| Index: webrtc/api/peerconnectioninterface.h
|
| diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h
|
| index bcb6170c1226ee57f4b26d8855a25ac5dbbd2d5e..f88ccf65538ae58988d7e0b369101a30f2598256 100644
|
| --- a/webrtc/api/peerconnectioninterface.h
|
| +++ b/webrtc/api/peerconnectioninterface.h
|
| @@ -252,6 +252,16 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
|
| RTCConfiguration() = default;
|
| RTCConfiguration(RTCConfigurationType type) {
|
| if (type == RTCConfigurationType::kAggressive) {
|
| + // These parameters are also defined in Java and IOS configurations,
|
| + // so their values may be overwritten by the Java or IOS configuration.
|
| + bundle_policy = kBundlePolicyMaxBundle;
|
| + rtcp_mux_policy = kRtcpMuxPolicyRequire;
|
| + ice_connection_receiving_timeout =
|
| + kAggressiveIceConnectionReceivingTimeout;
|
| +
|
| + // These parameters are not defined in Java or IOS configuration,
|
| + // so their values will not be overwritten.
|
| + enable_ice_renomination = true;
|
| redetermine_role_on_ice_restart = false;
|
| }
|
| }
|
| @@ -288,6 +298,8 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
|
| static const int kUndefined = -1;
|
| // Default maximum number of packets in the audio jitter buffer.
|
| static const int kAudioJitterBufferMaxPackets = 50;
|
| + // ICE connection receiving timeout for aggressive configuration.
|
| + static const int kAggressiveIceConnectionReceivingTimeout = 1000;
|
| // TODO(pthatcher): Rename this ice_transport_type, but update
|
| // Chromium at the same time.
|
| IceTransportsType type = kAll;
|
|
|