title: running AppRTC for ios ,RtcEventLog issue
body: <p>I would like to add an interface in AppRTCMobile, this interface can start webrtc Call module, in order to achieve the audio call between two phones (LAN, already know both the IP address and port number), but when I run successfully , The software crashes every time an exception occurs when the method is called by RtcEventLog. I do not know if Calling Call is reasonable or not. I sincerely thank you for your help in the absence of a solution.
  Below the source code, please help me find the problem.</p>
  
  <pre><code>    std::unique_ptr&lt;RtcEventLog&gt; event_log = webrtc::RtcEventLog::Create();
      webrtc::Call::Config callConfig = webrtc::Call::Config(event_log.get());
      callConfig.bitrate_config.max_bitrate_bps = 500*1000;
      callConfig.bitrate_config.min_bitrate_bps = 100*1000;
      callConfig.bitrate_config.start_bitrate_bps = 250*1000;
  
      webrtc::AudioState::Config audio_state_config = webrtc::AudioState::Config();
      cricket::VoEWrapper* g_voe  = nullptr;
      rtc::scoped_refptr&lt;webrtc::AudioDecoderFactory&gt; g_audioDecoderFactory;
      g_audioDecoderFactory = webrtc::CreateBuiltinAudioDecoderFactory();
      g_voe = new cricket::VoEWrapper();
      audio_state_config.audio_processing = webrtc::AudioProcessing::Create();
      g_voe-&gt;base()-&gt;Init(NULL,audio_state_config.audio_processing,g_audioDecoderFactory);
      audio_state_config.voice_engine = g_voe-&gt;engine();
  
      audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create();
      callConfig.audio_state = AudioState::Create(audio_state_config);
      std::unique_ptr&lt;RtcEventLog&gt; event_logg = webrtc::RtcEventLog::Create();
      callConfig.event_log = event_logg.get();
      g_call = webrtc::Call::Create(callConfig);
  
  
  
      g_audioSendTransport = new AudioLoopbackTransport();
      webrtc::AudioSendStream::Config config(g_audioSendTransport);
      g_audioSendChannelId = g_voe-&gt;base()-&gt;CreateChannel();
      config.voe_channel_id = g_audioSendChannelId;
      g_audioSendStream = g_call-&gt;CreateAudioSendStream(config);
  
  
      webrtc::AudioReceiveStream::Config AudioReceiveConfig;
      AudioReceiveConfig.decoder_factory = g_audioDecoderFactory;
      g_audioReceiveChannelId = g_voe-&gt;base()-&gt;CreateChannel();
      AudioReceiveConfig.voe_channel_id = g_audioReceiveChannelId;
      g_audioReceiveStream = g_call-&gt;CreateAudioReceiveStream(AudioReceiveConfig);
  
      g_audioSendStream-&gt;Start();
      g_audioReceiveStream-&gt;Start();
  </code></pre>
  
  <p><a href="https://i.stack.imgur.com/NCxPv.png" rel="nofollow noreferrer">Here's a screenshot of the error that occurred when the crash occurred. Please tell me if you want to know more.</a></p>