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(); |