증상은 연결 성공 후 바로 통화가 끊깁니다. RemonCall 옵션은 다음과 같이 줬습니다.
mRemonCall = RemonCall.remonBuilder() .context(this) .serviceId(getString(R.string.remon_service_id)) .key(getString(R.string.remon_key)) .isVideoCall(false) .build();
그리고 끊겼을 때 로그는 다음과 같습니다. 빠뜨린 부분이 있다면 알려주시면 감사하겠습니다.
I/audio_device_impl.cc: (line 820): output: 0
I/channel.cc: (line 867): Changing voice state, recv=1 send=1
I/WebRtcAudioRecord: AudioRecordThread@[name=AudioRecordJavaThread, id=25833]
I/port.cc: (line 1756): Conn[c73bb600:audio:Net[wlan0:192.168.0.x/24:Wifi:id=3]:zsC/qmro:1:0:local:udp:192.168.0.x:39034->gw+UvzLq:1:2122260223:local:udp:10.0.2.x:43315|C–I|-|0|0|9115038255631187454|-]: Sent STUN ping, id=4d41417948734d4667313167, use_candidate=0, nomination=0
I/port.cc: (line 528): Received STUN ping id=6e6843436c764e4c5150334a from unknown address 211.244.71.x:62970
I/turn_port.cc: (line 1586): Port[b8c21500:audio:1:0:relay:Net[wlan0:192.168.0.x/24:Wifi:id=3]]: TURN create permission request sent, id=2b364779674457424f553238
I/port.cc: (line 1126): Conn[b89cd800:audio:Net[wlan0:192.168.0.x/24:Wifi:id=3]:pq3AHMvM:1:0:relay:udp:52.79.76.x:44159->Es7g5YoX:1:1853824767:prflx:udp:211.244.71.x:62970|C–W|-|0|0|179896594391252478|-]: Connection created
I/p2p_transport_channel.cc: (line 988): Adding connection from peer reflexive candidate: Cand[:3372217884:1:udp:1853824767:211.244.71.74:62970:prflx::0:t8B+:o83EBvSeay8cjtA3L9vEa3Vr:5:10:0]
I/port.cc: (line 853): Port[b8c21500:audio:1:0:relay:Net[wlan0:192.168.0.x/24:Wifi:id=3]]: Sent STUN ping response, to=211.244.71.x:62970, id=6e6843436c764e4c5150334a
I/port.cc: (line 1404): Conn[b89cd800:audio:Net[wlan0:192.168.0.x/24:Wifi:id=3]:pq3AHMvM:1:0:relay:udp:52.79.76.x:44159->Es7g5YoX:1:1853824767:prflx:udp:211.244.71.x:62970|CR-W|-|0|0|179896594391252478|-]: Connection pruned
I/turn_port.cc: (line 1593): Port[b8c21500:audio:1:0:relay:Net[wlan0:192.168.0.x/24:Wifi:id=3]]: TURN permission requested successfully, id=2b364779674457424f553238, code=0, rtt=4
I/turn_port.cc: (line 1771): Port[b8c21500:audio:1:0:relay:Net[wlan0:192.168.0.x/24:Wifi:id=3]]: Create permission for 211.244.71.x:62970 succeeded
I/turn_port.cc: (line 1783): Port[b8c21500:audio:1:0:relay:Net[wlan0:192.168.0.x/24:Wifi:id=3]]: Scheduled create-permission-request in 240000ms.
I/port.cc: (line 1756): Conn[c73bb600:audio:Net[wlan0:192.168.0.x/24:Wifi:id=3]:zsC/qmro:1:0:local:udp:192.168.0.x:39034->gw+UvzLq:1:2122260223:local:udp:10.0.2.x:43315|C–I|-|0|0|9115038255631187454|-]: Sent STUN ping, id=71366f434469442b74324a6c, use_candidate=0, nomination=0
I/port.cc: (line 528): Received STUN ping id=494f4f616676554c6c474857 from unknown address 192.168.0.x:61314
I/port.cc: (line 1126): Conn[b89ce200:audio:Net[wlan0:192.168.0.x/24:Wifi:id=3]:zsC/qmro:1:0:local:udp:192.168.0.x:39034->4FBauv3t:1:1853693695:prflx:udp:192.168.0.x:61314|C–W|-|0|0|7961553801070919167|-]: Connection created
I/p2p_transport_channel.cc: (line 988): Adding connection from peer reflexive candidate: Cand[:170042222:1:udp:1853693695:192.168.0.6:61314:prflx::0:t8B+:o83EBvSeay8cjtA3L9vEa3Vr:3:900:0]
I/port.cc: (line 853): Port[b8c1f400:audio:1:0:local:Net[wlan0:192.168.0.x/24:Wifi:id=3]]: Sent STUN ping response, to=192.168.0.x:61314, id=494f4f616676554c6c474857
I/port.cc: (line 1404): Conn[b89ce200:audio:Net[wlan0:192.168.0.x/24:Wifi:id=3]:zsC/qmro:1:0:local:udp:192.168.0.x:39034->4FBauv3t:1:1853693695:prflx:udp:192.168.0.x:61314|CR-W|-|0|0|7961553801070919167|-]: Connection pruned
I/audio_device_buffer.cc: (line 236): Size of recording buffer: 480
I/rtp_sender_audio.cc: (line 264): First audio RTP packet sent to pacer
I/port.cc: (line 1756): Conn[c73bb600:audio:Net[wlan0:192.168.0.x/24:Wifi:id=3]:zsC/qmro:1:0:local:udp:192.168.0.x:39034->gw+UvzLq:1:2122260223:local:udp:10.0.2.x:43315|C–I|-|0|0|9115038255631187454|-]: Sent STUN ping, id=4e4c4b786b51367162714962, use_candidate=0, nomination=0
I/probe_controller.cc: (line 335): kWaitingForProbingResult: timeout
I/reconnect Test: disconnectChannelListener.onDisconnectChannel()
I/RestServiceHandler: send Reconnect msg
I/WebSocketClient: doConnect() - connectStatus=RECONNECT
I/WebSocketClientHandler: handshakeFuture start
I/openssl_stream_adapter.cc: (line 905): Cleanup
I/dtls_transport.cc: (line 651): DtlsTransport[audio|1|_W]: DTLS transport closed
I/jsep_transport_controller.cc: (line 1249): Transport audio writability changed to 0.
I/channel.cc: (line 600): Channel not writable (audio)
I/srtp_transport.cc: (line 363): The params in SRTP transport are reset.
I/peer_connection.cc: (line 3954): Changing IceConnectionState 2 => 5
I/channel.cc: (line 867): Changing voice state, recv=1 send=1
I/WebSocketClientHandler: WebSocket Client connected!
I/WebSocketClientHandler: channelInactive() - WebSocket Client disconnected!
I/reconnect Test: disconnectChannelListener.onDisconnectChannel()
I/RestServiceHandler: send Reconnect msg
I/WebSocketClient: doConnect() - connectStatus=RECONNECT
I/peer_connection.cc: (line 4018): Session: 8518931667388194587 Old state: kStable New state: kClosed
I/audio_device_impl.cc: (line 827): StopRecording
I/audio_device_template.h: (line 209): StopRecording
I/audio_record_jni.cc: (line 178): StopRecording
I/WebRtcAudioRecord: stopRecording
I/WebRtcAudioRecord: stopThread
I/WebSocketClientHandler: channelInactive() - WebSocket Client disconnected!
I/WebRtcAudioEffects: release
I/WebRtcAudioRecord: releaseAudioResources
I/audio_device_buffer.cc: (line 151): StopRecording
I/audio_device_buffer.cc: (line 174): total recording time: 1399
I/audio_device_impl.cc: (line 831): output: 0
I/message_queue.cc: (line 518): Message took 209ms to dispatch. Posted from: ClearSend@…/…/…/…/usr/local/google/home/sakal/code/webrtc-aar-release/src/pc/rtp_sender.cc:532
I/webrtc_voice_engine.cc: (line 1972): SetOutputVolume() to 0 for recv stream with ssrc 3853270849
I/channel.cc: (line 567): Channel disabled
I/audio_device_impl.cc: (line 796): StopPlayout
I/audio_device_template.h: (line 194): Playing
I/audio_device_template.h: (line 188): StopPlayout
I/audio_track_jni.cc: (line 145): StopPlayout
I/WebRtcAudioTrack: stopPlayout
I/WebRtcAudioTrack: underrun count: 0
I/WebRtcAudioTrack: stopThread
I/WebRtcAudioTrack: Stopping the AudioTrackThread…
I/WebRtcAudioTrack: Calling AudioTrack.stop…
D/AudioTrack: stop(2152): called with 65760 frames delivered
I/WebRtcAudioTrack: AudioTrack.stop is done.
I/WebRtcAudioTrack: AudioTrackThread has now been stopped.
I/WebRtcAudioTrack: releaseAudioResources
I/audio_device_buffer.cc: (line 137): StopPlayout
I/audio_device_buffer.cc: (line 143): total playout time: 1549
I/audio_device_impl.cc: (line 800): output: 0
I/channel.cc: (line 867): Changing voice state, recv=0 send=0
I/webrtc_voice_engine.cc: (line 1833): RemoveSendStream: 2126543501
I/call.cc: (line 1104): UpdateAggregateNetworkState: aggregate_state=up
I/rtp_transport_controller_send.cc: (line 263): SignalNetworkState Up
I/audio_send_stream.cc: (line 172): ~AudioSendStream: 2126543501
I/webrtc_voice_engine.cc: (line 1910): RemoveRecvStream: 3853270849
I/call.cc: (line 1104): UpdateAggregateNetworkState: aggregate_state=down
I/rtp_transport_controller_send.cc: (line 263): SignalNetworkState Down
I/audio_receive_stream.cc: (line 135): ~AudioReceiveStream: 3853270849
I/paced_sender.cc: (line 115): PacedSender paused.
I/control_handler.cc: (line 97): Bitrate estimate state changed, BWE: 300 kbps.
I/channel.cc: (line 167): Destroyed channel: audio
I/openssl_stream_adapter.cc: (line 905): Cleanup
I/NetworkMonitor: Stop monitoring with native observer 505854362304
I/NetworkMonitorAutoDetect: Unregister network callback
I/NetworkMonitorAutoDetect: Unregister network callback
I/turn_port.cc: (line 1498): Port[b8c21500:audio:1:0:relay:Net[wlan0:192.168.0.x/24:Wifi:id=3]]: TURN refresh request sent, id=54792f79417177494e4e4b45
I/paced_sender.cc: (line 427): ProcessThreadAttached 0x0
I/paced_sender.cc: (line 427): ProcessThreadAttached 0x0
I/rtc_event_log_impl.cc: (line 210): Stopping WebRTC event log.
I/rtc_event_log_impl.cc: (line 227): WebRTC event log successfully stopped.
I/message_queue.cc: (line 518): Message took 255ms to dispatch. Posted from: Close@…/…/…/…/usr/local/google/home/sakal/code/webrtc-aar-release/src/api/peer_connection_proxy.h:139