Use the same draft version in SDP data channel answers as used in the offer.

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 383 matching lines @@
 class VideoContentDescription : public MediaContentDescriptionImpl<VideoCodec> {
  public:
   virtual ContentDescription* Copy() const {
     return new VideoContentDescription(*this);
   }
   virtual MediaType type() const { return MEDIA_TYPE_VIDEO; }
 };
 
 class DataContentDescription : public MediaContentDescriptionImpl<DataCodec> {
  public:
+  DataContentDescription() {}
+
   virtual ContentDescription* Copy() const {
     return new DataContentDescription(*this);
   }
   virtual MediaType type() const { return MEDIA_TYPE_DATA; }
+
+  bool use_sctpmap() const { return use_sctpmap_; }
+  void set_use_sctpmap(bool enable) { use_sctpmap_ = enable; }
+
+ private:
+  bool use_sctpmap_ = true;
 };
 
 // Creates media session descriptions according to the supplied codecs and
 // other fields, as well as the supplied per-call options.
 // When creating answers, performs the appropriate negotiation
 // of the various fields to determine the proper result.
 class MediaSessionDescriptionFactory {
  public:
   // Default ctor; use methods below to set configuration.
   // The TransportDescriptionFactory is not owned by MediaSessionDescFactory,
@@ skipping 31 matching lines @@
   // Decides if a StreamParams shall be added to the audio and video media
   // content in SessionDescription when CreateOffer and CreateAnswer is called
   // even if |options| don't include a Stream. This is needed to support legacy
   // applications. |add_legacy_| is true per default.
   void set_add_legacy_streams(bool add_legacy) { add_legacy_ = add_legacy; }
 
   SessionDescription* CreateOffer(
       const MediaSessionOptions& options,
       const SessionDescription* current_description) const;
   SessionDescription* CreateAnswer(
-        const SessionDescription* offer,
-        const MediaSessionOptions& options,
-        const SessionDescription* current_description) const;
+      const SessionDescription* offer,
+      const MediaSessionOptions& options,
+      const SessionDescription* current_description) const;
 
  private:
   const AudioCodecs& GetAudioCodecsForOffer(
       const RtpTransceiverDirection& direction) const;
   const AudioCodecs& GetAudioCodecsForAnswer(
       const RtpTransceiverDirection& offer,
       const RtpTransceiverDirection& answer) const;
   void GetCodecsToOffer(const SessionDescription* current_description,
                         const AudioCodecs& supported_audio_codecs,
                         const VideoCodecs& supported_video_codecs,
@@ skipping 130 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_