Accept SDP with TRANSPORT attributes missing from bundled m= sections.

webrtc/pc/mediasession.h

 /*
  *  Copyright 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 476 matching lines @@
   bool AddTransportOffer(
       const std::string& content_name,
       const TransportOptions& transport_options,
       const SessionDescription* current_desc,
       SessionDescription* offer) const;
 
   TransportDescription* CreateTransportAnswer(
       const std::string& content_name,
       const SessionDescription* offer_desc,
       const TransportOptions& transport_options,
-      const SessionDescription* current_desc) const;
+      const SessionDescription* current_desc,
+      bool require_transport_attributes) const;
 
   bool AddTransportAnswer(
       const std::string& content_name,
       const TransportDescription& transport_desc,
       SessionDescription* answer_desc) const;
 
   // Helpers for adding media contents to the SessionDescription. Returns true
   // it succeeds or the media content is not needed, or false if there is any
   // error.
 
@@ skipping 13 matching lines @@
       StreamParamsVec* current_streams,
       SessionDescription* desc) const;
 
   bool AddDataContentForOffer(
       const MediaSessionOptions& options,
       const SessionDescription* current_description,
       DataCodecs* data_codecs,
       StreamParamsVec* current_streams,
       SessionDescription* desc) const;
 
-  bool AddAudioContentForAnswer(
-      const SessionDescription* offer,
-      const MediaSessionOptions& options,
-      const SessionDescription* current_description,
-      StreamParamsVec* current_streams,
-      SessionDescription* answer) const;
+  bool AddAudioContentForAnswer(const SessionDescription* offer,
+                                const MediaSessionOptions& options,
+                                const SessionDescription* current_description,
+                                const TransportInfo* bundle_transport,
+                                StreamParamsVec* current_streams,
+                                SessionDescription* answer) const;
 
-  bool AddVideoContentForAnswer(
-      const SessionDescription* offer,
-      const MediaSessionOptions& options,
-      const SessionDescription* current_description,
-      StreamParamsVec* current_streams,
-      SessionDescription* answer) const;
+  bool AddVideoContentForAnswer(const SessionDescription* offer,
+                                const MediaSessionOptions& options,
+                                const SessionDescription* current_description,
+                                const TransportInfo* bundle_transport,
+                                StreamParamsVec* current_streams,
+                                SessionDescription* answer) const;
 
-  bool AddDataContentForAnswer(
-      const SessionDescription* offer,
-      const MediaSessionOptions& options,
-      const SessionDescription* current_description,
-      StreamParamsVec* current_streams,
-      SessionDescription* answer) const;
+  bool AddDataContentForAnswer(const SessionDescription* offer,
+                               const MediaSessionOptions& options,
+                               const SessionDescription* current_description,
+                               const TransportInfo* bundle_transport,
+                               StreamParamsVec* current_streams,
+                               SessionDescription* answer) const;
 
   AudioCodecs audio_send_codecs_;
   AudioCodecs audio_recv_codecs_;
   AudioCodecs audio_sendrecv_codecs_;
   RtpHeaderExtensions audio_rtp_extensions_;
   VideoCodecs video_codecs_;
   RtpHeaderExtensions video_rtp_extensions_;
   DataCodecs data_codecs_;
   SecurePolicy secure_;
   bool add_legacy_;
@@ skipping 49 matching lines @@
 void GetSupportedVideoCryptoSuiteNames(const rtc::CryptoOptions& crypto_options,
     std::vector<std::string>* crypto_suite_names);
 void GetSupportedDataCryptoSuiteNames(const rtc::CryptoOptions& crypto_options,
     std::vector<std::string>* crypto_suite_names);
 void GetDefaultSrtpCryptoSuiteNames(const rtc::CryptoOptions& crypto_options,
     std::vector<std::string>* crypto_suite_names);
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_MEDIASESSION_H_