Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(75)

Side by Side Diff: pc/channel.h

Issue 3018513002: Revert of Completed the functionalities of SrtpTransport. (Closed)
Patch Set: Created 3 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « p2p/base/fakepackettransport.h ('k') | pc/channel.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2004 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
(...skipping 15 matching lines...) Expand all
26 #include "media/base/videosinkinterface.h" 26 #include "media/base/videosinkinterface.h"
27 #include "media/base/videosourceinterface.h" 27 #include "media/base/videosourceinterface.h"
28 #include "p2p/base/dtlstransportinternal.h" 28 #include "p2p/base/dtlstransportinternal.h"
29 #include "p2p/base/packettransportinternal.h" 29 #include "p2p/base/packettransportinternal.h"
30 #include "p2p/base/transportcontroller.h" 30 #include "p2p/base/transportcontroller.h"
31 #include "p2p/client/socketmonitor.h" 31 #include "p2p/client/socketmonitor.h"
32 #include "pc/audiomonitor.h" 32 #include "pc/audiomonitor.h"
33 #include "pc/mediamonitor.h" 33 #include "pc/mediamonitor.h"
34 #include "pc/mediasession.h" 34 #include "pc/mediasession.h"
35 #include "pc/rtcpmuxfilter.h" 35 #include "pc/rtcpmuxfilter.h"
36 #include "pc/rtptransportinternal.h"
36 #include "pc/srtpfilter.h" 37 #include "pc/srtpfilter.h"
37 #include "rtc_base/asyncinvoker.h" 38 #include "rtc_base/asyncinvoker.h"
38 #include "rtc_base/asyncudpsocket.h" 39 #include "rtc_base/asyncudpsocket.h"
39 #include "rtc_base/criticalsection.h" 40 #include "rtc_base/criticalsection.h"
40 #include "rtc_base/network.h" 41 #include "rtc_base/network.h"
41 #include "rtc_base/sigslot.h" 42 #include "rtc_base/sigslot.h"
42 #include "rtc_base/window.h" 43 #include "rtc_base/window.h"
43 44
44 namespace webrtc { 45 namespace webrtc {
45 class AudioSinkInterface; 46 class AudioSinkInterface;
46 class RtpTransportInternal;
47 class SrtpTransport;
48 } // namespace webrtc 47 } // namespace webrtc
49 48
50 namespace cricket { 49 namespace cricket {
51 50
52 struct CryptoParams; 51 struct CryptoParams;
53 class MediaContentDescription; 52 class MediaContentDescription;
54 53
55 // BaseChannel contains logic common to voice and video, including enable, 54 // BaseChannel contains logic common to voice and video, including enable,
56 // marshaling calls to a worker and network threads, and connection and media 55 // marshaling calls to a worker and network threads, and connection and media
57 // monitors. 56 // monitors.
(...skipping 35 matching lines...) Expand 10 before | Expand all | Expand 10 after
93 // done. 92 // done.
94 void Deinit(); 93 void Deinit();
95 94
96 rtc::Thread* worker_thread() const { return worker_thread_; } 95 rtc::Thread* worker_thread() const { return worker_thread_; }
97 rtc::Thread* network_thread() const { return network_thread_; } 96 rtc::Thread* network_thread() const { return network_thread_; }
98 const std::string& content_name() const { return content_name_; } 97 const std::string& content_name() const { return content_name_; }
99 // TODO(deadbeef): This is redundant; remove this. 98 // TODO(deadbeef): This is redundant; remove this.
100 const std::string& transport_name() const { return transport_name_; } 99 const std::string& transport_name() const { return transport_name_; }
101 bool enabled() const { return enabled_; } 100 bool enabled() const { return enabled_; }
102 101
103 // This function returns true if we are using SDES. 102 // This function returns true if we are using SRTP.
104 bool sdes_active() const { return sdes_negotiator_.IsActive(); } 103 bool secure() const { return srtp_filter_.IsActive(); }
105 // The following function returns true if we are using DTLS-based keying. 104 // The following function returns true if we are using
106 bool dtls_active() const { return dtls_active_; } 105 // DTLS-based keying. If you turned off SRTP later, however
107 // This function returns true if using SRTP (DTLS-based keying or SDES). 106 // you could have secure() == false and dtls_secure() == true.
108 bool srtp_active() const { return sdes_active() || dtls_active(); } 107 bool secure_dtls() const { return dtls_keyed_; }
109 108
110 bool writable() const { return writable_; } 109 bool writable() const { return writable_; }
111 110
112 // Set the transport(s), and update writability and "ready-to-send" state. 111 // Set the transport(s), and update writability and "ready-to-send" state.
113 // |rtp_transport| must be non-null. 112 // |rtp_transport| must be non-null.
114 // |rtcp_transport| must be supplied if NeedsRtcpTransport() is true (meaning 113 // |rtcp_transport| must be supplied if NeedsRtcpTransport() is true (meaning
115 // RTCP muxing is not fully active yet). 114 // RTCP muxing is not fully active yet).
116 // |rtp_transport| and |rtcp_transport| must share the same transport name as 115 // |rtp_transport| and |rtcp_transport| must share the same transport name as
117 // well. 116 // well.
118 // Can not start with "rtc::PacketTransportInternal" and switch to 117 // Can not start with "rtc::PacketTransportInternal" and switch to
(...skipping 57 matching lines...) Expand 10 before | Expand all | Expand 10 after
176 bool NeedsRtcpTransport(); 175 bool NeedsRtcpTransport();
177 176
178 // From RtpTransport - public for testing only 177 // From RtpTransport - public for testing only
179 void OnTransportReadyToSend(bool ready); 178 void OnTransportReadyToSend(bool ready);
180 179
181 // Only public for unit tests. Otherwise, consider protected. 180 // Only public for unit tests. Otherwise, consider protected.
182 int SetOption(SocketType type, rtc::Socket::Option o, int val) 181 int SetOption(SocketType type, rtc::Socket::Option o, int val)
183 override; 182 override;
184 int SetOption_n(SocketType type, rtc::Socket::Option o, int val); 183 int SetOption_n(SocketType type, rtc::Socket::Option o, int val);
185 184
185 SrtpFilter* srtp_filter() { return &srtp_filter_; }
186
186 virtual cricket::MediaType media_type() = 0; 187 virtual cricket::MediaType media_type() = 0;
187 188
188 // This function returns true if we require SRTP for call setup. 189 // This function returns true if we require SRTP for call setup.
189 bool srtp_required_for_testing() const { return srtp_required_; } 190 bool srtp_required_for_testing() const { return srtp_required_; }
190 191
191 // Public for testing. 192 // Public for testing.
192 // TODO(zstein): Remove this once channels register themselves with 193 // TODO(zstein): Remove this once channels register themselves with
193 // an RtpTransport in a more explicit way. 194 // an RtpTransport in a more explicit way.
194 bool HandlesPayloadType(int payload_type) const; 195 bool HandlesPayloadType(int payload_type) const;
195 196
(...skipping 165 matching lines...) Expand 10 before | Expand all | Expand 10 after
361 rtc::PacketTransportInternal* rtp_packet_transport, 362 rtc::PacketTransportInternal* rtp_packet_transport,
362 rtc::PacketTransportInternal* rtcp_packet_transport); 363 rtc::PacketTransportInternal* rtcp_packet_transport);
363 void DisconnectTransportChannels_n(); 364 void DisconnectTransportChannels_n();
364 void SignalSentPacket_n(rtc::PacketTransportInternal* transport, 365 void SignalSentPacket_n(rtc::PacketTransportInternal* transport,
365 const rtc::SentPacket& sent_packet); 366 const rtc::SentPacket& sent_packet);
366 void SignalSentPacket_w(const rtc::SentPacket& sent_packet); 367 void SignalSentPacket_w(const rtc::SentPacket& sent_packet);
367 bool IsReadyToSendMedia_n() const; 368 bool IsReadyToSendMedia_n() const;
368 void CacheRtpAbsSendTimeHeaderExtension_n(int rtp_abs_sendtime_extn_id); 369 void CacheRtpAbsSendTimeHeaderExtension_n(int rtp_abs_sendtime_extn_id);
369 int GetTransportOverheadPerPacket() const; 370 int GetTransportOverheadPerPacket() const;
370 void UpdateTransportOverhead(); 371 void UpdateTransportOverhead();
371 // Wraps the existing RtpTransport in an SrtpTransport.
372 void EnableSrtpTransport_n();
373 372
374 rtc::Thread* const worker_thread_; 373 rtc::Thread* const worker_thread_;
375 rtc::Thread* const network_thread_; 374 rtc::Thread* const network_thread_;
376 rtc::Thread* const signaling_thread_; 375 rtc::Thread* const signaling_thread_;
377 rtc::AsyncInvoker invoker_; 376 rtc::AsyncInvoker invoker_;
378 377
379 const std::string content_name_; 378 const std::string content_name_;
380 std::unique_ptr<ConnectionMonitor> connection_monitor_; 379 std::unique_ptr<ConnectionMonitor> connection_monitor_;
381 380
382 // Won't be set when using raw packet transports. SDP-specific thing. 381 // Won't be set when using raw packet transports. SDP-specific thing.
383 std::string transport_name_; 382 std::string transport_name_;
384 383
385 const bool rtcp_mux_required_; 384 const bool rtcp_mux_required_;
386 385
387 // Separate DTLS/non-DTLS pointers to support using BaseChannel without DTLS. 386 // Separate DTLS/non-DTLS pointers to support using BaseChannel without DTLS.
388 // Temporary measure until more refactoring is done. 387 // Temporary measure until more refactoring is done.
389 // If non-null, "X_dtls_transport_" will always equal "X_packet_transport_". 388 // If non-null, "X_dtls_transport_" will always equal "X_packet_transport_".
390 DtlsTransportInternal* rtp_dtls_transport_ = nullptr; 389 DtlsTransportInternal* rtp_dtls_transport_ = nullptr;
391 DtlsTransportInternal* rtcp_dtls_transport_ = nullptr; 390 DtlsTransportInternal* rtcp_dtls_transport_ = nullptr;
392 std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport_; 391 std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport_;
393 webrtc::SrtpTransport* srtp_transport_ = nullptr;
394 std::vector<std::pair<rtc::Socket::Option, int> > socket_options_; 392 std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
395 std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_; 393 std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
396 SrtpFilter sdes_negotiator_; 394 SrtpFilter srtp_filter_;
397 RtcpMuxFilter rtcp_mux_filter_; 395 RtcpMuxFilter rtcp_mux_filter_;
398 bool writable_ = false; 396 bool writable_ = false;
399 bool was_ever_writable_ = false; 397 bool was_ever_writable_ = false;
400 bool has_received_packet_ = false; 398 bool has_received_packet_ = false;
401 bool dtls_active_ = false; 399 bool dtls_keyed_ = false;
402 const bool srtp_required_ = true; 400 const bool srtp_required_ = true;
401 int rtp_abs_sendtime_extn_id_ = -1;
403 402
404 // MediaChannel related members that should be accessed from the worker 403 // MediaChannel related members that should be accessed from the worker
405 // thread. 404 // thread.
406 MediaChannel* const media_channel_; 405 MediaChannel* const media_channel_;
407 // Currently the |enabled_| flag is accessed from the signaling thread as 406 // Currently the |enabled_| flag is accessed from the signaling thread as
408 // well, but it can be changed only when signaling thread does a synchronous 407 // well, but it can be changed only when signaling thread does a synchronous
409 // call to the worker thread, so it should be safe. 408 // call to the worker thread, so it should be safe.
410 bool enabled_ = false; 409 bool enabled_ = false;
411 std::vector<StreamParams> local_streams_; 410 std::vector<StreamParams> local_streams_;
412 std::vector<StreamParams> remote_streams_; 411 std::vector<StreamParams> remote_streams_;
(...skipping 304 matching lines...) Expand 10 before | Expand all | Expand 10 after
717 // SetSendParameters. 716 // SetSendParameters.
718 DataSendParameters last_send_params_; 717 DataSendParameters last_send_params_;
719 // Last DataRecvParameters sent down to the media_channel() via 718 // Last DataRecvParameters sent down to the media_channel() via
720 // SetRecvParameters. 719 // SetRecvParameters.
721 DataRecvParameters last_recv_params_; 720 DataRecvParameters last_recv_params_;
722 }; 721 };
723 722
724 } // namespace cricket 723 } // namespace cricket
725 724
726 #endif // PC_CHANNEL_H_ 725 #endif // PC_CHANNEL_H_
OLDNEW
« no previous file with comments | « p2p/base/fakepackettransport.h ('k') | pc/channel.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698