C++ 在webrtc对等连接中创建UDP套接字失败

我正在处理webrtc PeerConnection，我发现UDP套接字创建失败 下面介绍了调用CreatePeerConnection方法的代码段。 我使用了我自己的stun和turn服务器，并在给定的代码中提到了它们的ip和端口，我也尝试过使用google stun服务器地址和端口（

stun:stun.l.google.com:19302

），但遇到了同样的问题

webrtc::PeerConnectionInterface::IceServers ice_servers;
webrtc::PeerConnectionInterface::IceServer ice_server;
webrtc::PeerConnectionInterface::RTCConfiguration config;
ice_server.uri = "stun:address_stun:port_stun";
config.servers.push_back(ice_server);

webrtc::PeerConnectionInterface::IceServer turn_server;
std::string url = "turn:address_turn:port_turn?transport=udp";

turn_server.urls.push_back(url);
turn_server.username = "username";
turn_server.password = "password";
config.servers.push_back(turn_server);

webrtc::FakeConstraints constraints;
constraints.AddOptional(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, "true");

config.candidate_network_policy = webrtc::PeerConnectionInterface::kCandidateNetworkPolicyLowCost;
config.tcp_candidate_policy = webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled;

constraints.AddOptional(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                   webrtc::MediaConstraintsInterface::kValueFalse);
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kOfferToReceiveAudio,
                    webrtc::MediaConstraintsInterface::kValueTrue);


rtc::ThreadManager::Instance()->WrapCurrentThread();

u_worker_thread = rtc::Thread::Create();
u_worker_thread->SetName("worker_thread", NULL);
RTC_CHECK(u_worker_thread->Start()) << "Failed to start thread";

u_signaling_thread = rtc::Thread::Create();
u_signaling_thread->SetName("signaling_thread", NULL);
RTC_CHECK(u_signaling_thread->Start()) << "Failed to start thread";


m_networkThread = rtc::Thread::Create();
m_networkThread->SetName("networking_thread", NULL);
RTC_CHECK(m_networkThread->Start()) << "Failed to start thread";


cricket::WebRtcVideoEncoderFactory* video_encoder_factory = nullptr;
cricket::WebRtcVideoDecoderFactory* video_decoder_factory = nullptr;
auto audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
auto audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();


webrtc::AudioDeviceModule* adm = nullptr;

fake_network_manager_.reset(new rtc::FakeNetworkManager());
static const SocketAddress kDefaultLocalAddress("local_ip", 0);
fake_network_manager_->AddInterface(kDefaultLocalAddress);
std::unique_ptr<cricket::PortAllocator> port_allocator_(new cricket::BasicPortAllocator(fake_network_manager_.get()));

_peerConnectionFactory = webrtc::CreatePeerConnectionFactory(m_networkThread.get(),u_worker_thread.get(),u_signaling_thread.get(),adm,audio_encoder_factory,audio_decoder_factory,video_encoder_factory,video_decoder_factory); 

if (!_peerConnectionFactory.get()) {

}
else
{
    __android_log_print(ANDROID_LOG_INFO, TAG,"Going to initialise CreatePeerConnection" );
    _peerConnection = _peerConnectionFactory->CreatePeerConnection(
                        config, &constraints, std::move(port_allocator_), NULL, this);

}

webrtc:：PeerConnectionInterface:：IceServers ice\u服务器；
webrtc:：PeerConnectionInterface:：IceServer ice_服务器；
webrtc:：PeerConnectionInterface:：RTC配置；
ice\u server.uri=“stun:address\u stun:port\u stun”；
config.servers.push_back（ice_服务器）；
webrtc:：PeerConnectionInterface:：IceServer turn\u服务器；
std:：string url=“turn:address\u turn:port\u turn？transport=udp”；
打开服务器.url.push-back（url）；
打开\u server.username=“username”；
打开\u server.password=“password”；
config.servers.push_back（打开_服务器）；
webrtc:：伪造约束；
AddOptional（webrtc:：MediaConstraintsInterface:：kEnableDtlsSrtp，“true”）；
config.candidate\u network\u policy=webrtc:：PeerConnectionInterface:：kCandidateNetworkPolicyLowCost；
config.tcp_candidate_policy=webrtc:：PeerConnectionInterface:：kTcpCandidatePolicyDisabled；
constraints.AddOptional（webrtc:：MediaConstraintsInterface:：kEnableDtlsSrtp、，
webrtc:：MediaConstraintInterface:：kValueFalse）；
constraints.add强制（webrtc:：MediaConstraintsInterface:：kOfferToReceiveAudio，
webrtc:：MediaConstraintInterface:：kValueTrue）；
rtc:：ThreadManager:：Instance（）->WrapCurrentThread（）；
u_worker_thread=rtc:：thread:：Create（）；
工作线程->设置名称（“工作线程”，NULL）；
RTC_CHECK（u_worker_thread->Start（））SetName（“signaling_thread”，NULL）；
RTC_检查（u_信令_线程->开始（））SetName（“网络_线程”，NULL）；
RTC_CHECK（m_networkThread->Start（））AddInterface（kDefaultLocalAddress）；
std:：unique_ptr port_分配器（新的cricket:：BasicPortAllocator（伪网络管理器_u.get（））；
_peerConnectionFactory=webrtc:：CreatePeerConnectionFactory（m_networkThread.get（）、u_worker_thread.get（）、u_信令_thread.get（）、adm、音频编码器工厂、音频解码器工厂、视频编码器工厂、视频解码器工厂）；
if（！\u peerConnectionFactory.get（））{
}
其他的
{
__android_log_print（android_log_信息，标签，“将初始化CreatePeerConnection”）；
_peerConnection=\u peerConnectionFactory->CreatePeerConnection(
配置和约束，std:：move（端口分配器），NULL，this）；
}

我也面临同样的问题