crash when peerConnection establish connection and begin media transfer

[PureCoderPan]

sender begin transfer media data using WebRTC and server can received data, but little seconds later application will crashed in sub thread.

console log stack backtrace:
(lldb) bt

• thread Add messaging interface to RestCommClient #8, name = 'Thread 0x0x105d154d0', stop reason = EXC_BAD_ACCESS (code=1, address=0x0)
  • frame #0: 0x00000001051c3050 WebRTCwebrtc::voe::TransportFeedbackProxy::OnTransportFeedback(webrtc::rtcp::TransportFeedback const&) + 40 frame #1: 0x00000001052c4e58 WebRTCwebrtc::RTCPReceiver::TriggerCallbacksFromRTCPPacket(webrtc::RTCPReceiver::PacketInformation const&) + 868
    frame Cannot hang up after receiving a call to iOS #2: 0x00000001052c45dc WebRTCwebrtc::RTCPReceiver::IncomingPacket(unsigned char const*, unsigned long) + 120 frame #3: 0x00000001052da69c WebRTCwebrtc::ModuleRtpRtcpImpl::IncomingRtcpPacket(unsigned char const*, unsigned long) + 16
    frame Not receiving evens from sofia SIP #4: 0x00000001051bf7dc WebRTCwebrtc::voe::Channel::ReceivedRTCPPacket(unsigned char const*, unsigned long) + 112 frame #5: 0x00000001051c42bc WebRTCwebrtc::voe::ChannelProxy::ReceivedRTCPPacket(unsigned char const*, unsigned long) + 20
    frame Facilitate RCConnection::disconnect (either cancel or bye) and RCConnection::reject functionality #6: 0x00000001051b4010 WebRTCwebrtc::internal::Call::DeliverRtcp(webrtc::MediaType, unsigned char const*, unsigned long) + 448 frame #7: 0x0000000105413f18 WebRTCcricket::WebRtcVoiceMediaChannel::OnRtcpReceived(rtc::CopyOnWriteBuffer*, rtc::PacketTime const&) + 92
    frame Add messaging interface to RestCommClient #8: 0x00000001054ae174 WebRTCcricket::BaseChannel::OnPacketReceived(bool, rtc::CopyOnWriteBuffer const&, rtc::PacketTime const&) + 80 frame #9: 0x00000001050b4d54 WebRTCrtc::AsyncInvoker::OnMessage(rtc::Message*) + 84
    frame Make client AOR and registrar configurable in the UI #10: 0x00000001050c3694 WebRTCrtc::MessageQueue::Dispatch(rtc::Message*) + 228 frame #11: 0x00000001050db66c WebRTCrtc::Thread::ProcessMessages(int) + 172
    frame Disable auto correct for fields holding SIP URIs #12: 0x00000001050db48c WebRTCrtc::Thread::PreRun(void*) + 108 frame #13: 0x00000001b71ab914 libsystem_pthread.dylib_pthread_start + 168

any suggestions will appreciated.