| Index: remoting/host/cast_extension_session.cc
|
| diff --git a/remoting/host/cast_extension_session.cc b/remoting/host/cast_extension_session.cc
|
| index 1b8df914dee9adba2fb571bcde6ce0df778f3bb7..9e4b12df4ab44575c82573fbf69cd44adaaf3609 100644
|
| --- a/remoting/host/cast_extension_session.cc
|
| +++ b/remoting/host/cast_extension_session.cc
|
| @@ -12,7 +12,7 @@
|
| #include "net/url_request/url_request_context_getter.h"
|
| #include "remoting/host/client_session.h"
|
| #include "remoting/proto/control.pb.h"
|
| -#include "remoting/protocol/chromium_port_allocator_factory.h"
|
| +#include "remoting/protocol/chromium_port_allocator.h"
|
| #include "remoting/protocol/client_stub.h"
|
| #include "remoting/protocol/webrtc_video_capturer_adapter.h"
|
| #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
|
| @@ -50,8 +50,9 @@ const char kVideoLabel[] = "cast_video_label";
|
| const char kStreamLabel[] = "stream_label";
|
|
|
| // Default STUN server used to construct
|
| -// webrtc::PeerConnectionInterface::RTCConfiguration for the PeerConnection.
|
| -const char kDefaultStunURI[] = "stun:stun.l.google.com:19302";
|
| +// ChromiumPortAllocator for the PeerConnection.
|
| +const char kDefaultStunHost[] = "stun.l.google.com";
|
| +const int kDefaultStunPort = 19302;
|
|
|
| const char kWorkerThreadName[] = "CastExtensionSessionWorkerThread";
|
|
|
| @@ -481,13 +482,6 @@ bool CastExtensionSession::InitializePeerConnection() {
|
|
|
| VLOG(1) << "Created PeerConnectionFactory successfully.";
|
|
|
| - webrtc::PeerConnectionInterface::IceServers servers;
|
| - webrtc::PeerConnectionInterface::IceServer server;
|
| - server.uri = kDefaultStunURI;
|
| - servers.push_back(server);
|
| - webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
|
| - rtc_config.servers = servers;
|
| -
|
| // DTLS-SRTP is the preferred encryption method. If set to kValueFalse, the
|
| // peer connection uses SDES. Disabling SDES as well will cause the peer
|
| // connection to fail to connect.
|
| @@ -497,12 +491,17 @@ bool CastExtensionSession::InitializePeerConnection() {
|
| constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
| webrtc::MediaConstraintsInterface::kValueTrue);
|
|
|
| - rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
|
| - port_allocator_factory = protocol::ChromiumPortAllocatorFactory::Create(
|
| - network_settings_, url_request_context_getter_);
|
| + rtc::scoped_ptr<protocol::ChromiumPortAllocator> port_allocator(
|
| + protocol::ChromiumPortAllocator::Create(url_request_context_getter_,
|
| + network_settings_)
|
| + .release());
|
| + std::vector<rtc::SocketAddress> stun_hosts;
|
| + stun_hosts.push_back(rtc::SocketAddress(kDefaultStunHost, kDefaultStunPort));
|
| + port_allocator->SetStunHosts(stun_hosts);
|
|
|
| + webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
|
| peer_connection_ = peer_conn_factory_->CreatePeerConnection(
|
| - rtc_config, &constraints, port_allocator_factory, nullptr, this);
|
| + rtc_config, &constraints, port_allocator.Pass(), nullptr, this);
|
|
|
| if (!peer_connection_.get()) {
|
| CleanupPeerConnection();
|
|
|