1 // Copyright 2013 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file.
5 // Multiply-included message file, hence no include guard.
7 // This file defines the IPCs for the browser-side implementation of
8 // WebSockets. For the legacy renderer-side implementation, see
9 // socket_stream_messages.h.
10 // TODO(ricea): Fix this comment when the legacy implementation has been
13 // This IPC interface is based on the WebSocket multiplexing draft spec,
14 // http://tools.ietf.org/html/draft-ietf-hybi-websocket-multiplexing-09
19 #include "base/basictypes.h"
20 #include "content/common/content_export.h"
21 #include "content/common/websocket.h"
22 #include "ipc/ipc_message_macros.h"
24 #include "url/origin.h"
26 #undef IPC_MESSAGE_EXPORT
27 #define IPC_MESSAGE_EXPORT CONTENT_EXPORT
28 #define IPC_MESSAGE_START WebSocketMsgStart
30 IPC_ENUM_TRAITS_MAX_VALUE(content::WebSocketMessageType
,
31 content::WEB_SOCKET_MESSAGE_TYPE_LAST
)
33 IPC_STRUCT_TRAITS_BEGIN(content::WebSocketHandshakeRequest
)
34 IPC_STRUCT_TRAITS_MEMBER(url
)
35 IPC_STRUCT_TRAITS_MEMBER(headers
)
36 IPC_STRUCT_TRAITS_MEMBER(headers_text
)
37 IPC_STRUCT_TRAITS_MEMBER(request_time
)
38 IPC_STRUCT_TRAITS_END()
40 IPC_STRUCT_TRAITS_BEGIN(content::WebSocketHandshakeResponse
)
41 IPC_STRUCT_TRAITS_MEMBER(url
)
42 IPC_STRUCT_TRAITS_MEMBER(status_code
)
43 IPC_STRUCT_TRAITS_MEMBER(status_text
)
44 IPC_STRUCT_TRAITS_MEMBER(headers
)
45 IPC_STRUCT_TRAITS_MEMBER(headers_text
)
46 IPC_STRUCT_TRAITS_MEMBER(response_time
)
47 IPC_STRUCT_TRAITS_END()
49 // WebSocket messages sent from the renderer to the browser.
51 // Open new virtual WebSocket connection to |socket_url|. |channel_id| is an
52 // identifier chosen by the renderer for the new channel. It cannot correspond
53 // to an existing open channel, and must be between 1 and
54 // 0x7FFFFFFF. |requested_protocols| is a list of tokens identifying
55 // sub-protocols the renderer would like to use, as described in RFC6455
56 // "Subprotocols Using the WebSocket Protocol".
58 // The browser process will not send |channel_id| as-is to the remote server; it
59 // will try to use a short id on the wire. This saves the renderer from
60 // having to try to choose the ids cleverly.
61 IPC_MESSAGE_ROUTED3(WebSocketHostMsg_AddChannelRequest
,
62 GURL
/* socket_url */,
63 std::vector
<std::string
> /* requested_protocols */,
64 url::Origin
/* origin */)
66 // WebSocket messages sent from the browser to the renderer.
68 // Respond to an AddChannelRequest for channel |channel_id|. |channel_id| is
69 // scoped to the renderer process; while it is unique per-renderer, the browser
70 // may have multiple renderers using the same id. If |fail| is true, the channel
71 // could not be established (the cause of the failure is not provided to the
72 // renderer in order to limit its ability to abuse WebSockets to perform network
73 // probing, etc.). If |fail| is set then the |channel_id| is available for
74 // re-use. |selected_protocol| is the sub-protocol the server selected,
75 // or empty if no sub-protocol was selected. |extensions| is the list of
76 // extensions negotiated for the connection.
77 IPC_MESSAGE_ROUTED3(WebSocketMsg_AddChannelResponse
,
79 std::string
/* selected_protocol */,
80 std::string
/* extensions */)
82 // Notify the renderer that the browser has started an opening handshake.
83 // This message is for showing the request in the inspector and
84 // can be omitted if the inspector is not active.
85 IPC_MESSAGE_ROUTED1(WebSocketMsg_NotifyStartOpeningHandshake
,
86 content::WebSocketHandshakeRequest
/* request */)
88 // Notify the renderer that the browser has finished an opening handshake.
89 // This message precedes AddChannelResponse.
90 // This message is for showing the response in the inspector and
91 // can be omitted if the inspector is not active.
92 IPC_MESSAGE_ROUTED1(WebSocketMsg_NotifyFinishOpeningHandshake
,
93 content::WebSocketHandshakeResponse
/* response */)
95 // Notify the renderer that the browser is required to fail the connection
96 // (see RFC6455 7.1.7 for details).
97 // When the renderer process receives this messages it does the following:
98 // 1. Fire an error event.
99 // 2. Show |message| to the inspector.
100 // 3. Close the channel immediately uncleanly, as if it received
101 // DropChannel(was_clean = false, code = 1006, reason = "").
102 // |message| will be shown in the inspector and won't be passed to the script.
103 // TODO(yhirano): Find the way to pass |message| directly to the inspector
105 IPC_MESSAGE_ROUTED1(WebSocketMsg_NotifyFailure
,
106 std::string
/* message */)
108 // WebSocket messages that can be sent in either direction.
110 // Send a non-control frame on |channel_id|. If the sender is the renderer, it
111 // will be sent to the remote server. If the sender is the browser, it comes
112 // from the remote server. |fin| indicates that this frame is the last in the
113 // current message. |type| is the type of the message. On the first frame of a
114 // message, it must be set to either WEB_SOCKET_MESSAGE_TYPE_TEXT or
115 // WEB_SOCKET_MESSAGE_TYPE_BINARY. On subsequent frames, it must be set to
116 // WEB_SOCKET_MESSAGE_TYPE_CONTINUATION, and the type is the same as that of the
117 // first message. If |type| is WEB_SOCKET_MESSAGE_TYPE_TEXT, then the
118 // concatenation of the |data| from every frame in the message must be valid
119 // UTF-8. If |fin| is not set, |data| must be non-empty.
120 IPC_MESSAGE_ROUTED3(WebSocketMsg_SendFrame
,
122 content::WebSocketMessageType
/* type */,
123 std::vector
<char> /* data */)
125 // Add |quota| tokens of send quota for channel |channel_id|. |quota| must be a
126 // positive integer. Both the browser and the renderer set send quota for the
127 // other side, and check that quota has not been exceeded when receiving
128 // messages. Both sides start a new channel with a quota of 0, and must wait for
129 // a FlowControl message before calling SendFrame. The total available quota on
130 // one side must never exceed 0x7FFFFFFFFFFFFFFF tokens.
131 IPC_MESSAGE_ROUTED1(WebSocketMsg_FlowControl
,
135 // When sent by the renderer, this will cause a DropChannel message to be sent
136 // if the multiplex extension is in use, otherwise a Close message will be sent
137 // and the TCP/IP connection will be closed.
138 // When sent by the browser, this indicates that a Close or DropChannel has been
139 // received, the connection was closed, or a network or protocol error
140 // occurred. On receiving DropChannel, the renderer process may consider the
141 // |channel_id| available for reuse by a new AddChannelRequest.
142 // |code| is one of the reason codes specified in RFC6455 or
143 // draft-ietf-hybi-websocket-multiplexing-09. |reason|, if non-empty, is a
144 // UTF-8 encoded string which may be useful for debugging but is not necessarily
145 // human-readable, as supplied by the server in the Close or DropChannel
147 // If |was_clean| is false on a message from the browser, then the WebSocket
148 // connection was not closed cleanly. If |was_clean| is false on a message from
149 // the renderer, then the connection should be closed immediately without a
150 // closing handshake and the renderer cannot accept any new messages on this
152 IPC_MESSAGE_ROUTED3(WebSocketMsg_DropChannel
,
153 bool /* was_clean */,
154 unsigned short /* code */,
155 std::string
/* reason */)
157 // Notify the renderer that a closing handshake has been initiated by the
158 // server, so that it can set the Javascript readyState to CLOSING.
159 IPC_MESSAGE_ROUTED0(WebSocketMsg_NotifyClosing
)