Only generate one CNAME per PeerConnection.

webrtc/api/peerconnection.cc

 /*
  *  Copyright 2012 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 48 matching lines @@
 // Number of tokens must be preset when TURN uri has transport param.
 static const size_t kTurnTransportTokensNum = 2;
 // The default stun port.
 static const int kDefaultStunPort = 3478;
 static const int kDefaultStunTlsPort = 5349;
 static const char kTransport[] = "transport";
 
 // NOTE: Must be in the same order as the ServiceType enum.
 static const char* kValidIceServiceTypes[] = {"stun", "stuns", "turn", "turns"};
 
+// The length of RTCP CNAMEs.
+static const int kRtcpCnameLength = 16;
+
 // NOTE: A loop below assumes that the first value of this enum is 0 and all
 // other values are incremental.
 enum ServiceType {
   STUN = 0,  // Indicates a STUN server.
   STUNS,     // Indicates a STUN server used with a TLS session.
   TURN,      // Indicates a TURN server
   TURNS,     // Indicates a TURN server used with a TLS session.
   INVALID,   // Unknown.
 };
 static_assert(INVALID == arraysize(kValidIceServiceTypes),
@@ skipping 291 matching lines @@
       session_options->AddSendStream(cricket::MEDIA_TYPE_DATA, streamid,
                                      sync_label);
     }
   }
 }
 
 }  // namespace
 
 namespace webrtc {
 
+// Generate a RTCP CNAME when a PeerConnection is created.
+std::string GenerateRtcpCname() {
+  std::string cname;
+  if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
+    LOG(LS_ERROR) << "Failed to generate CNAME.";
+    RTC_DCHECK(false);
+  }
+  return cname;
+}
+
 bool ExtractMediaSessionOptions(
     const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
     bool is_offer,
     cricket::MediaSessionOptions* session_options) {
   typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
   if (!IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_audio) ||
       !IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_video)) {
     return false;
   }
 
@@ skipping 111 matching lines @@
 }
 
 PeerConnection::PeerConnection(PeerConnectionFactory* factory)
     : factory_(factory),
       observer_(NULL),
       uma_observer_(NULL),
       signaling_state_(kStable),
       ice_state_(kIceNew),
       ice_connection_state_(kIceConnectionNew),
       ice_gathering_state_(kIceGatheringNew),
+      rtcp_cname_(GenerateRtcpCname()),
       local_streams_(StreamCollection::Create()),
       remote_streams_(StreamCollection::Create()) {}
 
 PeerConnection::~PeerConnection() {
   TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
   RTC_DCHECK(signaling_thread()->IsCurrent());
   // Need to detach RTP senders/receivers from WebRtcSession,
   // since it's about to be destroyed.
   for (const auto& sender : senders_) {
     sender->Stop();
@@ skipping 975 matching lines @@
         !remote_video_tracks_.empty();
   }
   session_options->bundle_enabled =
       session_options->bundle_enabled &&
       (session_options->has_audio() || session_options->has_video() ||
        session_options->has_data());
 
   if (session_->data_channel_type() == cricket::DCT_SCTP && HasDataChannels()) {
     session_options->data_channel_type = cricket::DCT_SCTP;
   }
+
+  session_options->rtcp_cname = rtcp_cname_;
   return true;
 }
 
 void PeerConnection::FinishOptionsForAnswer(
     cricket::MediaSessionOptions* session_options) {
   // TODO(deadbeef): Once we have transceivers, enumerate them here instead of
   // ContentInfos.
   if (session_->remote_description()) {
     // Initialize the transport_options map.
     for (const cricket::ContentInfo& content :
@@ skipping 17 matching lines @@
 }
 
 bool PeerConnection::GetOptionsForAnswer(
     const MediaConstraintsInterface* constraints,
     cricket::MediaSessionOptions* session_options) {
   session_options->recv_audio = false;
   session_options->recv_video = false;
   if (!ParseConstraintsForAnswer(constraints, session_options)) {
     return false;
   }
+  session_options->rtcp_cname = rtcp_cname_;
+
   FinishOptionsForAnswer(session_options);
   return true;
 }
 
 bool PeerConnection::GetOptionsForAnswer(
     const RTCOfferAnswerOptions& options,
     cricket::MediaSessionOptions* session_options) {
   session_options->recv_audio = false;
   session_options->recv_video = false;
   if (!ExtractMediaSessionOptions(options, false, session_options)) {
     return false;
   }
+  session_options->rtcp_cname = rtcp_cname_;
+
   FinishOptionsForAnswer(session_options);
   return true;
 }
 
 void PeerConnection::RemoveTracks(cricket::MediaType media_type) {
   UpdateLocalTracks(std::vector<cricket::StreamParams>(), media_type);
   UpdateRemoteStreamsList(std::vector<cricket::StreamParams>(), false,
                           media_type, nullptr);
 }
 
@@ skipping 493 matching lines @@
 DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
   for (const auto& channel : sctp_data_channels_) {
     if (channel->id() == sid) {
       return channel;
     }
   }
   return nullptr;
 }
 
 }  // namespace webrtc