Bug 4517: Enable connectivity for demo page when no STUN/TURN specifiefd.

talk/app/webrtc/peerconnectioninterface.h

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 203 matching lines @@
     kTcpCandidatePolicyDisabled
   };
 
   struct RTCConfiguration {
     // TODO(pthatcher): Rename this ice_transport_type, but update
     // Chromium at the same time.
     IceTransportsType type;
     // TODO(pthatcher): Rename this ice_servers, but update Chromium
     // at the same time.
     IceServers servers;
+    // If |enable_localhost_ice_candidate| is true, localhost will be issued as
+    // candidate in the case when multiple_routes is disabled.

[pthatcher1, 2015/08/07 21:59:49]

Would it be easier to reason about the rule "a localhost candidate is signaled
whenever a candidate with the any address is allocated"?

[guoweis_webrtc, 2015/08/13 15:08:52]

Done.

+    bool enable_localhost_ice_candidate;
     BundlePolicy bundle_policy;
     RtcpMuxPolicy rtcp_mux_policy;
     TcpCandidatePolicy tcp_candidate_policy;
     int audio_jitter_buffer_max_packets;
     bool audio_jitter_buffer_fast_accelerate;
 
     RTCConfiguration()
         : type(kAll),
+          enable_localhost_ice_candidate(false),
           bundle_policy(kBundlePolicyBalanced),
           rtcp_mux_policy(kRtcpMuxPolicyNegotiate),
           tcp_candidate_policy(kTcpCandidatePolicyEnabled),
           audio_jitter_buffer_max_packets(50),
           audio_jitter_buffer_fast_accelerate(false) {}
   };
 
   struct RTCOfferAnswerOptions {
     static const int kUndefined = -1;
     static const int kMaxOfferToReceiveMedia = 1;
@@ skipping 385 matching lines @@
 CreatePeerConnectionFactory(
     rtc::Thread* worker_thread,
     rtc::Thread* signaling_thread,
     AudioDeviceModule* default_adm,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory);
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_