Server's Opening Handshake</h4> 
   <pre class="newpage"><span class="grey"><span class="h4">

   When a client establishes a WebSocket connection to a server, the
   server MUST complete the following steps to accept the connection and
   send the server's opening handshake.

   1.  If the connection is happening on an HTTPS (HTTP-over-TLS) port,
       perform a TLS handshake over the connection.  If this fails
       (e.g., the client indicated a host name in the extended client
       hello "server_name" extension that the server does not host),
       then close the connection; otherwise, all further communication
       for the connection (including the server's handshake) MUST run
       through the encrypted tunnel [RFC5246].

   2.  The server can perform additional client authentication, for
       example, by returning a 401 status code with the corresponding
       |WWW-Authenticate| header field as described in [RFC2616].

   3.  The server MAY redirect the client using a 3xx status code
       [RFC2616].  Note that this step can happen together with, before,
       or after the optional authentication step described above.

   4.  Establish the following information:

       /origin/
          The |Origin| header field in the client's handshake indicates
          the origin of the script establishing the connection.  The
          origin is serialized to ASCII and converted to lowercase.  The
          server MAY use this information as part of a determination of
          whether to accept the incoming connection.  If the server does
          not validate the origin, it will accept connections from
          anywhere.  If the server does not wish to accept this
          connection, it MUST return an appropriate HTTP error code
          (e.g., 403 Forbidden) and abort the WebSocket handshake
          described in this section.  For more detail, refer to
          Section 10.

       /key/
          The |Sec-WebSocket-Key| header field in the client's handshake
          includes a base64-encoded value that, if decoded, is 16 bytes
          in length.  This (encoded) value is used in the creation of
          the server's handshake to indicate an acceptance of the
          connection.  It is not necessary for the server to base64-
          decode the |Sec-WebSocket-Key| value.



<span class="grey">Fette & Melnikov             Standards Track                   [Page 22]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


       /version/
          The |Sec-WebSocket-Version| header field in the client's
          handshake includes the version of the WebSocket Protocol with
          which the client is attempting to communicate.  If this
          version does not match a version understood by the server, the
          server MUST abort the WebSocket handshake described in this
          section and instead send an appropriate HTTP error code (such
          as 426 Upgrade Required) and a |Sec-WebSocket-Version| header
          field indicating the version(s) the server is capable of
          understanding.

       /resource name/
          An identifier for the service provided by the server.  If the
          server provides multiple services, then the value should be
          derived from the resource name given in the client's handshake
          in the "Request-URI" [RFC2616] of the GET method.  If the
          requested service is not available, the server MUST send an
          appropriate HTTP error code (such as 404 Not Found) and abort
          the WebSocket handshake.

       /subprotocol/
          Either a single value representing the subprotocol the server
          is ready to use or null.  The value chosen MUST be derived
          from the client's handshake, specifically by selecting one of
          the values from the |Sec-WebSocket-Protocol| field that the
          server is willing to use for this connection (if any).  If the
          client's handshake did not contain such a header field or if
          the server does not agree to any of the client's requested
          subprotocols, the only acceptable value is null.  The absence
          of such a field is equivalent to the null value (meaning that
          if the server does not wish to agree to one of the suggested
          subprotocols, it MUST NOT send back a |Sec-WebSocket-Protocol|
          header field in its response).  The empty string is not the
          same as the null value for these purposes and is not a legal
          value for this field.  The ABNF for the value of this header
          field is (token), where the definitions of constructs and
          rules are as given in [RFC2616].

       /extensions/
          A (possibly empty) list representing the protocol-level
          extensions the server is ready to use.  If the server supports
          multiple extensions, then the value MUST be derived from the
          client's handshake, specifically by selecting one or more of
          the values from the |Sec-WebSocket-Extensions| field.  The
          absence of such a field is equivalent to the null value.  The
          empty string is not the same as the null value for these





<span class="grey">Fette & Melnikov             Standards Track                   [Page 23]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


          purposes.  Extensions not listed by the client MUST NOT be
          listed.  The method by which these values should be selected
          and interpreted is discussed in Section 9.1.

   5.  If the server chooses to accept the incoming connection, it MUST
       reply with a valid HTTP response indicating the following.

       1.  A Status-Line with a 101 response code as per RFC 2616
           [RFC2616].  Such a response could look like "HTTP/1.1 101
           Switching Protocols".

       2.  An |Upgrade| header field with value "websocket" as per RFC
           2616 [RFC2616].

       3.  A |Connection| header field with value "Upgrade".

       4.  A |Sec-WebSocket-Accept| header field.  The value of this
           header field is constructed by concatenating /key/, defined
           above in step 4 in Section 4.2.2, with the string "258EAFA5-
           E914-47DA-95CA-C5AB0DC85B11", taking the SHA-1 hash of this
           concatenated value to obtain a 20-byte value and base64-
           encoding (see Section 4 of [RFC4648]) this 20-byte hash.

           The ABNF [RFC2616] of this header field is defined as
           follows:

           Sec-WebSocket-Accept     = base64-value-non-empty
           base64-value-non-empty = (1*base64-data [ base64-padding ]) |
                                    base64-padding
           base64-data      = 4base64-character
           base64-padding   = (2base64-character "==") |
                              (3base64-character "=")
           base64-character = ALPHA | DIGIT | "+" | "/"

   NOTE: As an example, if the value of the |Sec-WebSocket-Key| header
   field in the client's handshake were "dGhlIHNhbXBsZSBub25jZQ==", the
   server would append 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
   string, 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, to give the value
   "s3pPLMBiTxaQ9kYGzzhZRbK+xOo=", which would be returned in the
   |Sec-WebSocket-Accept| header field.

       5.  Optionally, a |Sec-WebSocket-Protocol| header field, with a
           value /subprotocol/ as defined in step 4 in Section 4.2.2.




<span class="grey">Fette & Melnikov             Standards Track                   [Page 24]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


       6.  Optionally, a |Sec-WebSocket-Extensions| header field, with a
           value /extensions/ as defined in step 4 in Section 4.2.2.  If
           multiple extensions are to be used, they can all be listed in
           a single |Sec-WebSocket-Extensions| header field or split
           between multiple instances of the |Sec-WebSocket-Extensions|
           header field.

   This completes the server's handshake.  If the server finishes these
   steps without aborting the WebSocket handshake, the server considers
   the WebSocket connection to be established and that the WebSocket
   connection is in the OPEN state.  At this point, the server may begin
   sending (and receiving) data.

</span></pre> 
   <h3>4.3. Collected ABNF for New Header Fields Used in Handshake</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   This section is using ABNF syntax/rules from Section 2.1 of
   [RFC2616], including the "implied *LWS rule".

   Note that the following ABNF conventions are used in this section.
   Some names of the rules correspond to names of the corresponding
   header fields.  Such rules express values of the corresponding header
   fields, for example, the Sec-WebSocket-Key ABNF rule describes syntax
   of the |Sec-WebSocket-Key| header field value.  ABNF rules with the
   "-Client" suffix in the name are only used in requests sent by the
   client to the server; ABNF rules with the "-Server" suffix in the
   name are only used in responses sent by the server to the client.
   For example, the ABNF rule Sec-WebSocket-Protocol-Client describes
   syntax of the |Sec-WebSocket-Protocol| header field value sent by the
   client to the server.

   The following new header fields can be sent during the handshake from
   the client to the server:

      Sec-WebSocket-Key = base64-value-non-empty
      Sec-WebSocket-Extensions = extension-list
      Sec-WebSocket-Protocol-Client = 1#token
      Sec-WebSocket-Version-Client = version

      base64-value-non-empty = (1*base64-data [ base64-padding ]) |
                                base64-padding
      base64-data      = 4base64-character
      base64-padding   = (2base64-character "==") |
                         (3base64-character "=")
      base64-character = ALPHA | DIGIT | "+" | "/"
      extension-list = 1#extension
      extension = extension-token *( ";" extension-param )
      extension-token = registered-token
      registered-token = token



<span class="grey">Fette & Melnikov             Standards Track                   [Page 25]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


      extension-param = token [ "=" (token | quoted-string) ]
           ; When using the quoted-string syntax variant, the value
           ; after quoted-string unescaping MUST conform to the
           ; 'token' ABNF.
      NZDIGIT       =  "1" | "2" | "3" | "4" | "5" | "6" |
                       "7" | "8" | "9"
      version = DIGIT | (NZDIGIT DIGIT) |
                ("1" DIGIT DIGIT) | ("2" DIGIT DIGIT)
                ; Limited to 0-255 range, with no leading zeros

   The following new header fields can be sent during the handshake from
   the server to the client:

      Sec-WebSocket-Extensions = extension-list
      Sec-WebSocket-Accept     = base64-value-non-empty
      Sec-WebSocket-Protocol-Server = token
      Sec-WebSocket-Version-Server = 1#version

</span></pre> 
   <h3>4.4. Supporting Multiple Versions of WebSocket Protocol</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   This section provides some guidance on supporting multiple versions
   of the WebSocket Protocol in clients and servers.

   Using the WebSocket version advertisement capability (the
   |Sec-WebSocket-Version| header field), a client can initially request
   the version of the WebSocket Protocol that it prefers (which doesn't
   necessarily have to be the latest supported by the client).  If the
   server supports the requested version and the handshake message is
   otherwise valid, the server will accept that version.  If the server
   doesn't support the requested version, it MUST respond with a
   |Sec-WebSocket-Version| header field (or multiple
   |Sec-WebSocket-Version| header fields) containing all versions it is
   willing to use.  At this point, if the client supports one of the
   advertised versions, it can repeat the WebSocket handshake using a
   new version value.

   The following example demonstrates version negotiation described
   above:

      GET /chat HTTP/1.1
      Host: server.example.com
      Upgrade: websocket
      Connection: Upgrade
      ...
      Sec-WebSocket-Version: 25






<span class="grey">Fette & Melnikov             Standards Track                   [Page 26]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   The response from the server might look as follows:

      HTTP/1.1 400 Bad Request
      ...
      Sec-WebSocket-Version: 13, 8, 7

   Note that the last response from the server might also look like:

      HTTP/1.1 400 Bad Request
      ...
      Sec-WebSocket-Version: 13
      Sec-WebSocket-Version: 8, 7

   The client now repeats the handshake that conforms to version 13:

      GET /chat HTTP/1.1
      Host: server.example.com
      Upgrade: websocket
      Connection: Upgrade
      ...
      Sec-WebSocket-Version: 13

</span></pre> 
   <h2>5. Data Framing</h2> 
   <h3>5.1. Overview</h3> 
   <pre class="newpage"><span class="grey"><span class="h2"><span class="h3">

   In the WebSocket Protocol, data is transmitted using a sequence of
   frames.  To avoid confusing network intermediaries (such as
   intercepting proxies) and for security reasons that are further
   discussed in Section 10.3, a client MUST mask all frames that it
   sends to the server (see Section 5.3 for further details).  (Note
   that masking is done whether or not the WebSocket Protocol is running
   over TLS.)  The server MUST close the connection upon receiving a
   frame that is not masked.  In this case, a server MAY send a Close
   frame with a status code of 1002 (protocol error) as defined in
   Section 7.4.1.  A server MUST NOT mask any frames that it sends to
   the client.  A client MUST close a connection if it detects a masked
   frame.  In this case, it MAY use the status code 1002 (protocol
   error) as defined in Section 7.4.1.  (These rules might be relaxed in
   a future specification.)

   The base framing protocol defines a frame type with an opcode, a
   payload length, and designated locations for "Extension data" and
   "Application data", which together define the "Payload data".
   Certain bits and opcodes are reserved for future expansion of the
   protocol.





<span class="grey">Fette & Melnikov             Standards Track                   [Page 27]</span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   A data frame MAY be transmitted by either the client or the server at
   any time after opening handshake completion and before that endpoint
   has sent a Close frame (Section 5.5.1).

</span></pre> 
   <h3>5.2. Base Framing Protocol</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   This wire format for the data transfer part is described by the ABNF
   [RFC5234] given in detail in this section.  (Note that, unlike in
   other sections of this document, the ABNF in this section is
   operating on groups of bits.  The length of each group of bits is
   indicated in a comment.  When encoded on the wire, the most
   significant bit is the leftmost in the ABNF).  A high-level overview
   of the framing is given in the following figure.  In a case of
   conflict between the figure below and the ABNF specified later in
   this section, the figure is authoritative.

      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
     +-+-+-+-+-------+-+-------------+-------------------------------+
     |F|R|R|R| opcode|M| Payload len |    Extended payload length    |
     |I|S|S|S|  (4)  |A|     (7)     |             (16/64)           |
     |N|V|V|V|       |S|             |   (if payload len==126/127)   |
     | |1|2|3|       |K|             |                               |
     +-+-+-+-+-------+-+-------------+ - - - - - - - - - - - - - - - +
     |     Extended payload length continued, if payload len == 127  |
     + - - - - - - - - - - - - - - - +-------------------------------+
     |                               |Masking-key, if MASK set to 1  |
     +-------------------------------+-------------------------------+
     | Masking-key (continued)       |          Payload Data         |
     +-------------------------------- - - - - - - - - - - - - - - - +
     :                     Payload Data continued ...                :
     + - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - +
     |                     Payload Data continued ...                |
     +---------------------------------------------------------------+

   FIN:  1 bit

      Indicates that this is the final fragment in a message.  The first
      fragment MAY also be the final fragment.

   RSV1, RSV2, RSV3:  1 bit each

      MUST be 0 unless an extension is negotiated that defines meanings
      for non-zero values.  If a nonzero value is received and none of
      the negotiated extensions defines the meaning of such a nonzero
      value, the receiving endpoint MUST _Fail the WebSocket
      Connection_.




<span class="grey">Fette & Melnikov             Standards Track                   [Page 28]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   Opcode:  4 bits

      Defines the interpretation of the "Payload data".  If an unknown
      opcode is received, the receiving endpoint MUST _Fail the
      WebSocket Connection_.  The following values are defined.

      *  %x0 denotes a continuation frame

      *  %x1 denotes a text frame

      *  %x2 denotes a binary frame

      *  %x3-7 are reserved for further non-control frames

      *  %x8 denotes a connection close

      *  %x9 denotes a ping

      *  %xA denotes a pong

      *  %xB-F are reserved for further control frames

   Mask:  1 bit

      Defines whether the "Payload data" is masked.  If set to 1, a
      masking key is present in masking-key, and this is used to unmask
      the "Payload data" as per Section 5.3.  All frames sent from
      client to server have this bit set to 1.

   Payload length:  7 bits, 7+16 bits, or 7+64 bits

      The length of the "Payload data", in bytes: if 0-125, that is the
      payload length.  If 126, the following 2 bytes interpreted as a
      16-bit unsigned integer are the payload length.  If 127, the
      following 8 bytes interpreted as a 64-bit unsigned integer (the
      most significant bit MUST be 0) are the payload length.  Multibyte
      length quantities are expressed in network byte order.  Note that
      in all cases, the minimal number of bytes MUST be used to encode
      the length, for example, the length of a 124-byte-long string
      can't be encoded as the sequence 126, 0, 124.  The payload length
      is the length of the "Extension data" + the length of the
      "Application data".  The length of the "Extension data" may be
      zero, in which case the payload length is the length of the
      "Application data".







<span class="grey">Fette & Melnikov             Standards Track                   [Page 29]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   Masking-key:  0 or 4 bytes

      All frames sent from the client to the server are masked by a
      32-bit value that is contained within the frame.  This field is
      present if the mask bit is set to 1 and is absent if the mask bit
      is set to 0.  See Section 5.3 for further information on client-
      to-server masking.

   Payload data:  (x+y) bytes

      The "Payload data" is defined as "Extension data" concatenated
      with "Application data".

   Extension data:  x bytes

      The "Extension data" is 0 bytes unless an extension has been
      negotiated.  Any extension MUST specify the length of the
      "Extension data", or how that length may be calculated, and how
      the extension use MUST be negotiated during the opening handshake.
      If present, the "Extension data" is included in the total payload
      length.

   Application data:  y bytes

      Arbitrary "Application data", taking up the remainder of the frame
      after any "Extension data".  The length of the "Application data"
      is equal to the payload length minus the length of the "Extension
      data".

   The base framing protocol is formally defined by the following ABNF
   [RFC5234].  It is important to note that the representation of this
   data is binary, not ASCII characters.  As such, a field with a length
   of 1 bit that takes values %x0 / %x1 is represented as a single bit
   whose value is 0 or 1, not a full byte (octet) that stands for the
   characters "0" or "1" in the ASCII encoding.  A field with a length
   of 4 bits with values between %x0-F again is represented by 4 bits,
   again NOT by an ASCII character or full byte (octet) with these
   values.  [RFC5234] does not specify a character encoding: "Rules
   resolve into a string of terminal values, sometimes called
   characters.  In ABNF, a character is merely a non-negative integer.
   In certain contexts, a specific mapping (encoding) of values into a
   character set (such as ASCII) will be specified."  Here, the
   specified encoding is a binary encoding where each terminal value is
   encoded in the specified number of bits, which varies for each field.







<span class="grey">Fette & Melnikov             Standards Track                   [Page 30]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


    ws-frame                = frame-fin           ; 1 bit in length
                              frame-rsv1          ; 1 bit in length
                              frame-rsv2          ; 1 bit in length
                              frame-rsv3          ; 1 bit in length
                              frame-opcode        ; 4 bits in length
                              frame-masked        ; 1 bit in length
                              frame-payload-length   ; either 7, 7+16,
                                                     ; or 7+64 bits in
                                                     ; length
                              [ frame-masking-key ]  ; 32 bits in length
                              frame-payload-data     ; n*8 bits in
                                                     ; length, where
                                                     ; n >= 0

    frame-fin               = %x0 ; more frames of this message follow
                            / %x1 ; final frame of this message
                                  ; 1 bit in length

    frame-rsv1              = %x0 / %x1
                              ; 1 bit in length, MUST be 0 unless
                              ; negotiated otherwise

    frame-rsv2              = %x0 / %x1
                              ; 1 bit in length, MUST be 0 unless
                              ; negotiated otherwise

    frame-rsv3              = %x0 / %x1
                              ; 1 bit in length, MUST be 0 unless
                              ; negotiated otherwise

    frame-opcode            = frame-opcode-non-control /
                              frame-opcode-control /
                              frame-opcode-cont

    frame-opcode-cont       = %x0 ; frame continuation

    frame-opcode-non-control= %x1 ; text frame
                            / %x2 ; binary frame
                            / %x3-7
                            ; 4 bits in length,
                            ; reserved for further non-control frames

    frame-opcode-control    = %x8 ; connection close
                            / %x9 ; ping
                            / %xA ; pong
                            / %xB-F ; reserved for further control
                                    ; frames
                                    ; 4 bits in length



<span class="grey">Fette & Melnikov             Standards Track                   [Page 31]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


    frame-masked            = %x0
                            ; frame is not masked, no frame-masking-key
                            / %x1
                            ; frame is masked, frame-masking-key present
                            ; 1 bit in length

    frame-payload-length    = ( %x00-7D )
                            / ( %x7E frame-payload-length-16 )
                            / ( %x7F frame-payload-length-63 )
                            ; 7, 7+16, or 7+64 bits in length,
                            ; respectively

    frame-payload-length-16 = %x0000-FFFF ; 16 bits in length

    frame-payload-length-63 = %x0000000000000000-7FFFFFFFFFFFFFFF
                            ; 64 bits in length

    frame-masking-key       = 4( %x00-FF )
                              ; present only if frame-masked is 1
                              ; 32 bits in length

    frame-payload-data      = (frame-masked-extension-data
                               frame-masked-application-data)
                            ; when frame-masked is 1
                              / (frame-unmasked-extension-data
                                frame-unmasked-application-data)
                            ; when frame-masked is 0

    frame-masked-extension-data     = *( %x00-FF )
                            ; reserved for future extensibility
                            ; n*8 bits in length, where n >= 0

    frame-masked-application-data   = *( %x00-FF )
                            ; n*8 bits in length, where n >= 0

    frame-unmasked-extension-data   = *( %x00-FF )
                            ; reserved for future extensibility
                            ; n*8 bits in length, where n >= 0

    frame-unmasked-application-data = *( %x00-FF )
                            ; n*8 bits in length, where n >= 0

</span></pre> 
   <h3>5.3. Client-to-Server Masking</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   A masked frame MUST have the field frame-masked set to 1, as defined
   in Section 5.2.





<span class="grey">Fette & Melnikov             Standards Track                   [Page 32]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   The masking key is contained completely within the frame, as defined
   in Section 5.2 as frame-masking-key.  It is used to mask the "Payload
   data" defined in the same section as frame-payload-data, which
   includes "Extension data" and "Application data".

   The masking key is a 32-bit value chosen at random by the client.
   When preparing a masked frame, the client MUST pick a fresh masking
   key from the set of allowed 32-bit values.  The masking key needs to
   be unpredictable; thus, the masking key MUST be derived from a strong
   source of entropy, and the masking key for a given frame MUST NOT
   make it simple for a server/proxy to predict the masking key for a
   subsequent frame.  The unpredictability of the masking key is
   essential to prevent authors of malicious applications from selecting
   the bytes that appear on the wire.  RFC 4086 [RFC4086] discusses what
   entails a suitable source of entropy for security-sensitive
   applications.

   The masking does not affect the length of the "Payload data".  To
   convert masked data into unmasked data, or vice versa, the following
   algorithm is applied.  The same algorithm applies regardless of the
   direction of the translation, e.g., the same steps are applied to
   mask the data as to unmask the data.

   Octet i of the transformed data ("transformed-octet-i") is the XOR of
   octet i of the original data ("original-octet-i") with octet at index
   i modulo 4 of the masking key ("masking-key-octet-j"):

     j                   = i MOD 4
     transformed-octet-i = original-octet-i XOR masking-key-octet-j

   The payload length, indicated in the framing as frame-payload-length,
   does NOT include the length of the masking key.  It is the length of
   the "Payload data", e.g., the number of bytes following the masking
   key.

</span></pre> 
   <h3>5.4. Fragmentation</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   The primary purpose of fragmentation is to allow sending a message
   that is of unknown size when the message is started without having to
   buffer that message.  If messages couldn't be fragmented, then an
   endpoint would have to buffer the entire message so its length could
   be counted before the first byte is sent.  With fragmentation, a
   server or intermediary may choose a reasonable size buffer and, when
   the buffer is full, write a fragment to the network.

   A secondary use-case for fragmentation is for multiplexing, where it
   is not desirable for a large message on one logical channel to
   monopolize the output channel, so the multiplexing needs to be free



<span class="grey">Fette & Melnikov             Standards Track                   [Page 33]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   to split the message into smaller fragments to better share the
   output channel.  (Note that the multiplexing extension is not
   described in this document.)

   Unless specified otherwise by an extension, frames have no semantic
   meaning.  An intermediary might coalesce and/or split frames, if no
   extensions were negotiated by the client and the server or if some
   extensions were negotiated, but the intermediary understood all the
   extensions negotiated and knows how to coalesce and/or split frames
   in the presence of these extensions.  One implication of this is that
   in absence of extensions, senders and receivers must not depend on
   the presence of specific frame boundaries.

   The following rules apply to fragmentation:

   o  An unfragmented message consists of a single frame with the FIN
      bit set (Section 5.2) and an opcode other than 0.

   o  A fragmented message consists of a single frame with the FIN bit
      clear and an opcode other than 0, followed by zero or more frames
      with the FIN bit clear and the opcode set to 0, and terminated by
      a single frame with the FIN bit set and an opcode of 0.  A
      fragmented message is conceptually equivalent to a single larger
      message whose payload is equal to the concatenation of the
      payloads of the fragments in order; however, in the presence of
      extensions, this may not hold true as the extension defines the
      interpretation of the "Extension data" present.  For instance,
      "Extension data" may only be present at the beginning of the first
      fragment and apply to subsequent fragments, or there may be
      "Extension data" present in each of the fragments that applies
      only to that particular fragment.  In the absence of "Extension
      data", the following example demonstrates how fragmentation works.

      EXAMPLE: For a text message sent as three fragments, the first
      fragment would have an opcode of 0x1 and a FIN bit clear, the
      second fragment would have an opcode of 0x0 and a FIN bit clear,
      and the third fragment would have an opcode of 0x0 and a FIN bit
      that is set.

   o  Control frames (see Section 5.5) MAY be injected in the middle of
      a fragmented message.  Control frames themselves MUST NOT be
      fragmented.

   o  Message fragments MUST be delivered to the recipient in the order
      sent by the sender.






<span class="grey">Fette & Melnikov             Standards Track                   [Page 34]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   o  The fragments of one message MUST NOT be interleaved between the
      fragments of another message unless an extension has been
      negotiated that can interpret the interleaving.

   o  An endpoint MUST be capable of handling control frames in the
      middle of a fragmented message.

   o  A sender MAY create fragments of any size for non-control
      messages.

   o  Clients and servers MUST support receiving both fragmented and
      unfragmented messages.

   o  As control frames cannot be fragmented, an intermediary MUST NOT
      attempt to change the fragmentation of a control frame.

   o  An intermediary MUST NOT change the fragmentation of a message if
      any reserved bit values are used and the meaning of these values
      is not known to the intermediary.

   o  An intermediary MUST NOT change the fragmentation of any message
      in the context of a connection where extensions have been
      negotiated and the intermediary is not aware of the semantics of
      the negotiated extensions.  Similarly, an intermediary that didn't
      see the WebSocket handshake (and wasn't notified about its
      content) that resulted in a WebSocket connection MUST NOT change
      the fragmentation of any message of such connection.

   o  As a consequence of these rules, all fragments of a message are of
      the same type, as set by the first fragment's opcode.  Since
      control frames cannot be fragmented, the type for all fragments in
      a message MUST be either text, binary, or one of the reserved
      opcodes.

   NOTE: If control frames could not be interjected, the latency of a
   ping, for example, would be very long if behind a large message.
   Hence, the requirement of handling control frames in the middle of a
   fragmented message.

   IMPLEMENTATION NOTE: In the absence of any extension, a receiver
   doesn't have to buffer the whole frame in order to process it.  For
   example, if a streaming API is used, a part of a frame can be
   delivered to the application.  However, note that this assumption
   might not hold true for all future WebSocket extensions.







<span class="grey">Fette & Melnikov             Standards Track                   [Page 35]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


</span></pre> 
   <h3>5.5. Control Frames</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   Control frames are identified by opcodes where the most significant
   bit of the opcode is 1.  Currently defined opcodes for control frames
   include 0x8 (Close), 0x9 (Ping), and 0xA (Pong).  Opcodes 0xB-0xF are
   reserved for further control frames yet to be defined.

   Control frames are used to communicate state about the WebSocket.
   Control frames can be interjected in the middle of a fragmented
   message.

   All control frames MUST have a payload length of 125 bytes or less
   and MUST NOT be fragmented.

</span></span></pre> 
   <h4>5.5.1. Close</h4> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h4">

   The Close frame contains an opcode of 0x8.

   The Close frame MAY contain a body (the "Application data" portion of
   the frame) that indicates a reason for closing, such as an endpoint
   shutting down, an endpoint having received a frame too large, or an
   endpoint having received a frame that does not conform to the format
   expected by the endpoint.  If there is a body, the first two bytes of
   the body MUST be a 2-byte unsigned integer (in network byte order)
   representing a status code with value /code/ defined in Section 7.4.
   Following the 2-byte integer, the body MAY contain UTF-8-encoded data
   with value /reason/, the interpretation of which is not defined by
   this specification.  This data is not necessarily human readable but
   may be useful for debugging or passing information relevant to the
   script that opened the connection.  As the data is not guaranteed to
   be human readable, clients MUST NOT show it to end users.

   Close frames sent from client to server must be masked as per
   Section 5.3.

   The application MUST NOT send any more data frames after sending a
   Close frame.

   If an endpoint receives a Close frame and did not previously send a
   Close frame, the endpoint MUST send a Close frame in response.  (When
   sending a Close frame in response, the endpoint typically echos the
   status code it received.)  It SHOULD do so as soon as practical.  An
   endpoint MAY delay sending a Close frame until its current message is
   sent (for instance, if the majority of a fragmented message is
   already sent, an endpoint MAY send the remaining fragments before
   sending a Close frame).  However, there is no guarantee that the
   endpoint that has already sent a Close frame will continue to process
   data.



<span class="grey">Fette & Melnikov             Standards Track                   [Page 36]</span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   After both sending and receiving a Close message, an endpoint
   considers the WebSocket connection closed and MUST close the
   underlying TCP connection.  The server MUST close the underlying TCP
   connection immediately; the client SHOULD wait for the server to
   close the connection but MAY close the connection at any time after
   sending and receiving a Close message, e.g., if it has not received a
   TCP Close from the server in a reasonable time period.

   If a client and server both send a Close message at the same time,
   both endpoints will have sent and received a Close message and should
   consider the WebSocket connection closed and close the underlying TCP
   connection.

</span></pre> 
   <h4>5.5.2. Ping</h4> 
   <pre class="newpage"><span class="grey"><span class="h4">

   The Ping frame contains an opcode of 0x9.

   A Ping frame MAY include "Application data".

   Upon receipt of a Ping frame, an endpoint MUST send a Pong frame in
   response, unless it already received a Close frame.  It SHOULD
   respond with Pong frame as soon as is practical.  Pong frames are
   discussed in Section 5.5.3.

   An endpoint MAY send a Ping frame any time after the connection is
   established and before the connection is closed.

   NOTE: A Ping frame may serve either as a keepalive or as a means to
   verify that the remote endpoint is still responsive.

</span></span></pre> 
   <h4>5.5.3. Pong</h4> 
   <pre class="newpage"><span class="grey"><span class="h4"><span class="h4">

   The Pong frame contains an opcode of 0xA.

   Section 5.5.2 details requirements that apply to both Ping and Pong
   frames.

   A Pong frame sent in response to a Ping frame must have identical
   "Application data" as found in the message body of the Ping frame
   being replied to.

   If an endpoint receives a Ping frame and has not yet sent Pong
   frame(s) in response to previous Ping frame(s), the endpoint MAY
   elect to send a Pong frame for only the most recently processed Ping
   frame.






<span class="grey">Fette & Melnikov             Standards Track                   [Page 37]</span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   A Pong frame MAY be sent unsolicited.  This serves as a
   unidirectional heartbeat.  A response to an unsolicited Pong frame is
   not expected.

</span></pre> 
   <h3>5.6. Data Frames</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   Data frames (e.g., non-control frames) are identified by opcodes
   where the most significant bit of the opcode is 0.  Currently defined
   opcodes for data frames include 0x1 (Text), 0x2 (Binary).  Opcodes
   0x3-0x7 are reserved for further non-control frames yet to be
   defined.

   Data frames carry application-layer and/or extension-layer data.  The
   opcode determines the interpretation of the data:

   Text

      The "Payload data" is text data encoded as UTF-8.  Note that a
      particular text frame might include a partial UTF-8 sequence;
      however, the whole message MUST contain valid UTF-8.  Invalid
      UTF-8 in reassembled messages is handled as described in
      Section 8.1.

   Binary

      The "Payload data" is arbitrary binary data whose interpretation
      is solely up to the application layer.

</span></span></pre> 
   <h3>5.7. Examples</h3> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h3">

   o  A single-frame unmasked text message

      *  0x81 0x05 0x48 0x65 0x6c 0x6c 0x6f (contains "Hello")

   o  A single-frame masked text message

      *  0x81 0x85 0x37 0xfa 0x21 0x3d 0x7f 0x9f 0x4d 0x51 0x58
         (contains "Hello")

   o  A fragmented unmasked text message

      *  0x01 0x03 0x48 0x65 0x6c (contains "Hel")

      *  0x80 0x02 0x6c 0x6f (contains "lo")







<span class="grey">Fette & Melnikov             Standards Track                   [Page 38]</span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   o  Unmasked Ping request and masked Ping response

      *  0x89 0x05 0x48 0x65 0x6c 0x6c 0x6f (contains a body of "Hello",
         but the contents of the body are arbitrary)

      *  0x8a 0x85 0x37 0xfa 0x21 0x3d 0x7f 0x9f 0x4d 0x51 0x58
         (contains a body of "Hello", matching the body of the ping)

   o  256 bytes binary message in a single unmasked frame

      *  0x82 0x7E 0x0100 [256 bytes of binary data]

   o  64KiB binary message in a single unmasked frame

      *  0x82 0x7F 0x0000000000010000 [65536 bytes of binary data]

</span></pre> 
   <h3>5.8. Extensibility</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   The protocol is designed to allow for extensions, which will add
   capabilities to the base protocol.  The endpoints of a connection
   MUST negotiate the use of any extensions during the opening
   handshake.  This specification provides opcodes 0x3 through 0x7 and
   0xB through 0xF, the "Extension data" field, and the frame-rsv1,
   frame-rsv2, and frame-rsv3 bits of the frame header for use by
   extensions.  The negotiation of extensions is discussed in further
   detail in Section 9.1.  Below are some anticipated uses of
   extensions.  This list is neither complete nor prescriptive.

   o  "Extension data" may be placed in the "Payload data" before the
      "Application data".

   o  Reserved bits can be allocated for per-frame needs.

   o  Reserved opcode values can be defined.

   o  Reserved bits can be allocated to the opcode field if more opcode
      values are needed.

   o  A reserved bit or an "extension" opcode can be defined that
      allocates additional bits out of the "Payload data" to define
      larger opcodes or more per-frame bits.

</span></span></pre> 
   <h2>6. Sending and Receiving Data</h2> 
   <h3>6.1. Sending Data</h3> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h2"><span class="h3">

   To _Send a WebSocket Message_ comprising of /data/ over a WebSocket
   connection, an endpoint MUST perform the following steps.



<span class="grey">Fette & Melnikov             Standards Track                   [Page 39]</span></span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   1.  The endpoint MUST ensure the WebSocket connection is in the OPEN
       state (cf. Sections 4.1 and 4.2.2.)  If at any point the state of
       the WebSocket connection changes, the endpoint MUST abort the
       following steps.

   2.  An endpoint MUST encapsulate the /data/ in a WebSocket frame as
       defined in Section 5.2.  If the data to be sent is large or if
       the data is not available in its entirety at the point the
       endpoint wishes to begin sending the data, the endpoint MAY
       alternately encapsulate the data in a series of frames as defined
       in Section 5.4.

   3.  The opcode (frame-opcode) of the first frame containing the data
       MUST be set to the appropriate value from Section 5.2 for data
       that is to be interpreted by the recipient as text or binary
       data.

   4.  The FIN bit (frame-fin) of the last frame containing the data
       MUST be set to 1 as defined in Section 5.2.

   5.  If the data is being sent by the client, the frame(s) MUST be
       masked as defined in Section 5.3.

   6.  If any extensions (Section 9) have been negotiated for the
       WebSocket connection, additional considerations may apply as per
       the definition of those extensions.

   7.  The frame(s) that have been formed MUST be transmitted over the
       underlying network connection.

</span></pre> 
   <h3>6.2. Receiving Data</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   To receive WebSocket data, an endpoint listens on the underlying
   network connection.  Incoming data MUST be parsed as WebSocket frames
   as defined in Section 5.2.  If a control frame (Section 5.5) is
   received, the frame MUST be handled as defined by Section 5.5.  Upon
   receiving a data frame (Section 5.6), the endpoint MUST note the
   /type/ of the data as defined by the opcode (frame-opcode) from
   Section 5.2.  The "Application data" from this frame is defined as
   the /data/ of the message.  If the frame comprises an unfragmented
   message (Section 5.4), it is said that _A WebSocket Message Has Been
   Received_ with type /type/ and data /data/.  If the frame is part of
   a fragmented message, the "Application data" of the subsequent data
   frames is concatenated to form the /data/.  When the last fragment is
   received as indicated by the FIN bit (frame-fin), it is said that _A
   WebSocket Message Has Been Received_ with data /data/ (comprised of
   the concatenation of the "Application data" of the fragments) and




<span class="grey">Fette & Melnikov             Standards Track                   [Page 40]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   type /type/ (noted from the first frame of the fragmented message).
   Subsequent data frames MUST be interpreted as belonging to a new
   WebSocket message.

   Extensions (Section 9) MAY change the semantics of how data is read,
   specifically including what comprises a message boundary.
   Extensions, in addition to adding "Extension data" before the
   "Application data" in a payload, MAY also modify the "Application
   data" (such as by compressing it).

   A server MUST remove masking for data frames received from a client
   as described in Section 5.3.

</span></pre> 
   <h2>7. Closing the Connection</h2> 
   <h3>7.1. Definitions</h3> 
   <h4>7.1.1. Close the WebSocket Connection</h4> 
   <pre class="newpage"><span class="grey"><span class="h2"><span class="h3"><span class="h4">

   To _Close the WebSocket Connection_, an endpoint closes the
   underlying TCP connection.  An endpoint SHOULD use a method that
   cleanly closes the TCP connection, as well as the TLS session, if
   applicable, discarding any trailing bytes that may have been
   received.  An endpoint MAY close the connection via any means
   available when necessary, such as when under attack.

   The underlying TCP connection, in most normal cases, SHOULD be closed
   first by the server, so that it holds the TIME_WAIT state and not the
   client (as this would prevent it from re-opening the connection for 2
   maximum segment lifetimes (2MSL), while there is no corresponding
   server impact as a TIME_WAIT connection is immediately reopened upon
   a new SYN with a higher seq number).  In abnormal cases (such as not
   having received a TCP Close from the server after a reasonable amount
   of time) a client MAY initiate the TCP Close.  As such, when a server
   is instructed to _Close the WebSocket Connection_ it SHOULD initiate
   a TCP Close immediately, and when a client is instructed to do the
   same, it SHOULD wait for a TCP Close from the server.

   As an example of how to obtain a clean closure in C using Berkeley
   sockets, one would call shutdown() with SHUT_WR on the socket, call
   recv() until obtaining a return value of 0 indicating that the peer
   has also performed an orderly shutdown, and finally call close() on
   the socket.








<span class="grey">Fette & Melnikov             Standards Track                   [Page 41]</span></span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


</span></pre> 
   <h4>7.1.2. Start the WebSocket Closing Handshake</h4> 
   <pre class="newpage"><span class="grey"><span class="h4">

   To _Start the WebSocket Closing Handshake_ with a status code
   (Section 7.4) /code/ and an optional close reason (Section 7.1.6)
   /reason/, an endpoint MUST send a Close control frame, as described
   in Section 5.5.1, whose status code is set to /code/ and whose close
   reason is set to /reason/.  Once an endpoint has both sent and
   received a Close control frame, that endpoint SHOULD _Close the
   WebSocket Connection_ as defined in Section 7.1.1.

</span></span></pre> 
   <h4>7.1.3. The WebSocket Closing Handshake is Started</h4> 
   <pre class="newpage"><span class="grey"><span class="h4"><span class="h4">

   Upon either sending or receiving a Close control frame, it is said
   that _The WebSocket Closing Handshake is Started_ and that the
   WebSocket connection is in the CLOSING state.

</span></span></span></pre> 
   <h4>7.1.4. The WebSocket Connection is Closed</h4> 
   <pre class="newpage"><span class="grey"><span class="h4"><span class="h4"><span class="h4">

   When the underlying TCP connection is closed, it is said that _The
   WebSocket Connection is Closed_ and that the WebSocket connection is
   in the CLOSED state.  If the TCP connection was closed after the
   WebSocket closing handshake was completed, the WebSocket connection
   is said to have been closed _cleanly_.

   If the WebSocket connection could not be established, it is also said
   that _The WebSocket Connection is Closed_, but not _cleanly_.

</span></span></span></span></pre> 
   <h4>7.1.5. The WebSocket Connection Close Code</h4> 
   <pre class="newpage"><span class="grey"><span class="h4"><span class="h4"><span class="h4"><span class="h4">

   As defined in Sections 5.5.1 and 7.4, a Close control frame may
   contain a status code indicating a reason for closure.  A closing of
   the WebSocket connection may be initiated by either endpoint,
   potentially simultaneously. _The WebSocket Connection Close Code_ is
   defined as the status code (Section 7.4) contained in the first Close
   control frame received by the application implementing this protocol.
   If this Close control frame contains no status code, _The WebSocket
   Connection Close Code_ is considered to be 1005.  If _The WebSocket
   Connection is Closed_ and no Close control frame was received by the
   endpoint (such as could occur if the underlying transport connection
   is lost), _The WebSocket Connection Close Code_ is considered to be
   1006.

   NOTE: Two endpoints may not agree on the value of _The WebSocket
   Connection Close Code_.  As an example, if the remote endpoint sent a
   Close frame but the local application has not yet read the data
   containing the Close frame from its socket's receive buffer, and the
   local application independently decided to close the connection and
   send a Close frame, both endpoints will have sent and received a



<span class="grey">Fette & Melnikov Standards Track [Page 42]</span></span></span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   Close frame and will not send further Close frames.  Each endpoint
   will see the status code sent by the other end as _The WebSocket
   Connection Close Code_.  As such, it is possible that the two
   endpoints may not agree on the value of _The WebSocket Connection
   Close Code_ in the case that both endpoints _Start the WebSocket
   Closing Handshake_ independently and at roughly the same time.

</span></pre> 
   <h4>7.1.6. The WebSocket Connection Close Reason</h4> 
   <pre class="newpage"><span class="grey"><span class="h4">

   As defined in Sections 5.5.1 and 7.4, a Close control frame may
   contain a status code indicating a reason for closure, followed by
   UTF-8-encoded data, the interpretation of said data being left to the
   endpoints and not defined by this protocol.  A closing of the
   WebSocket connection may be initiated by either endpoint, potentially
   simultaneously. _The WebSocket Connection Close Reason_ is defined as
   the UTF-8-encoded data following the status code (Section 7.4)
   contained in the first Close control frame received by the
   application implementing this protocol.  If there is no such data in
   the Close control frame, _The WebSocket Connection Close Reason_ is
   the empty string.

   NOTE: Following the same logic as noted in Section 7.1.5, two
   endpoints may not agree on _The WebSocket Connection Close Reason_.

</span></span></pre> 
   <h4>7.1.7. Fail the WebSocket Connection</h4> 
   <pre class="newpage"><span class="grey"><span class="h4"><span class="h4">

   Certain algorithms and specifications require an endpoint to _Fail
   the WebSocket Connection_.  To do so, the client MUST _Close the
   WebSocket Connection_, and MAY report the problem to the user (which
   would be especially useful for developers) in an appropriate manner.
   Similarly, to do so, the server MUST _Close the WebSocket
   Connection_, and SHOULD log the problem.

   If _The WebSocket Connection is Established_ prior to the point where
   the endpoint is required to _Fail the WebSocket Connection_, the
   endpoint SHOULD send a Close frame with an appropriate status code
   (Section 7.4) before proceeding to _Close the WebSocket Connection_.
   An endpoint MAY omit sending a Close frame if it believes the other
   side is unlikely to be able to receive and process the Close frame,
   due to the nature of the error that led the WebSocket connection to
   fail in the first place.  An endpoint MUST NOT continue to attempt to
   process data (including a responding Close frame) from the remote
   endpoint after being instructed to _Fail the WebSocket Connection_.

   Except as indicated above or as specified by the application layer
   (e.g., a script using the WebSocket API), clients SHOULD NOT close
   the connection.




<span class="grey">Fette & Melnikov             Standards Track                   [Page 43]</span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


</span></pre> 
   <h3>7.2. Abnormal Closures</h3> 
   <h4>7.2.1. Client-Initiated Closure</h4> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h4">

   Certain algorithms, in particular during the opening handshake,
   require the client to _Fail the WebSocket Connection_.  To do so, the
   client MUST _Fail the WebSocket Connection_ as defined in
   Section 7.1.7.

   If at any point the underlying transport layer connection is
   unexpectedly lost, the client MUST _Fail the WebSocket Connection_.

   Except as indicated above or as specified by the application layer
   (e.g., a script using the WebSocket API), clients SHOULD NOT close
   the connection.

</span></span></span></pre> 
   <h4>7.2.2. Server-Initiated Closure</h4> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h4"><span class="h4">

   Certain algorithms require or recommend that the server _Abort the
   WebSocket Connection_ during the opening handshake.  To do so, the
   server MUST simply _Close the WebSocket Connection_ (Section 7.1.1).

</span></span></span></span></pre> 
   <h4>7.2.3. Recovering from Abnormal Closure</h4> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h4"><span class="h4"><span class="h4">

   Abnormal closures may be caused by any number of reasons.  Such
   closures could be the result of a transient error, in which case
   reconnecting may lead to a good connection and a resumption of normal
   operations.  Such closures may also be the result of a nontransient
   problem, in which case if each deployed client experiences an
   abnormal closure and immediately and persistently tries to reconnect,
   the server may experience what amounts to a denial-of-service attack
   by a large number of clients trying to reconnect.  The end result of
   such a scenario could be that the service is unable to recover in a
   timely manner or recovery is made much more difficult.

   To prevent this, clients SHOULD use some form of backoff when trying
   to reconnect after abnormal closures as described in this section.

   The first reconnect attempt SHOULD be delayed by a random amount of
   time.  The parameters by which this random delay is chosen are left
   to the client to decide; a value chosen randomly between 0 and 5
   seconds is a reasonable initial delay though clients MAY choose a
   different interval from which to select a delay length based on
   implementation experience and particular application.

   Should the first reconnect attempt fail, subsequent reconnect
   attempts SHOULD be delayed by increasingly longer amounts of time,
   using a method such as truncated binary exponential backoff.



<span class="grey">Fette & Melnikov Standards Track [Page 44]</span></span></span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


</span></pre> 
   <h3>7.3. Normal Closure of Connections</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   Servers MAY close the WebSocket connection whenever desired.  Clients
   SHOULD NOT close the WebSocket connection arbitrarily.  In either
   case, an endpoint initiates a closure by following the procedures to
   _Start the WebSocket Closing Handshake_ (Section 7.1.2).

</span></span></pre> 
   <h3>7.4. Status Codes</h3> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h3">

   When closing an established connection (e.g., when sending a Close
   frame, after the opening handshake has completed), an endpoint MAY
   indicate a reason for closure.  The interpretation of this reason by
   an endpoint, and the action an endpoint should take given this
   reason, are left undefined by this specification.  This specification
   defines a set of pre-defined status codes and specifies which ranges
   may be used by extensions, frameworks, and end applications.  The
   status code and any associated textual message are optional
   components of a Close frame.

</span></span></span></pre> 
   <h4>7.4.1. Defined Status Codes</h4> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h3"><span class="h4">

   Endpoints MAY use the following pre-defined status codes when sending
   a Close frame.

   1000

      1000 indicates a normal closure, meaning that the purpose for
      which the connection was established has been fulfilled.

   1001

      1001 indicates that an endpoint is "going away", such as a server
      going down or a browser having navigated away from a page.

   1002

      1002 indicates that an endpoint is terminating the connection due
      to a protocol error.

   1003

      1003 indicates that an endpoint is terminating the connection
      because it has received a type of data it cannot accept (e.g., an
      endpoint that understands only text data MAY send this if it
      receives a binary message).






<span class="grey">Fette & Melnikov             Standards Track                   [Page 45]</span></span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   1004

      Reserved.  The specific meaning might be defined in the future.

   1005

      1005 is a reserved value and MUST NOT be set as a status code in a
      Close control frame by an endpoint.  It is designated for use in
      applications expecting a status code to indicate that no status
      code was actually present.

   1006

      1006 is a reserved value and MUST NOT be set as a status code in a
      Close control frame by an endpoint.  It is designated for use in
      applications expecting a status code to indicate that the
      connection was closed abnormally, e.g., without sending or
      receiving a Close control frame.

   1007

      1007 indicates that an endpoint is terminating the connection
      because it has received data within a message that was not
      consistent with the type of the message (e.g., non-UTF-8 [RFC3629]
      data within a text message).

   1008

      1008 indicates that an endpoint is terminating the connection
      because it has received a message that violates its policy.  This
      is a generic status code that can be returned when there is no
      other more suitable status code (e.g., 1003 or 1009) or if there
      is a need to hide specific details about the policy.

   1009

      1009 indicates that an endpoint is terminating the connection
      because it has received a message that is too big for it to
      process.

   1010

      1010 indicates that an endpoint (client) is terminating the
      connection because it has expected the server to negotiate one or
      more extension, but the server didn't return them in the response
      message of the WebSocket handshake.  The list of extensions that





<span class="grey">Fette & Melnikov             Standards Track                   [Page 46]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


      are needed SHOULD appear in the /reason/ part of the Close frame.
      Note that this status code is not used by the server, because it
      can fail the WebSocket handshake instead.

   1011

      1011 indicates that a server is terminating the connection because
      it encountered an unexpected condition that prevented it from
      fulfilling the request.

   1015

      1015 is a reserved value and MUST NOT be set as a status code in a
      Close control frame by an endpoint.  It is designated for use in
      applications expecting a status code to indicate that the
      connection was closed due to a failure to perform a TLS handshake
      (e.g., the server certificate can't be verified).

</span></pre> 
   <h4>7.4.2. Reserved Status Code Ranges</h4> 
   <pre class="newpage"><span class="grey"><span class="h4">

   0-999

      Status codes in the range 0-999 are not used.

   1000-2999

      Status codes in the range 1000-2999 are reserved for definition by
      this protocol, its future revisions, and extensions specified in a
      permanent and readily available public specification.

   3000-3999

      Status codes in the range 3000-3999 are reserved for use by
      libraries, frameworks, and applications.  These status codes are
      registered directly with IANA.  The interpretation of these codes
      is undefined by this protocol.

   4000-4999

      Status codes in the range 4000-4999 are reserved for private use
      and thus can't be registered.  Such codes can be used by prior
      agreements between WebSocket applications.  The interpretation of
      these codes is undefined by this protocol.








<span class="grey">Fette & Melnikov             Standards Track                   [Page 47]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


</span></pre> 
   <h2>8. Error Handling</h2> 
   <h3>8.1. Handling Errors in UTF-8-Encoded Data</h3> 
   <pre class="newpage"><span class="grey"><span class="h2"><span class="h3">

   When an endpoint is to interpret a byte stream as UTF-8 but finds
   that the byte stream is not, in fact, a valid UTF-8 stream, that
   endpoint MUST _Fail the WebSocket Connection_.  This rule applies
   both during the opening handshake and during subsequent data
   exchange.

</span></span></span></pre> 
   <h2>9. Extensions</h2> 
   <pre class="newpage"><span class="grey"><span class="h2"><span class="h3"><span class="h2">

   WebSocket clients MAY request extensions to this specification, and
   WebSocket servers MAY accept some or all extensions requested by the
   client.  A server MUST NOT respond with any extension not requested
   by the client.  If extension parameters are included in negotiations
   between the client and the server, those parameters MUST be chosen in
   accordance with the specification of the extension to which the
   parameters apply.

</span></span></span></span></pre> 
   <h3>9.1. Negotiating Extensions</h3> 
   <pre class="newpage"><span class="grey"><span class="h2"><span class="h3"><span class="h2"><span class="h3">

   A client requests extensions by including a |Sec-WebSocket-
   Extensions| header field, which follows the normal rules for HTTP
   header fields (see [RFC2616], Section 4.2) and the value of the
   header field is defined by the following ABNF [RFC2616].  Note that
   this section is using ABNF syntax/rules from [RFC2616], including the
   "implied *LWS rule".  If a value is received by either the client or
   the server during negotiation that does not conform to the ABNF
   below, the recipient of such malformed data MUST immediately _Fail
   the WebSocket Connection_.

         Sec-WebSocket-Extensions = extension-list
         extension-list = 1#extension
         extension = extension-token *( ";" extension-param )
         extension-token = registered-token
         registered-token = token
         extension-param = token [ "=" (token | quoted-string) ]
             ;When using the quoted-string syntax variant, the value
             ;after quoted-string unescaping MUST conform to the
             ;'token' ABNF.










<span class="grey">Fette & Melnikov Standards Track [Page 48]</span></span></span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   Note that like other HTTP header fields, this header field MAY be
   split or combined across multiple lines.  Ergo, the following are
   equivalent:

         Sec-WebSocket-Extensions: foo
         Sec-WebSocket-Extensions: bar; baz=2

   is exactly equivalent to

         Sec-WebSocket-Extensions: foo, bar; baz=2

   Any extension-token used MUST be a registered token (see
   Section 11.4).  The parameters supplied with any given extension MUST
   be defined for that extension.  Note that the client is only offering
   to use any advertised extensions and MUST NOT use them unless the
   server indicates that it wishes to use the extension.

   Note that the order of extensions is significant.  Any interactions
   between multiple extensions MAY be defined in the documents defining
   the extensions.  In the absence of such definitions, the
   interpretation is that the header fields listed by the client in its
   request represent a preference of the header fields it wishes to use,
   with the first options listed being most preferable.  The extensions
   listed by the server in response represent the extensions actually in
   use for the connection.  Should the extensions modify the data and/or
   framing, the order of operations on the data should be assumed to be
   the same as the order in which the extensions are listed in the
   server's response in the opening handshake.

   For example, if there are two extensions "foo" and "bar" and if the
   header field |Sec-WebSocket-Extensions| sent by the server has the
   value "foo, bar", then operations on the data will be made as
   bar(foo(data)), be those changes to the data itself (such as
   compression) or changes to the framing that may "stack".

   Non-normative examples of acceptable extension header fields (note
   that long lines are folded for readability):

         Sec-WebSocket-Extensions: deflate-stream
         Sec-WebSocket-Extensions: mux; max-channels=4; flow-control,
          deflate-stream
         Sec-WebSocket-Extensions: private-extension

   A server accepts one or more extensions by including a
   |Sec-WebSocket-Extensions| header field containing one or more
   extensions that were requested by the client.  The interpretation of





<span class="grey">Fette & Melnikov             Standards Track                   [Page 49]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   any extension parameters, and what constitutes a valid response by a
   server to a requested set of parameters by a client, will be defined
   by each such extension.

</span></pre> 
   <h3>9.2. Known Extensions</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   Extensions provide a mechanism for implementations to opt-in to
   additional protocol features.  This document doesn't define any
   extension, but implementations MAY use extensions defined separately.

</span></span></pre> 
   <h2>10. Security Considerations</h2> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h2">

   This section describes some security considerations applicable to the
   WebSocket Protocol.  Specific security considerations are described
   in subsections of this section.

</span></span></span></pre> 
   <h3>10.1. Non-Browser Clients</h3> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h2"><span class="h3">

   The WebSocket Protocol protects against malicious JavaScript running
   inside a trusted application such as a web browser, for example, by
   checking of the |Origin| header field (see below).  See Section 1.6
   for additional details.  Such assumptions don't hold true in the case
   of a more-capable client.

   While this protocol is intended to be used by scripts in web pages,
   it can also be used directly by hosts.  Such hosts are acting on
   their own behalf and can therefore send fake |Origin| header fields,
   misleading the server.  Servers should therefore be careful about
   assuming that they are talking directly to scripts from known origins
   and must consider that they might be accessed in unexpected ways.  In
   particular, a server should not trust that any input is valid.

   EXAMPLE: If the server uses input as part of SQL queries, all input
   text should be escaped before being passed to the SQL server, lest
   the server be susceptible to SQL injection.

</span></span></span></span></pre> 
   <h3>10.2. Origin Considerations</h3> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h2"><span class="h3"><span class="h3">

   Servers that are not intended to process input from any web page but
   only for certain sites SHOULD verify the |Origin| field is an origin
   they expect.  If the origin indicated is unacceptable to the server,
   then it SHOULD respond to the WebSocket handshake with a reply
   containing HTTP 403 Forbidden status code.

   The |Origin| header field protects from the attack cases when the
   untrusted party is typically the author of a JavaScript application
   that is executing in the context of the trusted client.  The client
   itself can contact the server and, via the mechanism of the |Origin|



<span class="grey">Fette & Melnikov Standards Track [Page 50]</span></span></span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   header field, determine whether to extend those communication
   privileges to the JavaScript application.  The intent is not to
   prevent non-browsers from establishing connections but rather to
   ensure that trusted browsers under the control of potentially
   malicious JavaScript cannot fake a WebSocket handshake.

</span></pre> 
   <h3>10.3. Attacks On Infrastructure (Masking)</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   In addition to endpoints being the target of attacks via WebSockets,
   other parts of web infrastructure, such as proxies, may be the
   subject of an attack.

   As this protocol was being developed, an experiment was conducted to
   demonstrate a class of attacks on proxies that led to the poisoning
   of caching proxies deployed in the wild [TALKING].  The general form
   of the attack was to establish a connection to a server under the
   "attacker's" control, perform an UPGRADE on the HTTP connection
   similar to what the WebSocket Protocol does to establish a
   connection, and subsequently send data over that UPGRADEd connection
   that looked like a GET request for a specific known resource (which
   in an attack would likely be something like a widely deployed script
   for tracking hits or a resource on an ad-serving network).  The
   remote server would respond with something that looked like a
   response to the fake GET request, and this response would be cached
   by a nonzero percentage of deployed intermediaries, thus poisoning
   the cache.  The net effect of this attack would be that if a user
   could be convinced to visit a website the attacker controlled, the
   attacker could potentially poison the cache for that user and other
   users behind the same cache and run malicious script on other
   origins, compromising the web security model.

   To avoid such attacks on deployed intermediaries, it is not
   sufficient to prefix application-supplied data with framing that is
   not compliant with HTTP, as it is not possible to exhaustively
   discover and test that each nonconformant intermediary does not skip
   such non-HTTP framing and act incorrectly on the frame payload.
   Thus, the defense adopted is to mask all data from the client to the
   server, so that the remote script (attacker) does not have control
   over how the data being sent appears on the wire and thus cannot
   construct a message that could be misinterpreted by an intermediary
   as an HTTP request.

   Clients MUST choose a new masking key for each frame, using an
   algorithm that cannot be predicted by end applications that provide
   data.  For example, each masking could be drawn from a
   cryptographically strong random number generator.  If the same key is
   used or a decipherable pattern exists for how the next key is chosen,
   the attacker can send a message that, when masked, could appear to be



<span class="grey">Fette & Melnikov             Standards Track                   [Page 51]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   an HTTP request (by taking the message the attacker wishes to see on
   the wire and masking it with the next masking key to be used, the
   masking key will effectively unmask the data when the client applies
   it).

   It is also necessary that once the transmission of a frame from a
   client has begun, the payload (application-supplied data) of that
   frame must not be capable of being modified by the application.
   Otherwise, an attacker could send a long frame where the initial data
   was a known value (such as all zeros), compute the masking key being
   used upon receipt of the first part of the data, and then modify the
   data that is yet to be sent in the frame to appear as an HTTP request
   when masked.  (This is essentially the same problem described in the
   previous paragraph with using a known or predictable masking key.)
   If additional data is to be sent or data to be sent is somehow
   changed, that new or changed data must be sent in a new frame and
   thus with a new masking key.  In short, once transmission of a frame
   begins, the contents must not be modifiable by the remote script
   (application).

   The threat model being protected against is one in which the client
   sends data that appears to be an HTTP request.  As such, the channel
   that needs to be masked is the data from the client to the server.
   The data from the server to the client can be made to look like a
   response, but to accomplish this request, the client must also be
   able to forge a request.  As such, it was not deemed necessary to
   mask data in both directions (the data from the server to the client
   is not masked).

   Despite the protection provided by masking, non-compliant HTTP
   proxies will still be vulnerable to poisoning attacks of this type by
   clients and servers that do not apply masking.

</span></pre> 
   <h3>10.4. Implementation-Specific Limits</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   Implementations that have implementation- and/or platform-specific
   limitations regarding the frame size or total message size after
   reassembly from multiple frames MUST protect themselves against
   exceeding those limits.  (For example, a malicious endpoint can try
   to exhaust its peer's memory or mount a denial-of-service attack by
   sending either a single big frame (e.g., of size 2**60) or by sending
   a long stream of small frames that are a part of a fragmented
   message.)  Such an implementation SHOULD impose a limit on frame
   sizes and the total message size after reassembly from multiple
   frames.






<span class="grey">Fette & Melnikov             Standards Track                   [Page 52]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


</span></pre> 
   <h3>10.5. WebSocket Client Authentication</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   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.

</span></span></pre> 
   <h3>10.6. Connection Confidentiality and Integrity</h3> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h3">

   Connection confidentiality and integrity is provided by running the
   WebSocket Protocol over TLS (wss URIs).  WebSocket implementations
   MUST support TLS and SHOULD employ it when communicating with their
   peers.

   For connections using TLS, the amount of benefit provided by TLS
   depends greatly on the strength of the algorithms negotiated during
   the TLS handshake.  For example, some TLS cipher mechanisms don't
   provide connection confidentiality.  To achieve reasonable levels of
   protection, clients should use only Strong TLS algorithms.  "Web
   Security Context: User Interface Guidelines"
   [W3C.REC-wsc-ui-20100812] discusses what constitutes Strong TLS
   algorithms.  [RFC5246] provides additional guidance in Appendix A.5
   and Appendix D.3.

</span></span></span></pre> 
   <h3>10.7. Handling of Invalid Data</h3> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h3"><span class="h3">

   Incoming data MUST always be validated by both clients and servers.
   If, at any time, an endpoint is faced with data that it does not
   understand or that violates some criteria by which the endpoint
   determines safety of input, or when the endpoint sees an opening
   handshake that does not correspond to the values it is expecting
   (e.g., incorrect path or origin in the client request), the endpoint
   MAY drop the TCP connection.  If the invalid data was received after
   a successful WebSocket handshake, the endpoint SHOULD send a Close
   frame with an appropriate status code (Section 7.4) before proceeding
   to _Close the WebSocket Connection_.  Use of a Close frame with an
   appropriate status code can help in diagnosing the problem.  If the
   invalid data is sent during the WebSocket handshake, the server
   SHOULD return an appropriate HTTP [RFC2616] status code.

   A common class of security problems arises when sending text data
   using the wrong encoding.  This protocol specifies that messages with
   a Text data type (as opposed to Binary or other types) contain UTF-8-
   encoded data.  Although the length is still indicated and
   applications implementing this protocol should use the length to
   determine where the frame actually ends, sending data in an improper




<span class="grey">Fette & Melnikov             Standards Track                   [Page 53]</span></span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   encoding may still break assumptions that applications built on top
   of this protocol may make, leading to anything from misinterpretation
   of data to loss of data or potential security bugs.

</span></pre> 
   <h3>10.8. Use of SHA-1 by the WebSocket Handshake</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   The WebSocket handshake described in this document doesn't depend on
   any security properties of SHA-1, such as collision resistance or
   resistance to the second pre-image attack (as described in
   [RFC4270]).

</span></span></pre> 
   <h2>11. IANA Considerations</h2> 
   <h3>11.1. Registration of New URI Schemes</h3> 
   <h4>11.1.1. Registration of "ws" Scheme</h4> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h2"><span class="h3"><span class="h4">

   A |ws| URI identifies a WebSocket server and resource name.

   URI scheme name
      ws

   Status
      Permanent

   URI scheme syntax
      Using the ABNF [RFC5234] syntax and ABNF terminals from the URI
      specification [RFC3986]:

           "ws:" "//" authority path-abempty [ "?" query ]

   The <path-abempty> and <query> [RFC3986] components form the resource
   name sent to the server to identify the kind of service desired.
   Other components have the meanings described in [RFC3986].

   URI scheme semantics
      The only operation for this scheme is to open a connection using
      the WebSocket Protocol.

   Encoding considerations
      Characters in the host component that are excluded by the syntax
      defined above MUST be converted from Unicode to ASCII as specified
      in [RFC3987] or its replacement.  For the purposes of scheme-based
      normalization, Internationalized Domain Name (IDN) forms of the
      host component and their conversions to punycode are considered
      equivalent (see Section 5.3.3 of [RFC3987]).





<span class="grey">Fette & Melnikov Standards Track [Page 54]</span></span></span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


      Characters in other components that are excluded by the syntax
      defined above MUST be converted from Unicode to ASCII by first
      encoding the characters as UTF-8 and then replacing the
      corresponding bytes using their percent-encoded form as defined in
      the URI [RFC3986] and Internationalized Resource Identifier (IRI)
      [RFC3987] specifications.

   Applications/protocols that use this URI scheme name
      WebSocket Protocol

   Interoperability considerations
      Use of WebSocket requires use of HTTP version 1.1 or higher.

   Security considerations
      See "Security Considerations" section.

   Contact
      HYBI WG <hybi@ietf.org>

   Author/Change controller
      IETF <iesg@ietf.org>

   References
      RFC 6455

</span></pre> 
   <h4>11.1.2. Registration of "wss" Scheme</h4> 
   <pre class="newpage"><span class="grey"><span class="h4">

   A |wss| URI identifies a WebSocket server and resource name and
   indicates that traffic over that connection is to be protected via
   TLS (including standard benefits of TLS such as data confidentiality
   and integrity and endpoint authentication).

   URI scheme name
      wss

   Status
      Permanent

   URI scheme syntax
      Using the ABNF [RFC5234] syntax and ABNF terminals from the URI
      specification [RFC3986]:

           "wss:" "//" authority path-abempty [ "?" query ]

   The <path-abempty> and <query> components form the resource name sent
   to the server to identify the kind of service desired.  Other
   components have the meanings described in [RFC3986].




<span class="grey">Fette & Melnikov             Standards Track                   [Page 55]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   URI scheme semantics
      The only operation for this scheme is to open a connection using
      the WebSocket Protocol, encrypted using TLS.

   Encoding considerations
      Characters in the host component that are excluded by the syntax
      defined above MUST be converted from Unicode to ASCII as specified
      in [RFC3987] or its replacement.  For the purposes of scheme-based
      normalization IDN forms of the host component and their
      conversions to punycode are considered equivalent (see Section
      5.3.3 of [RFC3987]).

      Characters in other components that are excluded by the syntax
      defined above MUST be converted from Unicode to ASCII by first
      encoding the characters as UTF-8 and then replacing the
      corresponding bytes using their percent-encoded form as defined in
      the URI [RFC3986] and IRI [RFC3987] specifications.

   Applications/protocols that use this URI scheme name
      WebSocket Protocol over TLS

   Interoperability considerations
      Use of WebSocket requires use of HTTP version 1.1 or higher.

   Security considerations
      See "Security Considerations" section.

   Contact
      HYBI WG <hybi@ietf.org>

   Author/Change controller
      IETF <iesg@ietf.org>

   References
      RFC 6455

</span></pre> 
   <h3>11.2. Registration of the "WebSocket" HTTP Upgrade Keyword</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   This section defines a keyword registered in the HTTP Upgrade Tokens
   Registry as per RFC 2817 [RFC2817].

   Name of token
      WebSocket

   Author/Change controller
      IETF <iesg@ietf.org>





<span class="grey">Fette & Melnikov             Standards Track                   [Page 56]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   Contact
      HYBI <hybi@ietf.org>

   References
      RFC 6455

</span></pre> 
   <h3>11.3. Registration of New HTTP Header Fields</h3> 
   <h4>11.3.1. Sec-WebSocket-Key</h4> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h4">

   This section describes a header field registered in the Permanent
   Message Header Field Names registry [RFC3864].

   Header field name
      Sec-WebSocket-Key

   Applicable protocol
      http

   Status
      standard

   Author/Change controller
      IETF

   Specification document(s)
      RFC 6455

   Related information
      This header field is only used for WebSocket opening handshake.

   The |Sec-WebSocket-Key| header field is used in the WebSocket opening
   handshake.  It is sent from the client to the server to provide part
   of the information used by the server to prove that it received a
   valid WebSocket opening handshake.  This helps ensure that the server
   does not accept connections from non-WebSocket clients (e.g., HTTP
   clients) that are being abused to send data to unsuspecting WebSocket
   servers.

   The |Sec-WebSocket-Key| header field MUST NOT appear more than once
   in an HTTP request.










<span class="grey">Fette & Melnikov             Standards Track                   [Page 57]</span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


</span></pre> 
   <h4>11.3.2. Sec-WebSocket-Extensions</h4> 
   <pre class="newpage"><span class="grey"><span class="h4">

   This section describes a header field for registration in the
   Permanent Message Header Field Names registry [RFC3864].

   Header field name
      Sec-WebSocket-Extensions

   Applicable protocol
      http

   Status
      standard

   Author/Change controller
      IETF

   Specification document(s)
      RFC 6455

   Related information
      This header field is only used for WebSocket opening handshake.

   The |Sec-WebSocket-Extensions| header field is used in the WebSocket
   opening handshake.  It is initially sent from the client to the
   server, and then subsequently sent from the server to the client, to
   agree on a set of protocol-level extensions to use for the duration
   of the connection.

   The |Sec-WebSocket-Extensions| header field MAY appear multiple times
   in an HTTP request (which is logically the same as a single
   |Sec-WebSocket-Extensions| header field that contains all values.
   However, the |Sec-WebSocket-Extensions| header field MUST NOT appear
   more than once in an HTTP response.

</span></span></pre> 
   <h4>11.3.3. Sec-WebSocket-Accept</h4> 
   <pre class="newpage"><span class="grey"><span class="h4"><span class="h4">

   This section describes a header field registered in the Permanent
   Message Header Field Names registry [RFC3864].

   Header field name
      Sec-WebSocket-Accept

   Applicable protocol
      http

   Status
      standard



<span class="grey">Fette & Melnikov             Standards Track                   [Page 58]</span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   Author/Change controller
      IETF

   Specification document(s)
      RFC 6455

   Related information
      This header field is only used for the WebSocket opening
      handshake.

   The |Sec-WebSocket-Accept| header field is used in the WebSocket
   opening handshake.  It is sent from the server to the client to
   confirm that the server is willing to initiate the WebSocket
   connection.

   The |Sec-WebSocket-Accept| header MUST NOT appear more than once in
   an HTTP response.

</span></pre> 
   <h4>11.3.4. Sec-WebSocket-Protocol</h4> 
   <pre class="newpage"><span class="grey"><span class="h4">

   This section describes a header field registered in the Permanent
   Message Header Field Names registry [RFC3864].

   Header field name
      Sec-WebSocket-Protocol

   Applicable protocol
      http

   Status
      standard

   Author/Change controller
      IETF

   Specification document(s)
      RFC 6455

   Related information
      This header field is only used for the WebSocket opening
      handshake.

   The |Sec-WebSocket-Protocol| header field is used in the WebSocket
   opening handshake.  It is sent from the client to the server and back
   from the server to the client to confirm the subprotocol of the
   connection.  This enables scripts to both select a subprotocol and be
   sure that the server agreed to serve that subprotocol.




<span class="grey">Fette & Melnikov             Standards Track                   [Page 59]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   The |Sec-WebSocket-Protocol| header field MAY appear multiple times
   in an HTTP request (which is logically the same as a single
   |Sec-WebSocket-Protocol| header field that contains all values).
   However, the |Sec-WebSocket-Protocol| header field MUST NOT appear
   more than once in an HTTP response.

</span></pre> 
   <h4>11.3.5. Sec-WebSocket-Version</h4> 
   <pre class="newpage"><span class="grey"><span class="h4">

   This section describes a header field registered in the Permanent
   Message Header Field Names registry [RFC3864].

   Header field name
      Sec-WebSocket-Version

   Applicable protocol
      http

   Status
      standard

   Author/Change controller
      IETF

   Specification document(s)
      RFC 6455

   Related information
      This header field is only used for the WebSocket opening
      handshake.

   The |Sec-WebSocket-Version| header field is used in the WebSocket
   opening handshake.  It is sent from the client to the server to
   indicate the protocol version of the connection.  This enables
   servers to correctly interpret the opening handshake and subsequent
   data being sent from the data, and close the connection if the server
   cannot interpret that data in a safe manner.  The |Sec-WebSocket-
   Version| header field is also sent from the server to the client on
   WebSocket handshake error, when the version received from the client
   does not match a version understood by the server.  In such a case,
   the header field includes the protocol version(s) supported by the
   server.

   Note that there is no expectation that higher version numbers are
   necessarily backward compatible with lower version numbers.







<span class="grey">Fette & Melnikov             Standards Track                   [Page 60]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   The |Sec-WebSocket-Version| header field MAY appear multiple times in
   an HTTP response (which is logically the same as a single
   |Sec-WebSocket-Version| header field that contains all values).
   However, the |Sec-WebSocket-Version| header field MUST NOT appear
   more than once in an HTTP request.

</span></pre> 
   <h3>11.4. WebSocket Extension Name Registry</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   This specification creates a new IANA registry for WebSocket
   Extension names to be used with the WebSocket Protocol in accordance
   with the principles set out in RFC 5226 [RFC5226].

   As part of this registry, IANA maintains the following information:

   Extension Identifier
      The identifier of the extension, as will be used in the
      |Sec-WebSocket-Extensions| header field registered in
      Section 11.3.2 of this specification.  The value must conform to
      the requirements for an extension-token as defined in Section 9.1
      of this specification.

   Extension Common Name
      The name of the extension, as the extension is generally referred
      to.

   Extension Definition
      A reference to the document in which the extension being used with
      the WebSocket Protocol is defined.

   Known Incompatible Extensions
      A list of extension identifiers with which this extension is known
      to be incompatible.

   WebSocket Extension names are to be subject to the "First Come First
   Served" IANA registration policy [RFC5226].

   There are no initial values in this registry.

</span></span></pre> 
   <h3>11.5. WebSocket Subprotocol Name Registry</h3> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h3">

   This specification creates a new IANA registry for WebSocket
   Subprotocol names to be used with the WebSocket Protocol in
   accordance with the principles set out in RFC 5226 [RFC5226].








<span class="grey">Fette & Melnikov             Standards Track                   [Page 61]</span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   As part of this registry, IANA maintains the following information:

   Subprotocol Identifier
      The identifier of the subprotocol, as will be used in the
      |Sec-WebSocket-Protocol| header field registered in Section 11.3.4
      of this specification.  The value must conform to the requirements
      given in item 10 of Section 4.1 of this specification -- namely,
      the value must be a token as defined by RFC 2616 [RFC2616].

   Subprotocol Common Name
      The name of the subprotocol, as the subprotocol is generally
      referred to.

   Subprotocol Definition
      A reference to the document in which the subprotocol being used
      with the WebSocket Protocol is defined.

   WebSocket Subprotocol names are to be subject to the "First Come
   First Served" IANA registration policy [RFC5226].

</span></pre> 
   <h3>11.6. WebSocket Version Number Registry</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   This specification creates a new IANA registry for WebSocket Version
   Numbers to be used with the WebSocket Protocol in accordance with the
   principles set out in RFC 5226 [RFC5226].

   As part of this registry, IANA maintains the following information:

   Version Number
      The version number to be used in the |Sec-WebSocket-Version| is
      specified in Section 4.1 of this specification.  The value must be
      a non-negative integer in the range between 0 and 255 (inclusive).

   Reference
      The RFC requesting a new version number or a draft name with
      version number (see below).

   Status
      Either "Interim" or "Standard".  See below for description.

   A version number is designated as either "Interim" or "Standard".

   A "Standard" version number is documented in an RFC and used to
   identify a major, stable version of the WebSocket protocol, such as
   the version defined by this RFC.  "Standard" version numbers are
   subject to the "IETF Review" IANA registration policy [RFC5226].





<span class="grey">Fette & Melnikov             Standards Track                   [Page 62]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   An "Interim" version number is documented in an Internet-Draft and
   used to help implementors identify and interoperate with deployed
   versions of the WebSocket protocol, such as versions developed before
   the publication of this RFC.  "Interim" version numbers are subject
   to the "Expert Review" IANA registration policy [RFC5226], with the
   chairs of the HYBI Working Group (or, if the working group closes,
   the Area Directors for the IETF Applications Area) being the initial
   Designated Experts.

   IANA has added initial values to the registry as follows.

   +--------+-----------------------------------------+----------+
   |Version |                Reference                |  Status  |
   | Number |                                         |          |
   +--------+-----------------------------------------+----------+
   | 0      + draft-ietf-hybi-thewebsocketprotocol-00 | Interim  |
   +--------+-----------------------------------------+----------+
   | 1      + draft-ietf-hybi-thewebsocketprotocol-01 | Interim  |
   +--------+-----------------------------------------+----------+
   | 2      + draft-ietf-hybi-thewebsocketprotocol-02 | Interim  |
   +--------+-----------------------------------------+----------+
   | 3      + draft-ietf-hybi-thewebsocketprotocol-03 | Interim  |
   +--------+-----------------------------------------+----------+
   | 4      + draft-ietf-hybi-thewebsocketprotocol-04 | Interim  |
   +--------+-----------------------------------------+----------+
   | 5      + draft-ietf-hybi-thewebsocketprotocol-05 | Interim  |
   +--------+-----------------------------------------+----------+
   | 6      + draft-ietf-hybi-thewebsocketprotocol-06 | Interim  |
   +--------+-----------------------------------------+----------+
   | 7      + draft-ietf-hybi-thewebsocketprotocol-07 | Interim  |
   +--------+-----------------------------------------+----------+
   | 8      + draft-ietf-hybi-thewebsocketprotocol-08 | Interim  |
   +--------+-----------------------------------------+----------+
   | 9      +                Reserved                 |          |
   +--------+-----------------------------------------+----------+
   | 10     +                Reserved                 |          |
   +--------+-----------------------------------------+----------+
   | 11     +                Reserved                 |          |
   +--------+-----------------------------------------+----------+
   | 12     +                Reserved                 |          |
   +--------+-----------------------------------------+----------+
   | 13     +                RFC 6455                 | Standard |
   +--------+-----------------------------------------+----------+








<span class="grey">Fette & Melnikov             Standards Track                   [Page 63]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


</span></pre> 
   <h3>11.7. WebSocket Close Code Number Registry</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   This specification creates a new IANA registry for WebSocket
   Connection Close Code Numbers in accordance with the principles set
   out in RFC 5226 [RFC5226].

   As part of this registry, IANA maintains the following information:

   Status Code
      The Status Code denotes a reason for a WebSocket connection
      closure as per Section 7.4 of this document.  The status code is
      an integer number between 1000 and 4999 (inclusive).

   Meaning
      The meaning of the status code.  Each status code has to have a
      unique meaning.

   Contact
      A contact for the entity reserving the status code.

   Reference
      The stable document requesting the status codes and defining their
      meaning.  This is required for status codes in the range 1000-2999
      and recommended for status codes in the range 3000-3999.

   WebSocket Close Code Numbers are subject to different registration
   requirements depending on their range.  Requests for status codes for
   use by this protocol and its subsequent versions or extensions are
   subject to any one of the "Standards Action", "Specification
   Required" (which implies "Designated Expert"), or "IESG Review" IANA
   registration policies and should be granted in the range 1000-2999.
   Requests for status codes for use by libraries, frameworks, and
   applications are subject to the "First Come First Served" IANA
   registration policy and should be granted in the range 3000-3999.
   The range of status codes from 4000-4999 is designated for Private
   Use.  Requests should indicate whether they are requesting status
   codes for use by the WebSocket Protocol (or a future version of the
   protocol), by extensions, or by libraries/frameworks/applications.













<span class="grey">Fette & Melnikov             Standards Track                   [Page 64]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   IANA has added initial values to the registry as follows.

     |Status Code | Meaning         | Contact       | Reference |
    -+------------+-----------------+---------------+-----------|
     | 1000       | Normal Closure  | hybi@ietf.org | RFC 6455  |
    -+------------+-----------------+---------------+-----------|
     | 1001       | Going Away      | hybi@ietf.org | RFC 6455  |
    -+------------+-----------------+---------------+-----------|
     | 1002       | Protocol error  | hybi@ietf.org | RFC 6455  |
    -+------------+-----------------+---------------+-----------|
     | 1003       | Unsupported Data| hybi@ietf.org | RFC 6455  |
    -+------------+-----------------+---------------+-----------|
     | 1004       | ---Reserved---- | hybi@ietf.org | RFC 6455  |
    -+------------+-----------------+---------------+-----------|
     | 1005       | No Status Rcvd  | hybi@ietf.org | RFC 6455  |
    -+------------+-----------------+---------------+-----------|
     | 1006       | Abnormal Closure| hybi@ietf.org | RFC 6455  |
    -+------------+-----------------+---------------+-----------|
     | 1007       | Invalid frame   | hybi@ietf.org | RFC 6455  |
     |            | payload data    |               |           |
    -+------------+-----------------+---------------+-----------|
     | 1008       | Policy Violation| hybi@ietf.org | RFC 6455  |
    -+------------+-----------------+---------------+-----------|
     | 1009       | Message Too Big | hybi@ietf.org | RFC 6455  |
    -+------------+-----------------+---------------+-----------|
     | 1010       | Mandatory Ext.  | hybi@ietf.org | RFC 6455  |
    -+------------+-----------------+---------------+-----------|
     | 1011       | Internal Server | hybi@ietf.org | RFC 6455  |
     |            | Error           |               |           |
    -+------------+-----------------+---------------+-----------|
     | 1015       | TLS handshake   | hybi@ietf.org | RFC 6455  |
    -+------------+-----------------+---------------+-----------|

</span></pre> 
   <h3>11.8. WebSocket Opcode Registry</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   This specification creates a new IANA registry for WebSocket Opcodes
   in accordance with the principles set out in RFC 5226 [RFC5226].

   As part of this registry, IANA maintains the following information:

   Opcode
      The opcode denotes the frame type of the WebSocket frame, as
      defined in Section 5.2.  The opcode is an integer number between 0
      and 15, inclusive.

   Meaning
      The meaning of the opcode value.




<span class="grey">Fette & Melnikov             Standards Track                   [Page 65]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   Reference
      The specification requesting the opcode.

   WebSocket Opcode numbers are subject to the "Standards Action" IANA
   registration policy [RFC5226].

   IANA has added initial values to the registry as follows.

     |Opcode  | Meaning                             | Reference |
    -+--------+-------------------------------------+-----------|
     | 0      | Continuation Frame                  | RFC 6455  |
    -+--------+-------------------------------------+-----------|
     | 1      | Text Frame                          | RFC 6455  |
    -+--------+-------------------------------------+-----------|
     | 2      | Binary Frame                        | RFC 6455  |
    -+--------+-------------------------------------+-----------|
     | 8      | Connection Close Frame              | RFC 6455  |
    -+--------+-------------------------------------+-----------|
     | 9      | Ping Frame                          | RFC 6455  |
    -+--------+-------------------------------------+-----------|
     | 10     | Pong Frame                          | RFC 6455  |
    -+--------+-------------------------------------+-----------|

</span></pre> 
   <h3>11.9. WebSocket Framing Header Bits Registry</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   This specification creates a new IANA registry for WebSocket Framing
   Header Bits in accordance with the principles set out in RFC 5226
   [RFC5226].  This registry controls assignment of the bits marked
   RSV1, RSV2, and RSV3 in Section 5.2.

   These bits are reserved for future versions or extensions of this
   specification.

   WebSocket Framing Header Bits assignments are subject to the
   "Standards Action" IANA registration policy [RFC5226].

</span></span></pre> 
   <h2>12. Using the WebSocket Protocol from Other Specifications</h2> 
   <pre class="newpage"><span class="grey"><span class="h3"><span class="h2">

   The WebSocket Protocol is intended to be used by another
   specification to provide a generic mechanism for dynamic author-
   defined content, e.g., in a specification defining a scripted API.

   Such a specification first needs to _Establish a WebSocket
   Connection_, providing that algorithm with:

   o  The destination, consisting of a /host/ and a /port/.





<span class="grey">Fette & Melnikov             Standards Track                   [Page 66]</span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   o  A /resource name/, which allows for multiple services to be
      identified at one host and port.

   o  A /secure/ flag, which is true if the connection is to be
      encrypted and false otherwise.

   o  An ASCII serialization of an origin [RFC6454] that is being made
      responsible for the connection.

   o  Optionally, a string identifying a protocol that is to be layered
      over the WebSocket connection.

   The /host/, /port/, /resource name/, and /secure/ flag are usually
   obtained from a URI using the steps to parse a WebSocket URI's
   components.  These steps fail if the URI does not specify a
   WebSocket.

   If at any time the connection is to be closed, then the specification
   needs to use the _Close the WebSocket Connection_ algorithm
   (Section 7.1.1).

   Section 7.1.4 defines when _The WebSocket Connection is Closed_.

   While a connection is open, the specification will need to handle the
   cases when _A WebSocket Message Has Been Received_ (Section 6.2).

   To send some data /data/ to an open connection, the specification
   needs to _Send a WebSocket Message_ (Section 6.1).

</span></pre> 
   <h2>13. Acknowledgements</h2> 
   <pre class="newpage"><span class="grey"><span class="h2">

   Special thanks are due to Ian Hickson, who was the original author
   and editor of this protocol.  The initial design of this
   specification benefitted from the participation of many people in the
   WHATWG and WHATWG mailing list.  Contributions to that specification
   are not tracked by section, but a list of all who contributed to that
   specification is given in the WHATWG HTML specification at
   http://whatwg.org/html5.

   Special thanks also to John Tamplin for providing a significant
   amount of text for the "Data Framing" section of this specification.

   Special thanks also to Adam Barth for providing a significant amount
   of text and background research for the "Data Masking" section of
   this specification.






<span class="grey">Fette & Melnikov             Standards Track                   [Page 67]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   Special thanks to Lisa Dusseault for the Apps Area review (and for
   helping to start this work), Richard Barnes for the Gen-Art review,
   and Magnus Westerlund for the Transport Area Review.  Special thanks
   to HYBI WG past and present WG chairs who tirelessly worked behind
   the scene to move this work toward completion: Joe Hildebrand,
   Salvatore Loreto, and Gabriel Montenegro.  And last but not least,
   special thank you to the responsible Area Director Peter Saint-Andre.

   Thank you to the following people who participated in discussions on
   the HYBI WG mailing list and contributed ideas and/or provided
   detailed reviews (the list is likely to be incomplete): Greg Wilkins,
   John Tamplin, Willy Tarreau, Maciej Stachowiak, Jamie Lokier, Scott
   Ferguson, Bjoern Hoehrmann, Julian Reschke, Dave Cridland, Andy
   Green, Eric Rescorla, Inaki Baz Castillo, Martin Thomson, Roberto
   Peon, Patrick McManus, Zhong Yu, Bruce Atherton, Takeshi Yoshino,
   Martin J. Duerst, James Graham, Simon Pieters, Roy T. Fielding,
   Mykyta Yevstifeyev, Len Holgate, Paul Colomiets, Piotr Kulaga, Brian
   Raymor, Jan Koehler, Joonas Lehtolahti, Sylvain Hellegouarch, Stephen
   Farrell, Sean Turner, Pete Resnick, Peter Thorson, Joe Mason, John
   Fallows, and Alexander Philippou.  Note that people listed above
   didn't necessarily endorse the end result of this work.

</span></pre> 
   <h2>14. References</h2> 
   <h3>14.1. Normative References</h3> 
   <pre class="newpage"><span class="grey"><span class="h2"><span class="h3">

   [ANSI.X3-4.1986]
              American National Standards Institute, "Coded Character
              Set - 7-bit American Standard Code for Information
              Interchange", ANSI X3.4, 1986.

   [FIPS.180-3]
              National Institute of Standards and Technology, "Secure
              Hash Standard", FIPS PUB 180-3, October 2008,
              <http://csrc.nist.gov/publications/fips/fips180-3/
              fips180-3_final.pdf>.

   [RFC1928]  Leech, M., Ganis, M., Lee, Y., Kuris, R., Koblas, D., and
              L. Jones, "SOCKS Protocol Version 5", RFC 1928,
              March 1996.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [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.




<span class="grey">Fette & Melnikov             Standards Track                   [Page 68]</span></span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   [RFC2817]  Khare, R. and S. Lawrence, "Upgrading to TLS Within
              HTTP/1.1", RFC 2817, May 2000.

   [RFC2818]  Rescorla, E., "HTTP Over TLS", RFC 2818, May 2000.

   [RFC3629]  Yergeau, F., "UTF-8, a transformation format of ISO
              10646", STD 63, RFC 3629, November 2003.

   [RFC3864]  Klyne, G., Nottingham, M., and J. Mogul, "Registration
              Procedures for Message Header Fields", BCP 90, RFC 3864,
              September 2004.

   [RFC3986]  Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
              Resource Identifier (URI): Generic Syntax", STD 66,
              RFC 3986, January 2005.

   [RFC3987]  Duerst, M. and M. Suignard, "Internationalized Resource
              Identifiers (IRIs)", RFC 3987, January 2005.

   [RFC4086]  Eastlake, D., Schiller, J., and S. Crocker, "Randomness
              Requirements for Security", BCP 106, RFC 4086, June 2005.

   [RFC4648]  Josefsson, S., "The Base16, Base32, and Base64 Data
              Encodings", RFC 4648, October 2006.

   [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.

   [RFC6066]  Eastlake, D., "Transport Layer Security (TLS) Extensions:
              Extension Definitions", RFC 6066, January 2011.

   [RFC6454]  Barth, A., "The Web Origin Concept", RFC 6454,
              December 2011.

</span></pre> 
   <h3>14.2. Informative References</h3> 
   <pre class="newpage"><span class="grey"><span class="h3">

   [RFC4122]  Leach, P., Mealling, M., and R. Salz, "A Universally
              Unique IDentifier (UUID) URN Namespace", RFC 4122,
              July 2005.





<span class="grey">Fette & Melnikov             Standards Track                   [Page 69]</span></span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


   [RFC4270]  Hoffman, P. and B. Schneier, "Attacks on Cryptographic
              Hashes in Internet Protocols", RFC 4270, November 2005.

   [RFC5321]  Klensin, J., "Simple Mail Transfer Protocol", RFC 5321,
              October 2008.

   [RFC6202]  Loreto, S., Saint-Andre, P., Salsano, S., and G. Wilkins,
              "Known Issues and Best Practices for the Use of Long
              Polling and Streaming in Bidirectional HTTP", RFC 6202,
              April 2011.

   [RFC6265]  Barth, A., "HTTP State Management Mechanism", RFC 6265,
              April 2011.

   [TALKING]  Huang, L-S., Chen, E., Barth, A., Rescorla, E., and C.
              Jackson, "Talking to Yourself for Fun and Profit", 2010,
              <http://w2spconf.com/2011/papers/websocket.pdf>.

   [W3C.REC-wsc-ui-20100812]
              Roessler, T. and A. Saldhana, "Web Security Context: User
              Interface Guidelines", World Wide Web Consortium
              Recommendation REC-wsc-ui-20100812, August 2010,
              <http://www.w3.org/TR/2010/REC-wsc-ui-20100812/>.

              Latest version available at
              <http://www.w3.org/TR/wsc-ui/>.

   [WSAPI]    Hickson, I., "The WebSocket API", W3C Working Draft WD-
              websockets-20110929, September 2011,
              <http://www.w3.org/TR/2011/WD-websockets-20110929/>.

              Latest version available at
              <http://www.w3.org/TR/websockets/>.

   [XMLHttpRequest]
              van Kesteren, A., Ed., "XMLHttpRequest", W3C Candidate
              Recommendation CR-XMLHttpRequest-20100803, August 2010,
              <http://www.w3.org/TR/2010/CR-XMLHttpRequest-20100803/>.

              Latest version available at
              <http://www.w3.org/TR/XMLHttpRequest/>.










<span class="grey">Fette & Melnikov             Standards Track                   [Page 70]</span></span></pre> 
   <hr class="noprint"> 
   <pre class="newpage"> 
<span class="grey">RFC 6455                 The WebSocket Protocol            December 2011


Authors' Addresses

   Ian Fette
   Google, Inc.

   EMail: ifette+ietf@google.com
   URI:   http://www.ianfette.com/


   Alexey Melnikov
   Isode Ltd.
   5 Castle Business Village
   36 Station Road
   Hampton, Middlesex  TW12 2BX
   UK

   EMail: Alexey.Melnikov@isode.com











</span></pre> 
  </div> 
 </div> 
</div>
                            </div>
                        </div>
                    </div>
                    <!--PC和WAP自适应版-->
                    <div id="SOHUCS" sid="1174685424048726016"></div>
                    <script type="text/javascript" src="/views/front/js/chanyan.js"></script>
                    <!-- 文章页-底部 动态广告位 -->
                    <div class="youdao-fixed-ad" id="detail_ad_bottom"></div>
                </div>
                <div class="col-md-3">
                    <div class="row" id="ad">
                        <!-- 文章页-右侧1 动态广告位 -->
                        <div id="right-1" class="col-lg-12 col-md-12 col-sm-4 col-xs-4 ad">
                            <div class="youdao-fixed-ad" id="detail_ad_1"> </div>
                        </div>
                        <!-- 文章页-右侧2 动态广告位 -->
                        <div id="right-2" class="col-lg-12 col-md-12 col-sm-4 col-xs-4 ad">
                            <div class="youdao-fixed-ad" id="detail_ad_2"></div>
                        </div>
                        <!-- 文章页-右侧3 动态广告位 -->
                        <div id="right-3" class="col-lg-12 col-md-12 col-sm-4 col-xs-4 ad">
                            <div class="youdao-fixed-ad" id="detail_ad_3"></div>
                        </div>
                    </div>
                </div>
            </div>
        </div>
    </div>
    <div class="container">
        <h4 class="pt20 mb15 mt0 border-top">你可能感兴趣的:(The WebSocket Protocol)</h4>
        <div id="paradigm-article-related">
            <div class="recommend-post mb30">
                <ul class="widget-links">
                    <li><a href="/article/1835393400380157952.htm"
                           title="VUE3 + xterm + nestjs实现web远程终端 或 连接开启SSH登录的路由器和交换机。" target="_blank">VUE3 + xterm + nestjs实现web远程终端 或 连接开启SSH登录的路由器和交换机。</a>
                        <span class="text-muted">焚木灵</span>
<a class="tag" taget="_blank" href="/search/node.js/1.htm">node.js</a><a class="tag" taget="_blank" href="/search/vue/1.htm">vue</a>
                        <div>可远程连接系统终端或开启SSH登录的路由器和交换机。相关资料：xtermjs/xterm.js:Aterminalfortheweb(github.com)后端实现(NestJS)：1、安装依赖：npminstallnode-ssh@nestjs/websockets@nestjs/platform-socket.io2、我们将创建一个名为RemoteControlModule的NestJS模块，</div>
                    </li>
                    <li><a href="/article/1835271449686929408.htm"
                           title="初识HTTP（1）" target="_blank">初识HTTP（1）</a>
                        <span class="text-muted">S1mple_easy</span>
<a class="tag" taget="_blank" href="/search/%E8%AE%A1%E7%AE%97%E6%9C%BA%E7%BD%91%E7%BB%9C%E5%AD%A6%E4%B9%A0%E7%AC%94%E8%AE%B0/1.htm">计算机网络学习笔记</a><a class="tag" taget="_blank" href="/search/http/1.htm">http</a>
                        <div>HTTP基本概念HTTP是超⽂本传输协议，也就是HyperTextTransferProtocol超文本传输协议：HTTP是一个在计算机世界⾥专⻔在两点之间传输⽂字、图⽚、⾳频、视频等超⽂本数据的约定和规范。HTTP常见的状态码具体含义常见状态码1xx提示信息，表示目前是协议处理状态，还需后续操作2xx成功，报文已经收到并被正确处理200/204/2063xx重定向，资源位置发生变动，需要客户端重</div>
                    </li>
                    <li><a href="/article/1835228079501438976.htm"
                           title="PCIe进阶之TL：Common Packet Header Fields & TLPs with Data Payloads Rules" target="_blank">PCIe进阶之TL：Common Packet Header Fields & TLPs with Data Payloads Rules</a>
                        <span class="text-muted">芯芯之火，可以燎原</span>
<a class="tag" taget="_blank" href="/search/PCIe%E8%BF%9B%E9%98%B6/1.htm">PCIe进阶</a><a class="tag" taget="_blank" href="/search/PCIe%E8%BF%9B%E9%98%B6/1.htm">PCIe进阶</a><a class="tag" taget="_blank" href="/search/%E7%A1%AC%E4%BB%B6%E5%B7%A5%E7%A8%8B/1.htm">硬件工程</a><a class="tag" taget="_blank" href="/search/%E4%BF%A1%E6%81%AF%E4%B8%8E%E9%80%9A%E4%BF%A1/1.htm">信息与通信</a>
                        <div>1TransactionLayerProtocol-PacketDefinitionTLP有四种事务类型：Memory、I/O、Configuration和Messages，两种地址格式：32bit和64bit。构成TLP时，所有标记为Reserved的字段（有时缩写为R）都必须全为0。接收者Rx必须忽略此字段中的值，PCIeSwitch必须对其进行原封不动的转发。请注意，对于某些字段，既有指定值</div>
                    </li>
                    <li><a href="/article/1835157681590071296.htm"
                           title="计算机网络基础" target="_blank">计算机网络基础</a>
                        <span class="text-muted">柒公子c</span>

                        <div>1通信协议1.1定义通信协议（communicationsprotocol）是指双方实体完成通信或服务所必须遵循的规则和约定。协议定义了数据单元使用的格式，信息单元应该包含的信息与含义，连接方式，信息发送和接收的时序，从而确保网络中数据顺利地传送到确定的地方。在计算机通信中，通信协议用于实现计算机与网络连接之间的标准，网络如果没有统一的通信协议，电脑之间的信息传递就无法识别。通信协议是指通信各方事</div>
                    </li>
                    <li><a href="/article/1835045616196087808.htm"
                           title="全新品牌升级的 Pencils Protocol 产品篇：层次化 Staking" target="_blank">全新品牌升级的 Pencils Protocol 产品篇：层次化 Staking</a>
                        <span class="text-muted">Blockchain_yumi</span>
<a class="tag" taget="_blank" href="/search/%E5%8C%BA%E5%9D%97%E9%93%BE/1.htm">区块链</a>
                        <div>PencilsProtocol是一个Scroll生态中的一个综合应用平台，在全新的品牌升级后(原为Penpad)，其在原有的LaunchPad的基础上，进一步向收益聚合器、RWA等板块进行全新的拓展。目前，PencilsProtocol生态的整体功能板块包括LaunchPad、Staking、Vault以及Shop功能，能够为Scroll生态用户提供全面的交易服务，并且这些板块与板块间，以及与其他</div>
                    </li>
                    <li><a href="/article/1834931265447489536.htm"
                           title="Protobuf学习 - 入门" target="_blank">Protobuf学习 - 入门</a>
                        <span class="text-muted">weixin_30892987</span>
<a class="tag" taget="_blank" href="/search/c%2Fc%2B%2B/1.htm">c/c++</a><a class="tag" taget="_blank" href="/search/python/1.htm">python</a><a class="tag" taget="_blank" href="/search/%E6%95%B0%E6%8D%AE%E7%BB%93%E6%9E%84%E4%B8%8E%E7%AE%97%E6%B3%95/1.htm">数据结构与算法</a>
                        <div>古之立大事者，不惟有超世之才，亦必有坚忍不拔之志--苏轼·《晁错论》从公司的项目源码中看到了这个东西，觉得挺好用的，写篇博客做下小总结。下面的操作以C++为编程语言，protoc的版本为libprotoc3.2.0。一、Protobuf？1.是什么？GoogleProtocolBuffer(简称Protobuf)是一种轻便高效的结构化数据存储格式，平台无关、语言无关、可扩展，可用于通讯协议和数据存</div>
                    </li>
                    <li><a href="/article/1834907690032263168.htm"
                           title="【保姆级】Protobuf详解及入门指南" target="_blank">【保姆级】Protobuf详解及入门指南</a>
                        <span class="text-muted">AQin1012</span>
<a class="tag" taget="_blank" href="/search/Java/1.htm">Java</a><a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C/1.htm">网络</a><a class="tag" taget="_blank" href="/search/protobuf/1.htm">protobuf</a><a class="tag" taget="_blank" href="/search/%E5%BA%8F%E5%88%97%E5%8C%96/1.htm">序列化</a><a class="tag" taget="_blank" href="/search/%E4%BA%8C%E8%BF%9B%E5%88%B6%E5%8D%8F%E8%AE%AE/1.htm">二进制协议</a><a class="tag" taget="_blank" href="/search/%E5%8D%8F%E8%AE%AE/1.htm">协议</a><a class="tag" taget="_blank" href="/search/Java/1.htm">Java</a>
                        <div>目录Protobuf概述什么是Protobuf为什么要使用ProtobufProtobuf实战环境配置创建文件解析/封装数据附录AQin.proto完整代码Protobuf概述什么是ProtobufProtobuf（ProtocolBuffers）协议Protobuf是一种由Google开发的二进制序列化格式和相关的技术，它用于高效地序列化和反序列化结构化数据，通常用于网络通信、数据存储等场景为什</div>
                    </li>
                    <li><a href="/article/1834877314052222976.htm"
                           title="MySQL8.0默认TCP端口介绍" target="_blank">MySQL8.0默认TCP端口介绍</a>
                        <span class="text-muted">zxrhhm</span>
<a class="tag" taget="_blank" href="/search/tcp%2Fip/1.htm">tcp/ip</a><a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C%E5%8D%8F%E8%AE%AE/1.htm">网络协议</a><a class="tag" taget="_blank" href="/search/mysql/1.htm">mysql</a>
                        <div>1、本文内容选择题TCP/IPMySQL8.0的默认TCP端口showvariables查看总结2、选择题A、3306B、33060C、33062D、330633、TCP/IPTCP/IP（TransmissionControlProtocol/InternetProtocol，传输控制协议/网际协议）是指能够在多个不同网络间实现信息传输的协议簇。以下是对TCP/IP协议的详细解释：3.1.定义与</div>
                    </li>
                    <li><a href="/article/1834816952753811456.htm"
                           title="VB验证码短信接口插件示例" target="_blank">VB验证码短信接口插件示例</a>
                        <span class="text-muted">短信接口开发</span>

                        <div>下面为您您提供了VB6版本的验证码短信接口对接DEMO例子免费体验注册地址：http://user.ihuyi.com/?DKimmuPrivateSubForm_Load()Winsock1.Protocol=sckTCPProtocolWinsock1.RemoteHost="106.ihuyi.com"Winsock1.RemotePort=80Winsock1.ConnectDoEvent</div>
                    </li>
                    <li><a href="/article/1834784912461361152.htm"
                           title="网络学习-eNSP配置VRRP" target="_blank">网络学习-eNSP配置VRRP</a>
                        <span class="text-muted">丢爸</span>
<a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C/1.htm">网络</a><a class="tag" taget="_blank" href="/search/Linux/1.htm">Linux</a><a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C/1.htm">网络</a><a class="tag" taget="_blank" href="/search/%E5%AD%A6%E4%B9%A0/1.htm">学习</a>
                        <div>虚拟路由冗余协议(VirtualRouterRedundancyProtocol，简称VRRP)VRRP广泛应用在边缘网络中，是一种路由冗余协议，它的设计目标是支持特定情况下IP数据流量失败转移不会引起混乱，允许主机使用单路由器，以及即使在实际第一跳路由器使用失败的情形下仍能够维护路由器间的连通性。#配置三层交换机和路由器IP，实现PC1与PC2的通信#---------------------1</div>
                    </li>
                    <li><a href="/article/1834761216749301760.htm"
                           title="LSP协议：技术创新背后的团队与愿景" target="_blank">LSP协议：技术创新背后的团队与愿景</a>
                        <span class="text-muted">BTColdman1</span>
<a class="tag" taget="_blank" href="/search/%E5%8C%BA%E5%9D%97%E9%93%BE/1.htm">区块链</a>
                        <div>随着去中心化金融（DeFi）和区块链技术的迅猛发展，创新性的项目层出不穷，LSP协议（LiquiditySlicingProtocolforNodeStaking）以其独特的技术优势和市场定位迅速成为行业焦点。在这背后，项目的成功离不开核心团队的领导与执行力，特别是CEOMichaelRey的卓越贡献。突破质押流动性瓶颈LSP协议作为全球首个专注于节点质押流动性的切片协议，其独特卖点在于实现了DP</div>
                    </li>
                    <li><a href="/article/1834724779085623296.htm"
                           title="TCP实现FTP功能" target="_blank">TCP实现FTP功能</a>
                        <span class="text-muted">学不会のC</span>
<a class="tag" taget="_blank" href="/search/tcp%2Fip/1.htm">tcp/ip</a><a class="tag" taget="_blank" href="/search/linux/1.htm">linux</a><a class="tag" taget="_blank" href="/search/c%E8%AF%AD%E8%A8%80/1.htm">c语言</a><a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C/1.htm">网络</a><a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C%E5%8D%8F%E8%AE%AE/1.htm">网络协议</a>
                        <div>文件传输协议（FileTransferProtocol，FTP）是一种在网络中进行文件传输的广泛使用的标准协议。作为网络通信中的基础工具，FTP允许用户通过客户端软件与服务器进行交互，实现文件的上传、下载和其他文件操作。FTP工作在OSI模型的应用层，通常使用TCP作为其传输协议，确保数据传输的可靠性和顺序性。项目要求有服务器和客户端代码，基于TCP写的。在同一路径下，将客户端可执行代码复制到其他</div>
                    </li>
                    <li><a href="/article/1834694774003691520.htm"
                           title="Websocket及三次握手/四次挥手" target="_blank">Websocket及三次握手/四次挥手</a>
                        <span class="text-muted">小童不学前端</span>
<a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C%E9%80%9A%E4%BF%A1/1.htm">网络通信</a><a class="tag" taget="_blank" href="/search/websocket/1.htm">websocket</a><a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C%E5%8D%8F%E8%AE%AE/1.htm">网络协议</a><a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C/1.htm">网络</a>
                        <div>Websocket-实时通信文章目录前言一、目前可实现实时通信的方式1、轮询（polling）2、长轮询（longpolling）3、Websocket3.1、WebSocket心跳机制3.2、WebSocket工作原理3.3、Websocket优点3.4、HTTP和Websocket关系二、HTTP协议-TCP三次握手、四次挥手1、TCP和TCP连接2、HTTP和TCP关系3、关于TCP连接的常</div>
                    </li>
                    <li><a href="/article/1834606788662620160.htm"
                           title="Linux从入门到开发实战(C/C++)Day12-ICMP协议" target="_blank">Linux从入门到开发实战(C/C++)Day12-ICMP协议</a>
                        <span class="text-muted">黒井深</span>
<a class="tag" taget="_blank" href="/search/linux/1.htm">linux</a><a class="tag" taget="_blank" href="/search/c%E8%AF%AD%E8%A8%80/1.htm">c语言</a><a class="tag" taget="_blank" href="/search/c%2B%2B/1.htm">c++</a>
                        <div>ICMP协议：InternetControlMessageProtocol网络控制报文协议作用：用来检测网络是否畅通ping命令实现流程：1.创建socketTCP:SOCK_STREAMUDP:SOCK_DGRAMICMP:SOCK_RAWSOCK_PACKETicmp协议只有root用户可以创建2.设置套字节setsockopt3.设置接收ip4.打包准备好要发送的数据包5.发包6.收包7.解</div>
                    </li>
                    <li><a href="/article/1834577959051161600.htm"
                           title="ios runtime 简单路由" target="_blank">ios runtime 简单路由</a>
                        <span class="text-muted">Mjs</span>

                        <div>1.协议本路由通过协议方式实现首先定义一个必须实现的方法typedefvoid(^JCompletionHandler)(id_Nullableresult,NSError*_Nullableerror);@protocolJRouteHandler@required+(void)handleRequestWithParameters:(nullableJParameters)parameters</div>
                    </li>
                    <li><a href="/article/1834546925853044736.htm"
                           title="sftp上传和下载文件" target="_blank">sftp上传和下载文件</a>
                        <span class="text-muted">王小工</span>
<a class="tag" taget="_blank" href="/search/Linux/1.htm">Linux</a><a class="tag" taget="_blank" href="/search/linux/1.htm">linux</a>
                        <div>SFTP（SSHFileTransferProtocol）是一种基于SSH协议的文件传输协议，它提供了文件传输过程中的加密和验证功能，确保了数据的安全性。SFTP上传和下载文件的过程可以通过多种方式实现，包括使用SFTP客户端软件、命令行工具或编程SDK等。以下是SFTP上传和下载文件的基本步骤和方法：一、SFTP客户端软件操作软件选择与安装：FileZilla：一款广受欢迎的开源FTP客户端，支</div>
                    </li>
                    <li><a href="/article/1834535206632124416.htm"
                           title="http和https区别" target="_blank">http和https区别</a>
                        <span class="text-muted">m0_73882020</span>
<a class="tag" taget="_blank" href="/search/http/1.htm">http</a>
                        <div>HTTP（超文本传输协议）和HTTPS（安全超文本传输协议）是用于在互联网上传输数据的协议。它们之间的主要区别在于安全性：HTTP（HypertextTransferProtocol）:加密:HTTP不对传输的数据进行加密，数据在传输过程中是以明文形式进行的。这意味着任何人都可以在网络中拦截并读取数据。端口:默认使用80端口。适用场景:适用于对安全要求不高的普通网页。HTTPS（Hypertext</div>
                    </li>
                    <li><a href="/article/1834497767620308992.htm"
                           title="websocket" target="_blank">websocket</a>
                        <span class="text-muted">aabbccQBB</span>
<a class="tag" taget="_blank" href="/search/websocket/1.htm">websocket</a><a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C%E5%8D%8F%E8%AE%AE/1.htm">网络协议</a><a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C/1.htm">网络</a>
                        <div>1、概念是一种在单个TCP连接上进行全双工通信的协议。websocket使得客户端和服务器之间的数据交换变得更加简单，允许服务端主动向客户端推送数据。在WebSocketAPI中，浏览器和服务器只需要完成一次握手，两者之间就直接可以创建持久性的连接，并进行双向数据传输。单工通信：数据传输只允许在一个方向上传输，只能一方发送数据，另一方接收数据并发送。半双工：数据传输允许两个方向上的传输，但在同一时</div>
                    </li>
                    <li><a href="/article/1834471678705496064.htm"
                           title="dubbo 服务消费原理分析之服务目录" target="_blank">dubbo 服务消费原理分析之服务目录</a>
                        <span class="text-muted">DEARM LINER</span>
<a class="tag" taget="_blank" href="/search/dubbo/1.htm">dubbo</a><a class="tag" taget="_blank" href="/search/java/1.htm">java</a><a class="tag" taget="_blank" href="/search/%E6%9E%B6%E6%9E%84/1.htm">架构</a><a class="tag" taget="_blank" href="/search/%E5%90%8E%E7%AB%AF/1.htm">后端</a><a class="tag" taget="_blank" href="/search/spring/1.htm">spring</a><a class="tag" taget="_blank" href="/search/boot/1.htm">boot</a>
                        <div>文章目录前言一、RegistryDirectory1、DynamicDirectory2、RegistryProtocol.doCreateInvoker2、RegistryProtocol.subscribe3、ListenerRegistryWrapper.subscribe4、FailbackRegistry.subscribe5、ZookeeperRegistry.doSubscribe6</div>
                    </li>
                    <li><a href="/article/1834452388887490560.htm"
                           title="网络套接字编程（二）" target="_blank">网络套接字编程（二）</a>
                        <span class="text-muted">knight-n</span>
<a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C/1.htm">网络</a><a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C/1.htm">网络</a><a class="tag" taget="_blank" href="/search/linux/1.htm">linux</a><a class="tag" taget="_blank" href="/search/c%2B%2B/1.htm">c++</a>
                        <div>socket常见API创建套接字：（TCP/UDP，客户端+服务器）intsocket(intdomain,inttype,intprotocol);绑定端口号：（TCP/UDP，服务器）intbind(intsockfd,conststructsockaddr*addr,socklen_taddrlen);监听套接字：（TCP，服务器）intlisten(intsockfd,intbacklog</div>
                    </li>
                    <li><a href="/article/1834256990159859712.htm"
                           title="简述三次握手和四次断开" target="_blank">简述三次握手和四次断开</a>
                        <span class="text-muted">Will_1130</span>
<a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C/1.htm">网络</a><a class="tag" taget="_blank" href="/search/tcp%2Fip/1.htm">tcp/ip</a><a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C%E5%8D%8F%E8%AE%AE/1.htm">网络协议</a>
                        <div>在计算机网络中，TCP（TransmissionControlProtocol）是一种面向连接的协议，它通过三次握手（Three-WayHandshake）建立连接，并通过四次挥手（Four-WayWave）断开连接。下面分别简述这两个过程。三次握手（Three-WayHandshake）三次握手的过程是为了确保双方都准备好进行通信，并且可以正确地发送和接收数据。以下是三次握手的具体步骤：SYN（</div>
                    </li>
                    <li><a href="/article/1834192716821524480.htm"
                           title="ping操作" target="_blank">ping操作</a>
                        <span class="text-muted">姜太公钓鲸233</span>
<a class="tag" taget="_blank" href="/search/%E6%9C%8D%E5%8A%A1%E5%99%A8/1.htm">服务器</a><a class="tag" taget="_blank" href="/search/%E8%BF%90%E7%BB%B4/1.htm">运维</a>
                        <div>Ping（发音为“拼”）是一种广泛使用的网络诊断工具，用于测试计算机或设备在IP网络上的可达性以及测量数据包往返所需的时间。它基于ICMP（InternetControlMessageProtocol，互联网控制消息协议）来发送和接收消息，从而帮助网络管理员和用户诊断网络连接问题。Ping操作的工作原理发送Echo请求：当你在命令行或终端中执行ping命令时，计算机会向目标主机发送一个ICMPEc</div>
                    </li>
                    <li><a href="/article/1834189312451768320.htm"
                           title="分布式中间件-几个常用的消息中间件" target="_blank">分布式中间件-几个常用的消息中间件</a>
                        <span class="text-muted">问道飞鱼</span>
<a class="tag" taget="_blank" href="/search/%E5%88%86%E5%B8%83%E5%BC%8F%E6%8A%80%E6%9C%AF/1.htm">分布式技术</a><a class="tag" taget="_blank" href="/search/%E5%88%86%E5%B8%83%E5%BC%8F/1.htm">分布式</a><a class="tag" taget="_blank" href="/search/%E4%B8%AD%E9%97%B4%E4%BB%B6/1.htm">中间件</a>
                        <div>文章目录常见消息中间件1.RabbitMQ2.ApacheKafka3.RedisPub/Sub4.ActiveMQ5.AmazonSimpleNotificationService(SNS)和SimpleQueueService(SQS)6.RocketMQ差异总结消息协议1.AMQP(AdvancedMessageQueuingProtocol)2.STOMP(SimpleTextOrient</div>
                    </li>
                    <li><a href="/article/1834169896339075072.htm"
                           title="消息总线RabbitMQ" target="_blank">消息总线RabbitMQ</a>
                        <span class="text-muted">Crystalqy</span>
<a class="tag" taget="_blank" href="/search/%E5%BE%AE%E6%9C%8D%E5%8A%A1/1.htm">微服务</a><a class="tag" taget="_blank" href="/search/%E6%B6%88%E6%81%AF%E9%98%9F%E5%88%97/1.htm">消息队列</a><a class="tag" taget="_blank" href="/search/rabbitmq/1.htm">rabbitmq</a>
                        <div>RabbitMQ是实现了高级消息队列协议（AMQP）的开源消息代理软件，也称为面向消息的中间件，是一个在AMQP基础上完整的，可复用的企业消息系统。它可以用于大型软件系统各个模块之间的高效通信，支持高并发，支持可扩展。AMQP，即AdvancedMessageQueuingProtocol,一个提供统一消息服务的应用层标准高级消息队列协议,是应用层协议的一个开放标准,为面向消息的中间件设计，它定义</div>
                    </li>
                    <li><a href="/article/1834114188868415488.htm"
                           title="C#使用TCP-S7协议读写西门子PLC(四)" target="_blank">C#使用TCP-S7协议读写西门子PLC(四)</a>
                        <span class="text-muted">斯内科</span>
<a class="tag" taget="_blank" href="/search/S7/1.htm">S7</a><a class="tag" taget="_blank" href="/search/TCP/1.htm">TCP</a><a class="tag" taget="_blank" href="/search/C%23/1.htm">C#</a><a class="tag" taget="_blank" href="/search/tcp%2Fip/1.htm">tcp/ip</a><a class="tag" taget="_blank" href="/search/S7/1.htm">S7</a><a class="tag" taget="_blank" href="/search/S7%E5%8D%8F%E8%AE%AE/1.htm">S7协议</a>
                        <div>接上一篇，我们连接PLC并握手成功，并且封装生成读写PLC的命令C#使用TCP-S7协议读写西门子PLC(三)-CSDN博客这里我们进行读写基础数据类型、读取DB块的字符串、宽字符串、以及一系列连续数组。新建部分类文件SiemensS7ProtocolUtil.Integrated.cs主要方法读取任意连续字节数据,任意多个Byte读取--每次最多读取200个字节，如果超过200，自动分多次读取p</div>
                    </li>
                    <li><a href="/article/1834047881938563072.htm"
                           title="HTTPS和HTTP区别是什么？" target="_blank">HTTPS和HTTP区别是什么？</a>
                        <span class="text-muted">全真王重阳</span>
<a class="tag" taget="_blank" href="/search/%23/1.htm">#</a><a class="tag" taget="_blank" href="/search/%E8%AE%A1%E7%AE%97%E6%9C%BA%E7%BD%91%E7%BB%9C/1.htm">计算机网络</a><a class="tag" taget="_blank" href="/search/http/1.htm">http</a><a class="tag" taget="_blank" href="/search/https/1.htm">https</a><a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C%E5%8D%8F%E8%AE%AE/1.htm">网络协议</a><a class="tag" taget="_blank" href="/search/%E7%BD%91%E7%BB%9C/1.htm">网络</a><a class="tag" taget="_blank" href="/search/408/1.htm">408</a><a class="tag" taget="_blank" href="/search/%E8%AE%A1%E7%AE%97%E6%9C%BA%E7%BD%91%E7%BB%9C/1.htm">计算机网络</a>
                        <div>HTTP和HTTPS是两种协议，分别是HyperTextTransferProtocol和HyperTextTransferProtocolSecure。HTTPS还经常被称为HTTPoverSSL或者HTTPoverTSL，HTTPS经由HTTP进行通信，但利用SSL/TLS来加密数据包。他们的区别主要由以下几个方面：安全性:HTTP:HTTP是明文传输的，这意味着数据在传输过程中不加密，容易受</div>
                    </li>
                    <li><a href="/article/1833958177465528320.htm"
                           title="Http协议、TCP协议、WebSocket协议" target="_blank">Http协议、TCP协议、WebSocket协议</a>
                        <span class="text-muted">俗人浮生</span>

                        <div>网络编程大家都不陌生，但平时我们只是知道怎么用，对个中的网络协议倒是没花多少工夫去学习，所以，我们今天特意来对网络协议相关的知识加以总结记录。1）网络七层协议，从上到下依次为：7应用层6表示层5会话层4传输层3网络层2数据链路层1物理层2）Http协议对应于应用层，TCP协议对应于传输层，Http协议是建立在TCP协议基础之上的3）属于应用层的常用协议：Http（超文本传输协议）、FTP（文件传输</div>
                    </li>
                    <li><a href="/article/1833956742602518528.htm"
                           title="@rabbitlistener注解" target="_blank">@rabbitlistener注解</a>
                        <span class="text-muted">小小懒懒</span>
<a class="tag" taget="_blank" href="/search/RabbitMQ/1.htm">RabbitMQ</a>
                        <div>@RabbitListener是SpringAMQP（AdvancedMessageQueuingProtocol）框架中的一个注解，用于创建RabbitMQ消息监听器。RabbitMQ是一个开源的消息代理和队列服务器，它实现了AMQP协议，用于在分布式系统中进行消息传递。SpringAMQP是Spring框架对RabbitMQ的集成，提供了方便的消息发送和接收的功能。通过使用@RabbitLis</div>
                    </li>
                    <li><a href="/article/1833955260633280512.htm"
                           title="Protocol Buffer编译器安装" target="_blank">Protocol Buffer编译器安装</a>
                        <span class="text-muted">雪域迷影</span>

                        <div>本文翻译自ProtocolBufferCompilerInstallationProtocolBufferCompilerInstallation如何安装protocolbuffer编译器尽管不是强制性的，但gRPC应用程序通常利用ProtocolBuufer来进行服务定义和数据序列化。该站点上的大多数示例代码都使用protocolbuffer语言（proto3）的版本3。protocolbuff</div>
                    </li>
                    <li><a href="/article/1833952836283625472.htm"
                           title="ESP32鉴权 讯飞科技" target="_blank">ESP32鉴权 讯飞科技</a>
                        <span class="text-muted">qq_42949957</span>
<a class="tag" taget="_blank" href="/search/%E7%A7%91%E6%8A%80/1.htm">科技</a><a class="tag" taget="_blank" href="/search/%E8%BF%90%E7%BB%B4/1.htm">运维</a>
                        <div>#include#include#include#include#include#includeconstchar*ssid="464564";constchar*password="546456.";WebSocketsClientwebSocket;voidwebSocketEvent(WStype_ttype,uint8_t*payload,size_tlength){switch(type</div>
                    </li>
                                <li><a href="/article/68.htm"
                                       title="mongodb3.03开启认证" target="_blank">mongodb3.03开启认证</a>
                                    <span class="text-muted">21jhf</span>
<a class="tag" taget="_blank" href="/search/mongodb/1.htm">mongodb</a>
                                    <div>下载了最新mongodb3.03版本，当使用--auth 参数命令行开启mongodb用户认证时遇到很多问题，现总结如下： 
（百度上搜到的基本都是老版本的，看到db.addUser的就是，请忽略） 
Windows下我做了一个bat文件，用来启动mongodb，命令行如下： 
mongod --dbpath db\data --port 27017 --directoryperdb --logp</div>
                                </li>
                                <li><a href="/article/195.htm"
                                       title="【Spark103】Task not serializable" target="_blank">【Spark103】Task not serializable</a>
                                    <span class="text-muted">bit1129</span>
<a class="tag" taget="_blank" href="/search/Serializable/1.htm">Serializable</a>
                                    <div>Task not serializable是Spark开发过程最令人头疼的问题之一，这里记录下出现这个问题的两个实例，一个是自己遇到的，另一个是stackoverflow上看到。等有时间了再仔细探究出现Task not serialiazable的各种原因以及出现问题后如何快速定位问题的所在，至少目前阶段碰到此类问题，没有什么章法 
1. 
&nbsp; 
package spark.exampl</div>
                                </li>
                                <li><a href="/article/322.htm"
                                       title="你所熟知的 LRU(最近最少使用)" target="_blank">你所熟知的 LRU(最近最少使用)</a>
                                    <span class="text-muted">dalan_123</span>
<a class="tag" taget="_blank" href="/search/java/1.htm">java</a>
                                    <div>关于LRU这个名词在很多地方或听说，或使用，接下来看下lru缓存回收的实现 
1、大体的想法 
&nbsp; &nbsp; a、查询出最近最晚使用的项 
&nbsp; &nbsp; b、给最近的使用的项做标记 
通过使用链表就可以完成这两个操作，关于最近最少使用的项只需要返回链表的尾部；标记最近使用的项，只需要将该项移除并放置到头部，那么难点就出现 你如何能够快速在链表定位对应的该项？ 
这时候多</div>
                                </li>
                                <li><a href="/article/449.htm"
                                       title="Javascript 跨域" target="_blank">Javascript 跨域</a>
                                    <span class="text-muted">周凡杨</span>
<a class="tag" taget="_blank" href="/search/JavaScript/1.htm">JavaScript</a><a class="tag" taget="_blank" href="/search/jsonp/1.htm">jsonp</a><a class="tag" taget="_blank" href="/search/%E8%B7%A8%E5%9F%9F/1.htm">跨域</a><a class="tag" taget="_blank" href="/search/cross-domain/1.htm">cross-domain</a>
                                    <div>&nbsp; 
&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp</div>
                                </li>
                                <li><a href="/article/576.htm"
                                       title="linux下安装apache服务器" target="_blank">linux下安装apache服务器</a>
                                    <span class="text-muted">g21121</span>
<a class="tag" taget="_blank" href="/search/apache/1.htm">apache</a>
                                    <div>安装apache  
下载windows版本apache，下载地址：http://httpd.apache.org/download.cgi 
&nbsp;  1.windows下安装apache   
Windows下安装apache比较简单，注意选择路径和端口即可，这里就不再赘述了。  2.linux下安装apache：  
下载之后上传到linux的相关目录，这里指定为/home/apach</div>
                                </li>
                                <li><a href="/article/703.htm"
                                       title="FineReport的JS编辑框和URL地址栏语法简介" target="_blank">FineReport的JS编辑框和URL地址栏语法简介</a>
                                    <span class="text-muted">老A不折腾</span>
<a class="tag" taget="_blank" href="/search/finereport/1.htm">finereport</a><a class="tag" taget="_blank" href="/search/web%E6%8A%A5%E8%A1%A8/1.htm">web报表</a><a class="tag" taget="_blank" href="/search/%E6%8A%A5%E8%A1%A8%E8%BD%AF%E4%BB%B6/1.htm">报表软件</a><a class="tag" taget="_blank" href="/search/%E8%AF%AD%E6%B3%95%E6%80%BB%E7%BB%93/1.htm">语法总结</a>
                                    <div>&nbsp; JS编辑框： 
1.FineReport的js。 
作为一款BS产品，browser端的JavaScript是必不可少的。 
FineReport中的js是已经调用了finereport.js的。 
大家知道，预览报表时，报表servlet会将cpt模板转为html，在这个html的head头部中会引入FineReport的js，这个finereport.js中包含了许多内置的fun</div>
                                </li>
                                <li><a href="/article/830.htm"
                                       title="根据STATUS信息对MySQL进行优化" target="_blank">根据STATUS信息对MySQL进行优化</a>
                                    <span class="text-muted">墙头上一根草</span>
<a class="tag" taget="_blank" href="/search/status/1.htm">status</a>
                                    <div>mysql &nbsp;查看当前正在执行的操作，即正在执行的sql语句的方法为: 
&nbsp; &nbsp; &nbsp;show processlist 命令 
&nbsp; 
mysql&gt; show global status;可以列出MySQL服务器运行各种状态值，我个人较喜欢的用法是show status like '查询值%';一、慢查询mysql&gt; show variab</div>
                                </li>
                                <li><a href="/article/957.htm"
                                       title="我的spring学习笔记7-Spring的Bean配置文件给Bean定义别名" target="_blank">我的spring学习笔记7-Spring的Bean配置文件给Bean定义别名</a>
                                    <span class="text-muted">aijuans</span>
<a class="tag" taget="_blank" href="/search/Spring+3/1.htm">Spring 3</a>
                                    <div>本文介绍如何给Spring的Bean配置文件的Bean定义别名？ 
 
原始的 
 
 
&lt;bean id=&quot;business&quot; class=&quot;onlyfun.caterpillar.device.Business&quot;&gt;
    &lt;property name=&quot;writer&quot;&gt;
        &lt;ref b</div>
                                </li>
                                <li><a href="/article/1084.htm"
                                       title="高性能mysql 之 性能剖析" target="_blank">高性能mysql 之 性能剖析</a>
                                    <span class="text-muted">annan211</span>
<a class="tag" taget="_blank" href="/search/%E6%80%A7%E8%83%BD/1.htm">性能</a><a class="tag" taget="_blank" href="/search/mysql/1.htm">mysql</a><a class="tag" taget="_blank" href="/search/mysql+%E6%80%A7%E8%83%BD%E5%89%96%E6%9E%90/1.htm">mysql 性能剖析</a><a class="tag" taget="_blank" href="/search/%E5%89%96%E6%9E%90/1.htm">剖析</a>
                                    <div>

1 定义性能优化
  mysql服务器性能，此处定义为 响应时间。
  在解释性能优化之前，先来消除一个误解，很多人认为，性能优化就是降低cpu的利用率或者减少对资源的使用。
  这是一个陷阱。
  资源时用来消耗并用来工作的，所以有时候消耗更多的资源能够加快查询速度，保持cpu忙绿，这是必要的。很多时候发现
  编译进了新版本的InnoDB之后，cpu利用率上升的很厉害，这并不</div>
                                </li>
                                <li><a href="/article/1211.htm"
                                       title="主外键和索引唯一性约束" target="_blank">主外键和索引唯一性约束</a>
                                    <span class="text-muted">百合不是茶</span>
<a class="tag" taget="_blank" href="/search/%E7%B4%A2%E5%BC%95/1.htm">索引</a><a class="tag" taget="_blank" href="/search/%E5%94%AF%E4%B8%80%E6%80%A7%E7%BA%A6%E6%9D%9F/1.htm">唯一性约束</a><a class="tag" taget="_blank" href="/search/%E4%B8%BB%E5%A4%96%E9%94%AE%E7%BA%A6%E6%9D%9F/1.htm">主外键约束</a><a class="tag" taget="_blank" href="/search/%E8%81%94%E6%9C%BA%E5%88%A0%E9%99%A4/1.htm">联机删除</a>
                                    <div>目标;第一步;创建两张表 用户表和文章表 
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 第二步;发表文章 
&nbsp; 
&nbsp; 
&nbsp; 1,建表; 
    ---用户表 BlogUsers
    --userID唯一的
    --userName
    --pwd
    --sex
        

    create </div>
                                </li>
                                <li><a href="/article/1338.htm"
                                       title="线程的调度" target="_blank">线程的调度</a>
                                    <span class="text-muted">bijian1013</span>
<a class="tag" taget="_blank" href="/search/java/1.htm">java</a><a class="tag" taget="_blank" href="/search/%E5%A4%9A%E7%BA%BF%E7%A8%8B/1.htm">多线程</a><a class="tag" taget="_blank" href="/search/thread/1.htm">thread</a><a class="tag" taget="_blank" href="/search/%E7%BA%BF%E7%A8%8B%E7%9A%84%E8%B0%83%E5%BA%A6/1.htm">线程的调度</a><a class="tag" taget="_blank" href="/search/java%E5%A4%9A%E7%BA%BF%E7%A8%8B/1.htm">java多线程</a>
                                    <div>1.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Java提供一个线程调度程序来监控程序中启动后进入可运行状态的所有线程。线程调度程序按照线程的优先级决定应调度哪些线程来执行。 
&nbsp; 
2.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 多数线程的调度是抢占式的（即我想中断程序运行就中断，不需要和将被中断的程序协商） 
a)&nbsp</div>
                                </li>
                                <li><a href="/article/1465.htm"
                                       title="查看日志常用命令" target="_blank">查看日志常用命令</a>
                                    <span class="text-muted">bijian1013</span>
<a class="tag" taget="_blank" href="/search/linux/1.htm">linux</a><a class="tag" taget="_blank" href="/search/%E5%91%BD%E4%BB%A4/1.htm">命令</a><a class="tag" taget="_blank" href="/search/unix/1.htm">unix</a>
                                    <div>一.日志查找方法，可以用通配符查某台主机上的所有服务器grep &quot;关键字&quot; /wls/applogs/custom-*/error.log 
&nbsp; 
二.查看日志常用命令1.grep '关键字' error.log：在error.log中搜索'关键字'2.grep -C10 '关键字' error.log：显示关键字前后10行记录3.grep '关键字' error.l</div>
                                </li>
                                <li><a href="/article/1592.htm"
                                       title="【持久化框架MyBatis3一】MyBatis版HelloWorld" target="_blank">【持久化框架MyBatis3一】MyBatis版HelloWorld</a>
                                    <span class="text-muted">bit1129</span>
<a class="tag" taget="_blank" href="/search/helloworld/1.htm">helloworld</a>
                                    <div>MyBatis这个系列的文章，主要参考《Java Persistence with MyBatis 3》。 
&nbsp; 样例数据 
本文以MySQL数据库为例，建立一个STUDENTS表，插入两条数据，然后进行单表的增删改查 
&nbsp; 
&nbsp; 
CREATE TABLE STUDENTS
(
stud_id int(11) NOT NULL AUTO_INCREMENT,
</div>
                                </li>
                                <li><a href="/article/1719.htm"
                                       title="【Hadoop十五】Hadoop Counter" target="_blank">【Hadoop十五】Hadoop Counter</a>
                                    <span class="text-muted">bit1129</span>
<a class="tag" taget="_blank" href="/search/hadoop/1.htm">hadoop</a>
                                    <div>&nbsp; 
&nbsp;1. 只有Map任务的Map Reduce Job  
	File System Counters
		FILE: Number of bytes read=3629530
		FILE: Number of bytes written=98312
		FILE: Number of read operations=0
		FILE: Number of lar</div>
                                </li>
                                <li><a href="/article/1846.htm"
                                       title="解决Tomcat数据连接池无法释放" target="_blank">解决Tomcat数据连接池无法释放</a>
                                    <span class="text-muted">ronin47</span>
<a class="tag" taget="_blank" href="/search/tomcat+%E8%BF%9E%E6%8E%A5%E6%B1%A0%E3%80%80%E4%BC%98%E5%8C%96/1.htm">tomcat 连接池　优化</a>
                                    <div> 
 近段时间，公司的检测中心报表系统(SMC)的开发人员时不时找到我，说用户老是出现无法登录的情况。前些日子因为手头上 有Jboss集群的测试工作，发现用户不能登录时，都是在Tomcat中将这个项目Reload一下就好了，不过只是治标而已，因为大概几个小时之后又会 再次出现无法登录的情况。 
  
今天上午，开发人员小毛又找到我，要我协助将这个问题根治一下，拖太久用户难保不投诉。 
简单分析了一</div>
                                </li>
                                <li><a href="/article/1973.htm"
                                       title="java-75-二叉树两结点的最低共同父结点" target="_blank">java-75-二叉树两结点的最低共同父结点</a>
                                    <span class="text-muted">bylijinnan</span>
<a class="tag" taget="_blank" href="/search/java/1.htm">java</a>
                                    <div>
import java.util.LinkedList;
import java.util.List;

import ljn.help.*;
public class BTreeLowestParentOfTwoNodes {

	public static void main(String[] args) {
		/*
		 * node data is stored in</div>
                                </li>
                                <li><a href="/article/2100.htm"
                                       title="行业垂直搜索引擎网页抓取项目" target="_blank">行业垂直搜索引擎网页抓取项目</a>
                                    <span class="text-muted">carlwu</span>
<a class="tag" taget="_blank" href="/search/Lucene/1.htm">Lucene</a><a class="tag" taget="_blank" href="/search/Nutch/1.htm">Nutch</a><a class="tag" taget="_blank" href="/search/Heritrix/1.htm">Heritrix</a><a class="tag" taget="_blank" href="/search/Solr/1.htm">Solr</a>
                                    <div>公司有一个搜索引擎项目，希望各路高人有空来帮忙指导，谢谢！ 
这是详细需求： 
（1） 通过提供的网站地址(大概100-200个网站)，网页抓取程序能不断抓取网页和其它类型的文件（如Excel、PDF、Word、ppt及zip类型），并且程序能够根据事先提供的规则，过滤掉不相干的下载内容。 
（2） 程序能够搜索这些抓取的内容，并能对这些抓取文件按照油田名进行分类，然后放到服务器不同的目录中。 
</div>
                                </li>
                                <li><a href="/article/2227.htm"
                                       title="[通讯与服务]在总带宽资源没有大幅增加之前,不适宜大幅度降低资费" target="_blank">[通讯与服务]在总带宽资源没有大幅增加之前,不适宜大幅度降低资费</a>
                                    <span class="text-muted">comsci</span>
<a class="tag" taget="_blank" href="/search/%E8%B5%84%E6%BA%90/1.htm">资源</a>
                                    <div> 
 
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 降低通讯服务资费，就意味着有更多的用户进入，就意味着通讯服务提供商要接待和服务更多的用户，在总体运维成本没有由于技术升级而大幅下降的情况下，这种降低资费的行为将导致每个用户的平均带宽不断下降，而享受到的服务质量也在下降，这对用户和服务商都是不利的。。。。。。。。 
 
 
&nbsp;&nbsp;&nbsp;&nbsp;&nbs</div>
                                </li>
                                <li><a href="/article/2354.htm"
                                       title="Java时区转换及时间格式" target="_blank">Java时区转换及时间格式</a>
                                    <span class="text-muted">Cwind</span>
<a class="tag" taget="_blank" href="/search/java/1.htm">java</a>
                                    <div>本文介绍Java API 中 Date, Calendar, TimeZone和DateFormat的使用，以及不同时区时间相互转化的方法和原理。 
&nbsp; 
问题描述： 
向处于不同时区的服务器发请求时需要考虑时区转换的问题。譬如，服务器位于东八区（北京时间，GMT+8:00），而身处东四区的用户想要查询当天的销售记录。则需把东四区的“今天”这个时间范围转换为服务器所在时区的时间范围。 
</div>
                                </li>
                                <li><a href="/article/2481.htm"
                                       title="readonly,只读，不可用" target="_blank">readonly,只读，不可用</a>
                                    <span class="text-muted">dashuaifu</span>
<a class="tag" taget="_blank" href="/search/js/1.htm">js</a><a class="tag" taget="_blank" href="/search/jsp/1.htm">jsp</a><a class="tag" taget="_blank" href="/search/disable/1.htm">disable</a><a class="tag" taget="_blank" href="/search/readOnly/1.htm">readOnly</a><a class="tag" taget="_blank" href="/search/readOnly/1.htm">readOnly</a>
                                    <div>readOnly 和 readonly 不同，在做js开发时一定要注意函数大小写和jsp黄线的警告！！！我就经历过这么一件事： 
使用readOnly在某些浏览器或同一浏览器不同版本有的可以实现“只读”功能，有的就不行，而且函数readOnly有黄线警告！！！就这样被折磨了不短时间！！！（期间使用过disable函数，但是发现disable函数之后后台接收不到前台的的数据！！！） 
&nbsp; </div>
                                </li>
                                <li><a href="/article/2608.htm"
                                       title="LABjs、RequireJS、SeaJS 介绍" target="_blank">LABjs、RequireJS、SeaJS 介绍</a>
                                    <span class="text-muted">dcj3sjt126com</span>
<a class="tag" taget="_blank" href="/search/js/1.htm">js</a><a class="tag" taget="_blank" href="/search/Web/1.htm">Web</a>
                                    <div>LABjs 的核心是 LAB（Loading and Blocking）：Loading 指异步并行加载，Blocking 是指同步等待执行。LABjs 通过优雅的语法（script 和 wait）实现了这两大特性，核心价值是性能优化。LABjs 是一个文件加载器。RequireJS 和 SeaJS 则是模块加载器，倡导的是一种模块化开发理念，核心价值是让 JavaScript 的模块化开发变得更</div>
                                </li>
                                <li><a href="/article/2735.htm"
                                       title="[应用结构]入口脚本" target="_blank">[应用结构]入口脚本</a>
                                    <span class="text-muted">dcj3sjt126com</span>
<a class="tag" taget="_blank" href="/search/PHP/1.htm">PHP</a><a class="tag" taget="_blank" href="/search/yii2/1.htm">yii2</a>
                                    <div>入口脚本 
入口脚本是应用启动流程中的第一环，一个应用（不管是网页应用还是控制台应用）只有一个入口脚本。终端用户的请求通过入口脚本实例化应用并将将请求转发到应用。 
Web 应用的入口脚本必须放在终端用户能够访问的目录下，通常命名为&nbsp;index.php，也可以使用 Web 服务器能定位到的其他名称。 
控制台应用的入口脚本一般在应用根目录下命名为&nbsp;yii（后缀为.php），该文</div>
                                </li>
                                <li><a href="/article/2862.htm"
                                       title="haoop shell命令" target="_blank">haoop shell命令</a>
                                    <span class="text-muted">eksliang</span>
<a class="tag" taget="_blank" href="/search/hadoop/1.htm">hadoop</a><a class="tag" taget="_blank" href="/search/hadoop+shell/1.htm">hadoop shell</a>
                                    <div> 
 cat 
 chgrp 
 chmod 
 chown 
 copyFromLocal 
 copyToLocal 
 cp 
 du 
 dus 
 expunge 
 get 
 getmerge 
 ls 
 lsr 
 mkdir 
 movefromLocal 
 mv 
 put 
 rm 
 rmr 
 setrep 
 stat 
 tail 
 test 
 text 
 </div>
                                </li>
                                <li><a href="/article/2989.htm"
                                       title="MultiStateView不同的状态下显示不同的界面" target="_blank">MultiStateView不同的状态下显示不同的界面</a>
                                    <span class="text-muted">gundumw100</span>
<a class="tag" taget="_blank" href="/search/android/1.htm">android</a>
                                    <div>只要将指定的view放在该控件里面，可以该view在不同的状态下显示不同的界面，这对ListView很有用，比如加载界面，空白界面，错误界面。而且这些见面由你指定布局，非常灵活。 
 
PS：ListView虽然可以设置一个EmptyView，但使用起来不方便，不灵活，有点累赘。 
 
 

&lt;com.kennyc.view.MultiStateView xmlns:android=&qu</div>
                                </li>
                                <li><a href="/article/3116.htm"
                                       title="jQuery实现页面内锚点平滑跳转" target="_blank">jQuery实现页面内锚点平滑跳转</a>
                                    <span class="text-muted">ini</span>
<a class="tag" taget="_blank" href="/search/JavaScript/1.htm">JavaScript</a><a class="tag" taget="_blank" href="/search/html/1.htm">html</a><a class="tag" taget="_blank" href="/search/jquery/1.htm">jquery</a><a class="tag" taget="_blank" href="/search/html5/1.htm">html5</a><a class="tag" taget="_blank" href="/search/css/1.htm">css</a>
                                    <div>平时我们做导航滚动到内容都是通过锚点来做，刷的一下就直接跳到内容了，没有一丝的滚动效果，而且 url 链接最后会有“小尾巴”，就像#keleyi，今天我就介绍一款 jquery 做的滚动的特效，既可以设置滚动速度，又可以在 url 链接上没有“小尾巴”。 
&nbsp; 
效果体验：http://keleyi.com/keleyi/phtml/jqtexiao/37.htmHTML文件代码： 
&</div>
                                </li>
                                <li><a href="/article/3243.htm"
                                       title="kafka offset迁移" target="_blank">kafka offset迁移</a>
                                    <span class="text-muted">kane_xie</span>
<a class="tag" taget="_blank" href="/search/kafka/1.htm">kafka</a>
                                    <div>在早前的kafka版本中（0.8.0），offset是被存储在zookeeper中的。 
&nbsp; 
到当前版本（0.8.2）为止，kafka同时支持offset存储在zookeeper和offset manager（broker）中。 
&nbsp; 
从官方的说明来看，未来offset的zookeeper存储将会被弃用。因此现有的基于kafka的项目如果今后计划保持更新的话，可以考虑在合适</div>
                                </li>
                                <li><a href="/article/3370.htm"
                                       title="android > 搭建 cordova 环境" target="_blank">android > 搭建 cordova 环境</a>
                                    <span class="text-muted">mft8899</span>
<a class="tag" taget="_blank" href="/search/android/1.htm">android</a>
                                    <div>&nbsp; 
1 , 安装 node.js &nbsp; 
&nbsp; &nbsp; &nbsp;http://nodejs.org 
&nbsp; &nbsp;&nbsp; 
node -v &nbsp; 查看版本 
&nbsp; 
2, 安装 npm 
&nbsp; 
可以先从 &nbsp;https://github.com/isaacs/npm/tags&nbsp; 下载 源码 解压到</div>
                                </li>
                                <li><a href="/article/3497.htm"
                                       title="java封装的比较器，比较是否全相同，获取不同字段名字" target="_blank">java封装的比较器，比较是否全相同，获取不同字段名字</a>
                                    <span class="text-muted">qifeifei</span>

                                    <div>&nbsp;非常实用的java比较器，贴上代码： 
import java.util.HashSet;
import java.util.List;
import java.util.Set;

import net.sf.json.JSONArray;
import net.sf.json.JSONObject;
import net.sf.json.JsonConfig;

i</div>
                                </li>
                                <li><a href="/article/3624.htm"
                                       title="记录一些函数用法" target="_blank">记录一些函数用法</a>
                                    <span class="text-muted">.Aky.</span>
<a class="tag" taget="_blank" href="/search/%E4%BD%8D%E8%BF%90%E7%AE%97/1.htm">位运算</a><a class="tag" taget="_blank" href="/search/PHP/1.htm">PHP</a><a class="tag" taget="_blank" href="/search/%E6%95%B0%E6%8D%AE%E5%BA%93/1.htm">数据库</a><a class="tag" taget="_blank" href="/search/%E5%87%BD%E6%95%B0/1.htm">函数</a><a class="tag" taget="_blank" href="/search/IP/1.htm">IP</a>
                                    <div>高手们照旧忽略。 
想弄个全天朝IP段数据库，找了个今天最新更新的国内所有运营商IP段，copy到文件，用文件函数，字符串函数把玩下。分割出startIp和endIp这样格式写入.txt文件，直接用phpmyadmin导入.csv文件的形式导入。（生命在于折腾，也许你们觉得我傻X，直接下载人家弄好的导入不就可以，做自己的菜鸟，让别人去说吧） 
 
当然用到了ip2long()函数把字符串转为整型数</div>
                                </li>
                                <li><a href="/article/3751.htm"
                                       title="sublime text 3 rust" target="_blank">sublime text 3 rust</a>
                                    <span class="text-muted">wudixiaotie</span>
<a class="tag" taget="_blank" href="/search/Sublime+Text/1.htm">Sublime Text</a>
                                    <div>1.sublime text 3 =&gt; install package =&gt; Rust 
2.cd ~/.config/sublime-text-3/Packages 
3.mkdir rust 
4.git clone https://github.com/sp0/rust-style 
5.cd rust-style 
6.cargo build --release 
7.ctrl</div>
                                </li>
                </ul>
            </div>
        </div>
    </div>

<div>
    <div class="container">
        <div class="indexes">
            <strong>按字母分类：</strong>
            <a href="/tags/A/1.htm" target="_blank">A</a><a href="/tags/B/1.htm" target="_blank">B</a><a href="/tags/C/1.htm" target="_blank">C</a><a
                href="/tags/D/1.htm" target="_blank">D</a><a href="/tags/E/1.htm" target="_blank">E</a><a href="/tags/F/1.htm" target="_blank">F</a><a
                href="/tags/G/1.htm" target="_blank">G</a><a href="/tags/H/1.htm" target="_blank">H</a><a href="/tags/I/1.htm" target="_blank">I</a><a
                href="/tags/J/1.htm" target="_blank">J</a><a href="/tags/K/1.htm" target="_blank">K</a><a href="/tags/L/1.htm" target="_blank">L</a><a
                href="/tags/M/1.htm" target="_blank">M</a><a href="/tags/N/1.htm" target="_blank">N</a><a href="/tags/O/1.htm" target="_blank">O</a><a
                href="/tags/P/1.htm" target="_blank">P</a><a href="/tags/Q/1.htm" target="_blank">Q</a><a href="/tags/R/1.htm" target="_blank">R</a><a
                href="/tags/S/1.htm" target="_blank">S</a><a href="/tags/T/1.htm" target="_blank">T</a><a href="/tags/U/1.htm" target="_blank">U</a><a
                href="/tags/V/1.htm" target="_blank">V</a><a href="/tags/W/1.htm" target="_blank">W</a><a href="/tags/X/1.htm" target="_blank">X</a><a
                href="/tags/Y/1.htm" target="_blank">Y</a><a href="/tags/Z/1.htm" target="_blank">Z</a><a href="/tags/0/1.htm" target="_blank">其他</a>
        </div>
    </div>
</div>
<footer id="footer" class="mb30 mt30">
    <div class="container">
        <div class="footBglm">
            <a target="_blank" href="/">首页</a> -
            <a target="_blank" href="/custom/about.htm">关于我们</a> -
            <a target="_blank" href="/search/Java/1.htm">站内搜索</a> -
            <a target="_blank" href="/sitemap.txt">Sitemap</a> -
            <a target="_blank" href="/custom/delete.htm">侵权投诉</a>
        </div>
        <div class="copyright">版权所有 IT知识库 CopyRight © 2000-2050 E-COM-NET.COM , All Rights Reserved.
<!--            <a href="https://beian.miit.gov.cn/" rel="nofollow" target="_blank">京ICP备09083238号</a><br>-->
        </div>
    </div>
</footer>
<!-- 代码高亮 -->
<script type="text/javascript" src="/static/syntaxhighlighter/scripts/shCore.js"></script>
<script type="text/javascript" src="/static/syntaxhighlighter/scripts/shLegacy.js"></script>
<script type="text/javascript" src="/static/syntaxhighlighter/scripts/shAutoloader.js"></script>
<link type="text/css" rel="stylesheet" href="/static/syntaxhighlighter/styles/shCoreDefault.css"/>
<script type="text/javascript" src="/static/syntaxhighlighter/src/my_start_1.js"></script>





</body>

</html>