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2020-12-03
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webrtc+ janus videocall 开发demo
问题: 呼叫端看不到被呼叫端视频, 怀疑被呼叫端的answer格式说步骤存在问题.
webrtc+ janus videocall 被呼叫端开发 执行日志 ------------------------------------------------------------------------------------------------------------------------------------------------
12/03 10:48:38: Launching 'janus-gateway-android-master-app' on HUAWEI EVA-AL00.
$ adb shell am start -n "org.appspot.apprtc/org.appspot.apprtc.ConnectActivity" -a android.intent.action.MAIN -c android.intent.category.LAUNCHER -D
Waiting for application to come online: org.appspot.apprtc.test | org.appspot.apprtc
Waiting for application to come online: org.appspot.apprtc.test | org.appspot.apprtc
Connected to process 17709 on device 'huawei-eva_al00-PBV7N16426019209'.
Capturing and displaying logcat messages from application. This behavior can be disabled in the "Logcat output" section of the "Debugger" settings page.
W/ActivityThread: Application org.appspot.apprtc is waiting for the debugger on port 8100...
I/System.out: Sending WAIT chunk
I/HwApiCacheMangerEx: apicache mCurPackageName=org.appspot.apprtc uptimes=63317499
I/HwApiCacheMangerEx: apicache oUid null
I/HwApiCacheMangerEx: apicache volumes null
I/HwApiCacheMangerEx: apicache path=/storage/emulated/0 state=mounted key=org.appspot.apprtc#10153#256
I/HwApiCacheMangerEx: apicache path=/storage/3138-6462 state=mounted key=org.appspot.apprtc#10153#256
I/HwApiCacheMangerEx: apicache oUid 10153
apicache volumes null
I/HwApiCacheMangerEx: apicache path=/storage/emulated/0 state=mounted key=org.appspot.apprtc#10153#0
apicache path=/storage/3138-6462 state=mounted key=org.appspot.apprtc#10153#0
apicache async read begin packageName=org.appspot.apprtc userid=0
apicache pi null
I/chatty: uid=10153(u0_a153) queued-work-loo identical 1 line
I/HwApiCacheMangerEx: apicache pi null
I/HwApiCacheMangerEx: apicache oUid null
apicache async read finished packageName=org.appspot.apprtc userid=0 totalus=2767
Waiting for application to come online: org.appspot.apprtc.test | org.appspot.apprtc
Connecting to org.appspot.apprtc
Connected to the target VM, address: 'localhost:8603', transport: 'socket'
Capturing and displaying logcat messages from application. This behavior can be disabled in the "Logcat output" section of the "Debugger" settings page.
I/: power log dlsym ok
I/zygote64: Late-enabling -Xcheck:jni
I/zygote64: Reinit property: dalvik.vm.checkjni= false
D/ActivityThread: ActivityThread,attachApplication
W/ActivityThread: Application org.appspot.apprtc is waiting for the debugger on port 8100...
I/System.out: Sending WAIT chunk
I/HwApiCacheMangerEx: apicache mCurPackageName=org.appspot.apprtc uptimes=63317499
I/HwApiCacheMangerEx: apicache oUid null
I/HwApiCacheMangerEx: apicache volumes null
I/HwApiCacheMangerEx: apicache path=/storage/emulated/0 state=mounted key=org.appspot.apprtc#10153#256
I/HwApiCacheMangerEx: apicache path=/storage/3138-6462 state=mounted key=org.appspot.apprtc#10153#256
I/HwApiCacheMangerEx: apicache oUid 10153
apicache volumes null
I/HwApiCacheMangerEx: apicache path=/storage/emulated/0 state=mounted key=org.appspot.apprtc#10153#0
apicache path=/storage/3138-6462 state=mounted key=org.appspot.apprtc#10153#0
apicache async read begin packageName=org.appspot.apprtc userid=0
apicache pi null
I/chatty: uid=10153(u0_a153) queued-work-loo identical 1 line
I/HwApiCacheMangerEx: apicache pi null
I/HwApiCacheMangerEx: apicache oUid null
apicache async read finished packageName=org.appspot.apprtc userid=0 totalus=2767
I/zygote64: Debugger is active
I/System.out: Debugger has connected
waiting for debugger to settle...
I/System.out: waiting for debugger to settle...
I/System.out: waiting for debugger to settle...
I/chatty: uid=10153(u0_a153) org.appspot.apprtc identical 2 lines
I/System.out: waiting for debugger to settle...
I/System.out: waiting for debugger to settle...
I/System.out: waiting for debugger to settle...
I/System.out: debugger has settled (1448)
D/HwFLClassLoader: get used feature list :/feature/used-list failed!
USE_FEATURE_LIST had not init!
I/HwApiCacheMangerEx: apicache pi null
E/Minikin: Could not get cmap table size!
E/MemoryLeakMonitorManager: MemoryLeakMonitor.jar is not exist!
I/HwCust: Constructor found for class android.app.HwCustActivityImpl
D/HwCust: Create obj success use class android.app.HwCustActivityImpl
V/HwPolicyFactory: : success to get AllImpl object and return....
I/HwCust: Constructor found for class android.app.HwCustHwWallpaperManagerImpl
D/HwCust: Create obj success use class android.app.HwCustHwWallpaperManagerImpl
V/HwWidgetFactory: : successes to get AllImpl object and return....
V/ActivityThread: ActivityThread,callActivityOnCreate
D/CubicBezierInterpolator: CubicBezierInterpolator mControlPoint1x = 0.23, mControlPoint1y = 0.06, mControlPoint2x = 0.09, mControlPoint2y = 0.97
D/CubicBezierInterpolator: CubicBezierInterpolator mControlPoint1x = 0.6, mControlPoint1y = 0.9, mControlPoint2x = 0.8, mControlPoint2y = 1.0
D/CubicBezierInterpolator: CubicBezierInterpolator mControlPoint1x = 0.23, mControlPoint1y = 0.06, mControlPoint2x = 0.09, mControlPoint2y = 0.97
D/CubicBezierInterpolator: CubicBezierInterpolator mControlPoint1x = 0.6, mControlPoint1y = 0.9, mControlPoint2x = 0.8, mControlPoint2y = 1.0
D/HwGalleryCacheManagerImpl: mIsEffect:false
D/HwRTBlurUtils: check blur style for HwPhoneWindow, themeResId : 0x7f0e00aa, context : org.appspot.apprtc.ConnectActivity@950ee65, Nhwext : 0, get Blur : disable with , null
W/WindowDecorActionBar: should not do the transition or the transition anim is null or it is running or the mContainer view is null or mContainer view hasn't been drawn to screen
E/HwResourcesImpl: readDefaultConfig : FileNotFoundException
I/HwCust: Constructor found for class android.content.res.HwCustHwResourcesImpl
D/HwCust: Create obj success use class android.content.res.HwCustHwResourcesImpl
D/ActivityThread: add activity client record, r= ActivityRecord{e7b0ac8 token=android.os.BinderProxy@37abefc {org.appspot.apprtc/org.appspot.apprtc.ConnectActivity}} token= android.os.BinderProxy@37abefc
W/VRSystemServiceManager: vr service is not alive
D/OpenGLRenderer: HWUI Binary is enabled
D/OpenGLRenderer: HWUI GL Pipeline
I/PressGestureDetector: onAttached begin
onAttached end
I/PressGestureDetector: enabledInPad = false,isPcCastMode = false
I/zygote64: Do partial code cache collection, code=22KB, data=29KB
After code cache collection, code=22KB, data=29KB
Increasing code cache capacity to 128KB
I/zygote64: Do partial code cache collection, code=22KB, data=49KB
After code cache collection, code=22KB, data=49KB
Increasing code cache capacity to 256KB
Compiler allocated 8MB to compile void android.widget.TextView.<init>(android.content.Context, android.util.AttributeSet, int, int)
I/OpenGLRenderer: Initialized EGL, version 1.4
D/OpenGLRenderer: Swap behavior 2
D/mali_winsys: EGLint new_window_surface(egl_winsys_display *, void *, EGLSurface, EGLConfig, egl_winsys_surface **, egl_color_buffer_format *, EGLBoolean) returns 0x3000
D/BinaryProgramHWUI: Key: 0x3c80000000001 has not found in mBinaryEntries, Compile it.
W/InputMethodManager: startInputReason = 1
W/InputMethodManager: startInputReason = 5
I/hwaps: JNI_OnLoad
W/Settings: mValues not put! needsGenerationTracker: true currentGeneration: -1 name: enable_navbar value: null
V/AudioManager: playSoundEffect effectType: 0
I/System.out: 视频回环....
D/ConnectActivity: Connecting to room 47307352 at URL wss://janus.conf.meetecho.com/ws
D/HwCust: Create obj success use class android.app.HwCustActivityImpl
D/HwCust: Create obj success use class android.app.HwCustHwWallpaperManagerImpl
V/ActivityThread: ActivityThread,callActivityOnCreate
D/HwRTBlurUtils: check blur style for HwPhoneWindow, themeResId : 0x7f0e00a6, context : org.appspot.apprtc.EchoTestActivity3@f8162e1, Nhwext : 0, get Blur : disable with , null
I/org.webrtc.Logging: EglBase14: SDK version: 26. isEGL14Supported: true
I/org.webrtc.Logging: EglRenderer: pip_video_viewInitializing EglRenderer
I/org.webrtc.Logging: EglRenderer: pip_video_viewEglBase.create shared context
EglBase14: SDK version: 26. isEGL14Supported: true
I/org.webrtc.Logging: EglRenderer: fullscreen_video_viewInitializing EglRenderer
I/org.webrtc.Logging: EglRenderer: fullscreen_video_viewEglBase.create shared context
EglBase14: SDK version: 26. isEGL14Supported: true
I/org.webrtc.Logging: CallRTCClient: setSwappedFeeds: true
EglRenderer: fullscreen_video_viewsetMirror: true
D/CallRTCClient: Room ID: 47307352
VIDEO_FILE: 'null'
I/CallActivity: roomUri=wss://janus.conf.meetecho.com/ws
roomid=47307352
urlParameters=null
I/zygote64: Do full code cache collection, code=92KB, data=78KB
I/zygote64: After code cache collection, code=89KB, data=61KB
D/PCRTCClient: Preferred video codec: VP8
D/PCRTCClient: Initialize WebRTC. Field trials: WebRTC-IntelVP8/Enabled/
D/AppRTCAudioManager: ctor
D/AppRTCBluetoothManager: create@[name=main, id=1]
ctor
I/org.webrtc.Logging: NativeLibrary: Loading native library: jingle_peerconnection_so
I/org.webrtc.Logging: NativeLibrary: Loading library: jingle_peerconnection_so
D/WebSocketChannelClient: Connecting WebSocket to(连接WebSocket...): wss://janus.conf.meetecho.com/ws
D/AppRTCAudioManager: useSpeakerphone: auto
D/AppRTCProximitySensor: AppRTCProximitySensor@[name=main, id=1]
D/de.tavendo.autobahn.WebSocketConnection: WebSocket connection created.
D/AppRTCAudioManager: defaultAudioDevice: SPEAKER_PHONE
D/AppRTCAudioManager: Android SDK: 26, Release: 8.0.0, Brand: HUAWEI, Device: HWEVA, Id: HUAWEIEVA-AL00, Hardware: hi3650, Manufacturer: HUAWEI, Model: EVA-AL00, Product: EVA-AL00
D/CallRTCClient: Starting the audio manager...
D/NetworkSecurityConfig: No Network Security Config specified, using platform default
D/AppRTCAudioManager: start
D/AppRTCAudioManager: AudioManager starts...
I/AudioSystem-JNI: listAudioPorts AudioSystem::listAudioPorts numPorts 8 generation 30 generation1 30
I/org.webrtc.Logging: PeerConnectionFactory: PeerConnectionFactory was initialized without an injected Loggable. Any existing Loggable will be deleted.
W/PCRTCClient: External OpenSLES ADM not implemented yet.
V/AudioManager: requestAudioFocus streamType: 0 durationHint: 2
V/AudioManager: registerAudioFocusListener...
D/AppRTCAudioManager: Audio focus request granted for VOICE_CALL streams
V/AudioManager: setMode mode: 3
V/AudioManager: getProperty key: android.media.property.OUTPUT_SAMPLE_RATE
I/org.webrtc.Logging: WebRtcAudioManagerExternal: Sample rate is set to 48000 Hz
E/org.webrtc.Logging: JavaAudioDeviceModule: HW AEC not supported
JavaAudioDeviceModule: HW NS not supported
I/org.webrtc.Logging: JavaAudioDeviceModule: createAudioDeviceModule
I/org.webrtc.Logging: JavaAudioDeviceModule: HW NS will not be used.
JavaAudioDeviceModule: HW AEC will not be used.
WebRtcAudioEffectsExternal: ctor@[name=pool-1-thread-1, id=1819]
D/PCRTCClient: Factory networkIgnoreMask option: 0
I/AudioManager: setMode mode:3
D/AppRTCBluetoothManager: start
V/AudioManager: isBluetoothScoAvailableOffCall...
isScoAvailableOffCall
W/VideoCapabilities: Unrecognized profile/level 1/32 for video/mp4v-es
I/VideoCapabilities: Unsupported profile 16384 for video/mp4v-es
Unsupported profile 16384 for video/mp4v-es
W/VideoCapabilities: Unsupported mime video/x-pn-realvideo
D/AppRTCBluetoothManager: BluetoothAdapter: enabled=false, state=OFF, name=HUAWEI P9, address=02:00:00:00:00:00
W/VideoCapabilities: Unsupported mime video/mpeg
W/VideoCapabilities: Unrecognized profile/level 0/0 for video/mpeg2
D/AppRTCBluetoothManager: HEADSET profile state: DISCONNECTED
Bluetooth proxy for headset profile has started
W/VideoCapabilities: Unrecognized profile/level 0/2 for video/mpeg2
D/AppRTCBluetoothManager: start done: BT state=HEADSET_UNAVAILABLE
W/VideoCapabilities: Unrecognized profile/level 0/3 for video/mpeg2
D/AppRTCAudioManager: --- updateAudioDeviceState: wired headset=false, BT state=HEADSET_UNAVAILABLE
Device status: available=[], selected=NONE, user selected=NONE
D/AppRTCAudioManager: setAudioDeviceInternal(device=SPEAKER_PHONE)
V/AudioManager: setSpeakerphoneOn on: true
W/VideoCapabilities: Unrecognized profile/level 32768/2 for video/mp4v-es
I/AudioManager: setSpeakerphoneOn on:true
D/AppRTCAudioManager: New device status: available=[EARPIECE, SPEAKER_PHONE], selected=SPEAKER_PHONE
D/CallRTCClient: onAudioManagerDevicesChanged: [EARPIECE, SPEAKER_PHONE], selected: SPEAKER_PHONE
D/AppRTCAudioManager: --- updateAudioDeviceState done
D/AppRTCAudioManager: AudioManager started
W/VideoCapabilities: Unsupported mime video/vc1
I/VideoCapabilities: Unsupported profile 4 for video/mp4v-es
D/ActivityThread: add activity client record, r= ActivityRecord{fcdb23d token=android.os.BinderProxy@ee4d23a {org.appspot.apprtc/org.appspot.apprtc.EchoTestActivity3}} token= android.os.BinderProxy@ee4d23a
W/VRSystemServiceManager: vr service is not alive
D/OpenGLRenderer: HWUI Binary is enabled
I/org.webrtc.Logging: PeerConnectionFactory: onNetworkThreadReady
I/org.webrtc.Logging: PeerConnectionFactory: onWorkerThreadReady
I/org.webrtc.Logging: PeerConnectionFactory: onSignalingThreadReady
D/PCRTCClient: Peer connection factory created.
I/Choreographer: Skipped 39 frames! The application may be doing too much work on its main thread.
I/PressGestureDetector: onAttached begin
onAttached end
I/PressGestureDetector: enabledInPad = false,isPcCastMode = false
I/org.webrtc.Logging: SurfaceViewRenderer: fullscreen_video_view: onMeasure(). New size: 1080x1719
SurfaceViewRenderer: pip_video_view: onMeasure(). New size: 1032x432
SurfaceViewRenderer: pip_video_view2: onMeasure(). New size: 984x432
I/org.webrtc.Logging: SurfaceViewRenderer: fullscreen_video_view: onMeasure(). New size: 1080x1920
SurfaceViewRenderer: pip_video_view: onMeasure(). New size: 1032x432
D/mali_winsys: EGLint new_window_surface(egl_winsys_display *, void *, EGLSurface, EGLConfig, egl_winsys_surface **, egl_color_buffer_format *, EGLBoolean) returns 0x3000
I/org.webrtc.Logging: SurfaceViewRenderer: pip_video_view2: onMeasure(). New size: 984x432
I/org.webrtc.Logging: EglRenderer: fullscreen_video_viewsetLayoutAspectRatio: 0.5625
EglRenderer: pip_video_viewsetLayoutAspectRatio: 2.3888888
EglRenderer: pip_video_view2setLayoutAspectRatio: 2.2777777
I/org.webrtc.Logging: SurfaceEglRenderer: fullscreen_video_view: surfaceChanged: format: 4 size: 1080x1920
D/mali_winsys: EGLint new_window_surface(egl_winsys_display *, void *, EGLSurface, EGLConfig, egl_winsys_surface **, egl_color_buffer_format *, EGLBoolean) returns 0x3000
I/org.webrtc.Logging: SurfaceEglRenderer: pip_video_view: surfaceChanged: format: 4 size: 1032x432
D/mali_winsys: EGLint new_window_surface(egl_winsys_display *, void *, EGLSurface, EGLConfig, egl_winsys_surface **, egl_color_buffer_format *, EGLBoolean) returns 0x3000
I/org.webrtc.Logging: SurfaceEglRenderer: pip_video_view2: surfaceChanged: format: 4 size: 984x432
D/AppRTCAudioManager: WiredHeadsetReceiver.onReceive@[name=main, id=1]: a=android.intent.action.HEADSET_PLUG, s=unplugged, m=mic, n=null, sb=true
--- updateAudioDeviceState: wired headset=false, BT state=HEADSET_UNAVAILABLE
Device status: available=[EARPIECE, SPEAKER_PHONE], selected=SPEAKER_PHONE, user selected=NONE
D/AppRTCAudioManager: --- updateAudioDeviceState done
D/OpenGLRenderer: HWUI Binary is enabled
W/InputMethodManager: startInputReason = 1
W/libEGL: EGLNativeWindowType 0x73b29cc010 disconnect failed
D/de.tavendo.autobahn.WebSocketReader: WebSocket reader created.
D/de.tavendo.autobahn.WebSocketConnection: WebSocket reader created and started.
D/de.tavendo.autobahn.WebSocketWriter: WebSocket writer created.
D/de.tavendo.autobahn.WebSocketReader: WebSocker reader running.
D/de.tavendo.autobahn.WebSocketWriter: WebSocker writer running.
D/de.tavendo.autobahn.WebSocketConnection: WebSocket writer created and started.
I/websock连接: -===================================================================
E/WSRTCClient: 创建session会话
E/WSRTCClient: Sending Create Janus in non connected state.
W/de.tavendo.autobahn.WebSocketReader: Status: 101 (Switching Protocols)
D/de.tavendo.autobahn.WebSocketConnection: opening handshake received
D/WebSocketChannelClient: WebSocket连接
WebSocket connection opened to: wss://janus.conf.meetecho.com/ws
I/WSRTCClient: onWebsocketOpen..
I/AppRTCUtils: randomPoz:25 charSetLen:62
I/AppRTCUtils: randomPoz:40 charSetLen:62
randomPoz:43 charSetLen:62
randomPoz:21 charSetLen:62
randomPoz:24 charSetLen:62
I/AppRTCUtils: randomPoz:51 charSetLen:62
randomPoz:54 charSetLen:62
randomPoz:17 charSetLen:62
I/AppRTCUtils: randomPoz:5 charSetLen:62
randomPoz:60 charSetLen:62
randomPoz:32 charSetLen:62
randomPoz:57 charSetLen:62
D/WSRTCClient: sending create msg to Janus . transactionID: nullZorVYz2RF8g5
D/WebSocketChannelClient: C->WSS: {"janus":"create","transaction":"nullZorVYz2RF8g5"}
D/WebSocketChannelClient: WSS->C: {
"janus": "success",
"transaction": "nullZorVYz2RF8g5",
"data": {
"id": 6886760364035320
}
}
I/WSRTCClient: Got wsmsg:{
"janus": "success",
"transaction": "nullZorVYz2RF8g5",
"data": {
"id": 6886760364035320
}
}
I/回环: ========================={"janus":"success","transaction":"nullZorVYz2RF8g5","data":{"id":6886760364035320}}
I/session连接: -===================================================================
I/AppRTCUtils: randomPoz:38 charSetLen:62
I/AppRTCUtils: randomPoz:11 charSetLen:62
randomPoz:25 charSetLen:62
randomPoz:15 charSetLen:62
randomPoz:58 charSetLen:62
I/AppRTCUtils: randomPoz:32 charSetLen:62
randomPoz:22 charSetLen:62
randomPoz:30 charSetLen:62
randomPoz:8 charSetLen:62
randomPoz:26 charSetLen:62
I/AppRTCUtils: randomPoz:19 charSetLen:62
randomPoz:53 charSetLen:62
D/WSRTCClient: publisherCreateHandle transactionID: nullmLZP6gWeIaT1
I/加载插件: -===================================================================
D/WSRTCClient: C->WSS: {"janus":"attach","plugin":"janus.plugin.videocall","transaction":"nullmLZP6gWeIaT1","session_id":6886760364035320,"opaque_id":"register-CwV1XT0CyCyF"}
D/WebSocketChannelClient: C->WSS: {"janus":"attach","plugin":"janus.plugin.videocall","transaction":"nullmLZP6gWeIaT1","session_id":6886760364035320,"opaque_id":"register-CwV1XT0CyCyF"}
I/AppRTCUtils: randomPoz:1 charSetLen:62
randomPoz:59 charSetLen:62
randomPoz:15 charSetLen:62
randomPoz:6 charSetLen:62
I/AppRTCUtils: randomPoz:5 charSetLen:62
randomPoz:44 charSetLen:62
randomPoz:6 charSetLen:62
randomPoz:35 charSetLen:62
randomPoz:60 charSetLen:62
I/AppRTCUtils: randomPoz:46 charSetLen:62
randomPoz:25 charSetLen:62
randomPoz:38 charSetLen:62
D/WebSocketChannelClient: C->WSS: {"janus":"keepalive","session_id":6886760364035320,"transaction":"nullB7PGFsGj8uZm"}
I/org.webrtc.Logging: EglRenderer: pip_video_viewDuration: 4016 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
I/org.webrtc.Logging: EglRenderer: fullscreen_video_viewDuration: 4013 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
D/WebSocketChannelClient: WSS->C: {
"janus": "success",
"session_id": 6886760364035320,
"transaction": "nullmLZP6gWeIaT1",
"data": {
"id": 5790718854272780
}
}
I/WSRTCClient: Got wsmsg:{
"janus": "success",
"session_id": 6886760364035320,
"transaction": "nullmLZP6gWeIaT1",
"data": {
"id": 5790718854272780
}
}
I/回环: ========================={"janus":"success","session_id":6886760364035320,"transaction":"nullmLZP6gWeIaT1","data":{"id":5790718854272780}}
I/AppRTCUtils: randomPoz:9 charSetLen:62
I/AppRTCUtils: randomPoz:20 charSetLen:62
randomPoz:51 charSetLen:62
randomPoz:4 charSetLen:62
I/AppRTCUtils: randomPoz:15 charSetLen:62
randomPoz:21 charSetLen:62
randomPoz:56 charSetLen:62
randomPoz:28 charSetLen:62
randomPoz:22 charSetLen:62
I/AppRTCUtils: randomPoz:1 charSetLen:62
randomPoz:25 charSetLen:62
randomPoz:11 charSetLen:62
D/WSRTCClient: publisherJoinRoom transactionID: nullJUzEPV4cWBZL
D/WSRTCClient: C->WSS: {"janus":"message","session_id":6886760364035320,"handle_id":5790718854272780,"transaction":"nullJUzEPV4cWBZL","body":{"request":"register","username":"65432100"}}
D/WSRTCClient: C->WSS: {"janus":"message","session_id":6886760364035320,"handle_id":5790718854272780,"transaction":"nullJUzEPV4cWBZL","body":{"request":"register","username":"65432100"}}
D/WebSocketChannelClient: C->WSS: {"janus":"message","session_id":6886760364035320,"handle_id":5790718854272780,"transaction":"nullJUzEPV4cWBZL","body":{"request":"register","username":"65432100"}}
D/WebSocketChannelClient: WSS->C: {
"janus": "ack",
"session_id": 6886760364035320,
"transaction": "nullB7PGFsGj8uZm"
}
I/WSRTCClient: Got wsmsg:{
"janus": "ack",
"session_id": 6886760364035320,
"transaction": "nullB7PGFsGj8uZm"
}
I/回环: ========================={"janus":"ack","session_id":6886760364035320,"transaction":"nullB7PGFsGj8uZm"}
I/WSRTCClient: Got an ack on session 6886760364035320
D/WebSocketChannelClient: WSS->C: {
"janus": "ack",
"session_id": 6886760364035320,
"transaction": "nullJUzEPV4cWBZL"
}
I/WSRTCClient: Got wsmsg:{
"janus": "ack",
"session_id": 6886760364035320,
"transaction": "nullJUzEPV4cWBZL"
}
I/回环: ========================={"janus":"ack","session_id":6886760364035320,"transaction":"nullJUzEPV4cWBZL"}
I/WSRTCClient: Got an ack on session 6886760364035320
D/WebSocketChannelClient: WSS->C: {
"janus": "event",
"session_id": 6886760364035320,
"transaction": "nullJUzEPV4cWBZL",
"sender": 5790718854272780,
"plugindata": {
"plugin": "janus.plugin.videocall",
"data": {
"videocall": "event",
"result": {
"event": "registered",
"username": "65432100"
}
}
}
}
I/WSRTCClient: Got wsmsg:{
"janus": "event",
"session_id": 6886760364035320,
"transaction": "nullJUzEPV4cWBZL",
"sender": 5790718854272780,
"plugindata": {
"plugin": "janus.plugin.videocall",
"data": {
"videocall": "event",
"result": {
"event": "registered",
"username": "65432100"
}
}
}
}
I/回环: ========================={"janus":"event","session_id":6886760364035320,"transaction":"nullJUzEPV4cWBZL","sender":5790718854272780,"plugindata":{"plugin":"janus.plugin.videocall","data":{"videocall":"event","result":{"event":"registered","username":"65432100"}}}}
D/WSRTCClient: Got a plugin event on session 6886760364035320
I/org.webrtc.Logging: EglRenderer: pip_video_viewDuration: 4019 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
I/org.webrtc.Logging: EglRenderer: fullscreen_video_viewDuration: 4017 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
I/org.webrtc.Logging: EglRenderer: pip_video_viewDuration: 4019 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
I/org.webrtc.Logging: EglRenderer: fullscreen_video_viewDuration: 4017 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
I/org.webrtc.Logging: EglRenderer: pip_video_viewDuration: 4017 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
I/org.webrtc.Logging: EglRenderer: pip_video_viewDuration: 4018 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
I/org.webrtc.Logging: EglRenderer: fullscreen_video_viewDuration: 4018 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
I/org.webrtc.Logging: EglRenderer: pip_video_viewDuration: 4018 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
I/org.webrtc.Logging: EglRenderer: fullscreen_video_viewDuration: 4020 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
I/org.webrtc.Logging: EglRenderer: pip_video_viewDuration: 4020 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
D/WebSocketChannelClient: WSS->C: {
"janus": "event",
"session_id": 6886760364035320,
"sender": 5790718854272780,
"plugindata": {
"plugin": "janus.plugin.videocall",
"data": {
"videocall": "event",
"result": {
"event": "incomingcall",
"username": "123456"
}
}
},
"jsep": {
"type": "offer",
"sdp": "v=0\r\no=- 3456360543492079365 2 IN IP4 188.213.167.189\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video data\r\na=msid-semantic: WMS janus\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 188.213.167.189\r\na=sendrecv\r\na=mid:audio\r\na=rtcp-mux\r\na=ice-ufrag:Kpy0\r\na=ice-pwd:D/X/n7YGkKcv51TAboN18d\r\na=ice-options:trickle\r\na=fingerprint:sha-256 17:E8:3E:77:2B:91:A1:00:D9:C7:C9:E9:13:47:6F:61:90:17:D9:33:CC:18:99:6D:77:2A:21:0E:F3:2D:B3:3E\r\na=setup:actpass\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000\r\na=rtpmap:113 telephone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\na=msid:janus janusa0\r\na=ssrc:3245288059 cname:janus\r\na=ssrc:3245288059 msid:janus janusa0\r\na=ssrc:3245288059 mslabel:janus\r\na=ssrc:3245288059 label:janusa0\r\na=candidate:1 1 udp 2015363327 188.213.167.189 45290 typ host\r\na=end-of-candidates\r\nm=video 9 UDP/TLS/RTP/SAVPF 96 98 100 102 127 125 108 124 123 97 99 101 103 104 126 109 105 106\r\nc=IN IP4 188.213.167.189\r\na=sendrecv\r\na=mid:video\r\na=rtcp-mux\r\na=ice-ufrag:Kpy0\r\na=ice-pwd:D/X/n7YGkKcv51TAboN18d\r\na=ice-options:trickle\r\na=fingerprint:sha-256 17:E8:3E:77:2B:91:A1:00:D9:C7:C9:E9:13:47:6F:61:90:17:D9:33:CC:18:99:6D:77:2A:21:0E:F3:2D:B3:3E\r\na=setup:actpass\r\na=extmap:14 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:13 urn:3gpp:video-orientation\r\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07\r\na=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=rtpmap:96 VP8/90000\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtpmap:98 VP9/90000\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=fmtp:98 profile-id=0\r\na=rtpmap:100 VP9/90000\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=fmtp:100 profile-id=2\r\na=rtpmap:102 H264/90000\r\na=rtcp-fb:102 goog-remb\r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\r\na=rtpmap:127 H264/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f\r\na=rtpmap:125 H264/90000\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na
I/WSRTCClient: Got wsmsg:{
"janus": "event",
"session_id": 6886760364035320,
"sender": 5790718854272780,
"plugindata": {
"plugin": "janus.plugin.videocall",
"data": {
"videocall": "event",
"result": {
"event": "incomingcall",
"username": "123456"
}
}
},
"jsep": {
"type": "offer",
"sdp": "v=0\r\no=- 3456360543492079365 2 IN IP4 188.213.167.189\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video data\r\na=msid-semantic: WMS janus\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 188.213.167.189\r\na=sendrecv\r\na=mid:audio\r\na=rtcp-mux\r\na=ice-ufrag:Kpy0\r\na=ice-pwd:D/X/n7YGkKcv51TAboN18d\r\na=ice-options:trickle\r\na=fingerprint:sha-256 17:E8:3E:77:2B:91:A1:00:D9:C7:C9:E9:13:47:6F:61:90:17:D9:33:CC:18:99:6D:77:2A:21:0E:F3:2D:B3:3E\r\na=setup:actpass\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000\r\na=rtpmap:113 telephone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\na=msid:janus janusa0\r\na=ssrc:3245288059 cname:janus\r\na=ssrc:3245288059 msid:janus janusa0\r\na=ssrc:3245288059 mslabel:janus\r\na=ssrc:3245288059 label:janusa0\r\na=candidate:1 1 udp 2015363327 188.213.167.189 45290 typ host\r\na=end-of-candidates\r\nm=video 9 UDP/TLS/RTP/SAVPF 96 98 100 102 127 125 108 124 123 97 99 101 103 104 126 109 105 106\r\nc=IN IP4 188.213.167.189\r\na=sendrecv\r\na=mid:video\r\na=rtcp-mux\r\na=ice-ufrag:Kpy0\r\na=ice-pwd:D/X/n7YGkKcv51TAboN18d\r\na=ice-options:trickle\r\na=fingerprint:sha-256 17:E8:3E:77:2B:91:A1:00:D9:C7:C9:E9:13:47:6F:61:90:17:D9:33:CC:18:99:6D:77:2A:21:0E:F3:2D:B3:3E\r\na=setup:actpass\r\na=extmap:14 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:13 urn:3gpp:video-orientation\r\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07\r\na=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=rtpmap:96 VP8/90000\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtpmap:98 VP9/90000\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=fmtp:98 profile-id=0\r\na=rtpmap:100 VP9/90000\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=fmtp:100 profile-id=2\r\na=rtpmap:102 H264/90000\r\na=rtcp-fb:102 goog-remb\r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\r\na=rtpmap:127 H264/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f\r\na=rtpmap:125 H264/90000\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:
I/回环: ========================={"janus":"event","session_id":6886760364035320,"sender":5790718854272780,"plugindata":{"plugin":"janus.plugin.videocall","data":{"videocall":"event","result":{"event":"incomingcall","username":"123456"}}},"jsep":{"type":"offer","sdp":"v=0\r\no=- 3456360543492079365 2 IN IP4 188.213.167.189\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video data\r\na=msid-semantic: WMS janus\r\nm=audio 9 UDP\/TLS\/RTP\/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 188.213.167.189\r\na=sendrecv\r\na=mid:audio\r\na=rtcp-mux\r\na=ice-ufrag:Kpy0\r\na=ice-pwd:D\/X\/n7YGkKcv51TAboN18d\r\na=ice-options:trickle\r\na=fingerprint:sha-256 17:E8:3E:77:2B:91:A1:00:D9:C7:C9:E9:13:47:6F:61:90:17:D9:33:CC:18:99:6D:77:2A:21:0E:F3:2D:B3:3E\r\na=setup:actpass\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:2 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/abs-send-time\r\na=extmap:3 http:\/\/www.ietf.org\/id\/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=rtpmap:111 opus\/48000\/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC\/16000\r\na=rtpmap:104 ISAC\/32000\r\na=rtpmap:9 G722\/8000\r\na=rtpmap:0 PCMU\/8000\r\na=rtpmap:8 PCMA\/8000\r\na=rtpmap:106 CN\/32000\r\na=rtpmap:105 CN\/16000\r\na=rtpmap:13 CN\/8000\r\na=rtpmap:110 telephone-event\/48000\r\na=rtpmap:112 telephone-event\/32000\r\na=rtpmap:113 telephone-event\/16000\r\na=rtpmap:126 telephone-event\/8000\r\na=msid:janus janusa0\r\na=ssrc:3245288059 cname:janus\r\na=ssrc:3245288059 msid:janus janusa0\r\na=ssrc:3245288059 mslabel:janus\r\na=ssrc:3245288059 label:janusa0\r\na=candidate:1 1 udp 2015363327 188.213.167.189 45290 typ host\r\na=end-of-candidates\r\nm=video 9 UDP\/TLS\/RTP\/SAVPF 96 98 100 102 127 125 108 124 123 97 99 101 103 104 126 109 105 106\r\nc=IN IP4 188.213.167.189\r\na=sendrecv\r\na=mid:video\r\na=rtcp-mux\r\na=ice-ufrag:Kpy0\r\na=ice-pwd:D\/X\/n7YGkKcv51TAboN18d\r\na=ice-options:trickle\r\na=fingerprint:sha-256 17:E8:3E:77:2B:91:A1:00:D9:C7:C9:E9:13:47:6F:61:90:17:D9:33:CC:18:99:6D:77:2A:21:0E:F3:2D:B3:3E\r\na=setup:actpass\r\na=extmap:14 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:2 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/abs-send-time\r\na=extmap:13 urn:3gpp:video-orientation\r\na=extmap:3 http:\/\/www.ietf.org\/id\/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:12 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/playout-delay\r\na=extmap:11 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/video-content-type\r\na=extmap:7 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/video-timing\r\na=extmap:8 http:\/\/tools.ietf.org\/html\/draft-ietf-avtext-framemarking-07\r\na=extmap:9 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/color-space\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=rtpmap:96 VP8\/90000\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtpmap:98 VP9\/90000\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=fmtp:98 profile-id=0\r\na=rtpmap:100 VP9\/90000\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=fmtp:100 profile-id=2\r\na=rtpmap:102 H264\/90000\r\na=rtcp-fb:102 goog-remb\r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\r\na=rtpmap:127 H264\/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f\r\na=rtpmap:125 H264\/90000\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=fmtp:125 level-asymmetry-all
D/WSRTCClient: Got a plugin event on session 6886760364035320
D/CallRTCClient: 登录--------->应答
D/HwRTBlurUtils: check blur style for HwToast-Toast, themeResId : 0x7f0e00a6, context : org.appspot.apprtc.EchoTestActivity3@f8162e1, Nhwext : 6, get Blur : disable with , android.graphics.drawable.NinePatchDrawable@356d5bd
I/CameraManagerGlobal: Connecting to camera service
I/CameraManagerGlobal: do not need hide aux camera, device number: 0
I/CameraManagerGlobal: do not need hide aux camera, device number: 1
do not need hide aux camera, device number: 2
I/CameraManagerGlobal: do not need hide aux camera, device number: 2
I/org.webrtc.Logging: CallRTCClient: Creating capturer using camera2 API.
I/CameraManagerGlobal: do not need hide aux camera, device number: 2
I/org.webrtc.Logging: CallRTCClient: Looking for front facing cameras.
I/org.webrtc.Logging: CallRTCClient: Creating front facing camera capturer.
I/CameraManagerGlobal: do not need hide aux camera, device number: 2
W/VRSystemServiceManager: vr service is not alive
I/org.webrtc.Logging: PCRTCClient: Capturing format: 1280x720@30
D/PCRTCClient: Create peer connection.
createPeerConnectioning...
Create peer connection.
D/PeerConnectionFactory: nativeCreatePeerConnectioning....
D/OpenGLRenderer: HWUI Binary is enabled
D/OpenGLRenderer: HWUI Binary is enabled
D/PCRTCClient: We are putting handleId=5790718854272780
D/PCRTCClient: Peer connection created.
W/ViewRootImpl[Toast]: EGLdebug relayoutWindow Surface isSurface(name=null) (appName =org.appspot.apprtc) ( mSurfaceControllerIsValid =false) (mNativeObject =0)/@0x4095ac3,viewVisibility is0
I/EglBase14: SDK version: 26. isEGL14Supported: true
I/opensslidentity.cc: (line 93): Returning key pair
I/opensslcertificate.cc: (line 58): Making certificate for WebRTC
E/BufferQueueProducer: [] Can not get hwsched service
I/androidvideotracksource.cc: (line 32): AndroidVideoTrackSource ctor
I/CameraCapturer: startCapture: 1280x720@30
I/paced_sender.cc: (line 104): PacedSender paused.
I/Camera2Session: Create new camera2 session on camera 1
I/Camera2Session: start
I/opensslcertificate.cc: (line 105): Returning certificate
W/ViewRootImpl[Toast]: EGLdebug relayoutWindow Surface isSurface(name=null) (appName =org.appspot.apprtc) ( mSurfaceControllerIsValid =true) (mNativeObject =73a3ee1000)/@0x4095ac3,relayoutResult is7
W/ViewRootImpl[Toast]: EGLdebug Surface isSurface(name=null) (appName =org.appspot.apprtc) ( mSurfaceControllerIsValid =true) (mNativeObject =73a3ee1000)/@0x4095ac3
D/mali_winsys: EGLint new_window_surface(egl_winsys_display *, void *, EGLSurface, EGLConfig, egl_winsys_surface **, egl_color_buffer_format *, EGLBoolean) returns 0x3000
I/peerconnection.cc: (line 1264): Adding video transceiver in response to a call to AddTrack.
E/rtpsender.cc: (line 468): Tried to get DTMF sender from video sender.
I/peerconnection.cc: (line 1264): Adding audio transceiver in response to a call to AddTrack.
E/rtpsender.cc: (line 468): Tried to get DTMF sender from video sender.
D/PCRTCClient: Found video sender.org.webrtc.RtpSender@3a940b8
D/PCRTCClient: Peer connection created.org.webrtc.PeerConnection@718ce91
Create peer connection.
D/PeerConnectionFactory: nativeCreatePeerConnectioning....
I/Camera2Session: Available preview sizes: [3264x2448, 2448x2448, 3264x1840, 1920x1080, 1440x1080, 1536x864, 1280x960, 1280x720, 960x720, 720x720, 640x480, 736x414, 544x408, 400x400, 352x288, 320x240, 208x144, 176x144]
I/peerconnectionfactory.cc: (line 496): Using default network controller factory
I/alr_experiment.cc: (line 65): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR start bandwidth usage percent: 80, ALR end budget level percent: 40, ALR end budget level percent: -60, ALR experiment group ID: 3
I/bitrate_prober.cc: (line 64): Bandwidth probing enabled, set to inactive
I/rtp_transport_controller_send.cc: (line 48): Using Legacy SSCC
I/aimd_rate_control.cc: (line 108): Using aimd rate control with back off factor 0.85
I/delay_based_bwe.cc: (line 103): Using Trendline filter for delay change estimation with window size 20
I/send_side_congestion_controller.cc: (line 308): SignalNetworkState Down
I/delay_based_bwe.cc: (line 311): BWE Setting start bitrate to: 300000
I/paced_sender.cc: (line 362): ProcessThreadAttached 0x0x73bf041580
I/Camera2Session: Available fps ranges: [[14.0:30.0], [30.0:30.0], [14.0:20.0], [20.0:20.0], [14.0:25.0], [25.0:25.0], [12.0:15.0], [15.0:15.0]]
I/aimd_rate_control.cc: (line 108): Using aimd rate control with back off factor 0.85
I/remote_bitrate_estimator_single_stream.cc: (line 55): RemoteBitrateEstimatorSingleStream: Instantiating.
I/peerconnection.cc: (line 4735): TCP candidates are disabled.
D/PCRTCClient: We are putting handleId=5790718854272780
Peer connection created.
PC create ANSWER
I/opensslidentity.cc: (line 45): Making key pair
I/Camera2Session: Using capture format: 1280x720@[14.0:30.0]
Opening camera 1
I/opensslidentity.cc: (line 93): Returning key pair
I/opensslcertificate.cc: (line 58): Making certificate for WebRTC
I/opensslcertificate.cc: (line 105): Returning certificate
I/webrtcsdp.cc: (line 2952): Ignored line: a=sendrecv
(line 1742): AddSctpDataCodec: Got SCTP Port Number 5000
(line 2952): Ignored line: a=end-of-candidates
I/p2ptransportchannel.cc: (line 442): Set continual_gathering_policy to 1
(line 451): Set backup connection ping interval to 25000 milliseconds.
(line 460): Set ICE receiving timeout to 2500 milliseconds
(line 467): Set ping most likely connection to 0
(line 474): Set stable_writable_connection_ping_interval to 2500
(line 487): Set presume writable when fully relayed to 0
(line 496): Set regather_on_failed_networks_interval to 300000
(line 515): Set receiving_switching_delay to 1000
I/dtlssrtptransport.cc: (line 60): Setting RTCP Transport on audio transport 0x0
(line 65): Setting RTP Transport on audio transport 0x73bf381700
I/p2ptransportchannel.cc: (line 399): Received remote ICE parameters: ufrag=Kpy0, renomination disabled
I/peerconnection.cc: (line 2803): Adding audio transceiver for MID=audio at i=0 in response to the remote description.
I/webrtcvoiceengine.cc: (line 1501): Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
I/webrtcvoiceengine.cc: (line 324): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
(line 512): NetEq capacity is 50
(line 518): NetEq fast mode? 0
(line 525): Typing detection is enabled? 0
I/apm_helpers.cc: (line 163): VAD set to 0 for typing detection.
I/webrtcvoiceengine.cc: (line 536): Delay agnostic aec is enabled? 0
(line 546): Extended filter aec is enabled? 0
(line 556): Experimental ns is enabled? 0
(line 562): Intelligibility Enhancer is enabled? 0
I/audio_processing_impl.cc: (line 673): Highpass filter activated: 1
(line 687): Gain Controller 2 activated: 0
(line 689): Pre-amplifier activated: 0
I/webrtcvoiceengine.cc: (line 1519): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
I/channel.cc: (line 113): Created channel for audio
I/call.cc: (line 1047): UpdateAggregateNetworkState: aggregate_state=down
I/send_side_congestion_controller.cc: (line 308): SignalNetworkState Down
I/peerconnection.cc: (line 2803): Adding video transceiver for MID=video at i=1 in response to the remote description.
I/webrtcvideoengine.cc: (line 564): CreateChannel. Options: VideoOptions {}
I/channel.cc: (line 113): Created channel for video
I/call.cc: (line 1047): UpdateAggregateNetworkState: aggregate_state=down
I/send_side_congestion_controller.cc: (line 308): SignalNetworkState Down
I/peerconnection.cc: (line 3585): Session: 1242318263477639705 Old state: kStable New state: kHaveRemoteOffer
D/PCRTCClient: SignalingState: HAVE_REMOTE_OFFER
I/channel.cc: (line 837): Setting remote voice description
I/webrtcvoiceengine.cc: (line 1323): WebRtcVoiceMediaChannel::SetSendParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:0:1], AudioCodec[104:ISAC:32000:0:1], AudioCodec[9:G722:8000:0:1], AudioCodec[0:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:1], AudioCodec[113:telephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 4}], max_bandwidth_bps: -1, mid: audio, options: AudioOptions {}}
(line 1718): Recreate all the receive streams because the send codec has changed.
W/webrtcmediaengine.cc: (line 170): Unsupported RTP extension: {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}
I/webrtcvoiceengine.cc: (line 2114): WebRtcVoiceMediaChannel::SetMaxSendBitrate.
(line 1501): Setting voice channel options: AudioOptions {}
(line 324): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
(line 512): NetEq capacity is 50
(line 518): NetEq fast mode? 0
(line 525): Typing detection is enabled? 0
I/apm_helpers.cc: (line 163): VAD set to 0 for typing detection.
I/webrtcvoiceengine.cc: (line 536): Delay agnostic aec is enabled? 0
(line 546): Extended filter aec is enabled? 0
(line 556): Experimental ns is enabled? 0
(line 562): Intelligibility Enhancer is enabled? 0
I/audio_processing_impl.cc: (line 673): Highpass filter activated: 1
(line 687): Gain Controller 2 activated: 0
(line 689): Pre-amplifier activated: 0
I/webrtcvoiceengine.cc: (line 1519): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
I/webrtcvoiceengine.cc: (line 1863): AddRecvStream: {id:janusa0;ssrcs:[3245288059];ssrc_groups:;cname:janus;stream_ids:janus;}
I/neteq_impl.cc: (line 114): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=50, enable_fast_accelerate=false, enable_muted_state= true
I/audio_coding_module.cc: (line 463): Created
I/audio_receive_stream.cc: (line 106): AudioReceiveStream: 3245288059
(line 329): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3245288059, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), sync_group: janus}
I/call.cc: (line 1047): UpdateAggregateNetworkState: aggregate_state=down
I/send_side_congestion_controller.cc: (line 308): SignalNetworkState Down
I/PermissionManager: camera remind result:true
I/channel.cc: (line 655): Add remote ssrc: 3245288059
I/CameraManager: open camera: 1, package name: org.appspot.apprtc
I/HwSystemManager: pkgName = org.appspot.apprtc pid = 17709uidOf3RdApk = 10153permType = 0permCfg=1
I/channel.cc: (line 778): Changing voice state, recv=0 send=0
I/BackgroundPermManager: pkgName: org.appspot.apprtc,pid: 17709 ,uidOf3RdApk: 10153 ,permType: 0 ,permCfg: 1
I/channel.cc: (line 972): Setting remote video description
I/webrtcvideoengine.cc: [1/2] (line 754): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9], VideoCodec[102:H264], VideoCodec[127:H264], VideoCodec[125:H264], VideoCodec[108:H264], VideoCodec[124:H264], VideoCodec[123:H264], VideoCodec[97:rtx], VideoCodec[99:rtx], VideoCodec[101:rtx], VideoCodec[103:rtx], VideoCodec[104:rtx], VideoCodec[126:rtx], VideoCodec[109:rtx], VideoCodec[105:rtx], VideoCodec[106:rtx]], conference_mode: no, extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: urn:3gpp:video-orientation, id: 13}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 12}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: http://tools.ietf
[2/2] .org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 4}], max_bandwidth_bps: -1, mid: video}
W/webrtcmediaengine.cc: (line 170): Unsupported RTP extension: {uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}
(line 170): Unsupported RTP extension: {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}
I/webrtcvideoengine.cc: (line 763): Using codec: VideoCodec[96:VP8]
(line 812): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed.
I/webrtcvideoengine.cc: (line 1205): AddRecvStream: {id:janusv0;ssrcs:[363841697,2079154895];ssrc_groups:{semantics:FID;ssrcs:[363841697,2079154895]};cname:janus;stream_ids:janus;}
I/paced_sender.cc: (line 104): PacedSender paused.
I/video_receive_stream.cc: (line 113): VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 96, payload_name: VP8, codec_params: {}}, {decoder: (VideoDecoder), payload_type: 98, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 363841697, local_ssrc: 1, rtcp_mode: RtcpMode::kCompound, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 102, red_type: 100, rtx_ssrc: 2079154895, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: janus, target_delay_ms: 0}
I/call.cc: (line 1047): UpdateAggregateNetworkState: aggregate_state=down
I/send_side_congestion_controller.cc: (line 308): SignalNetworkState Down
I/channel.cc: (line 655): Add remote ssrc: 363841697
I/channel.cc: (line 908): Changing video state, send=0
I/peerconnection.cc: (line 4830): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport.
(line 2435): Processing the addition of a new track for MID=audio (added to streams=[janus]).
I/webrtcvoiceengine.cc: (line 1972): SetOutputVolume() to 1 for recv stream with ssrc 3245288059
I/peerconnection.cc: (line 2435): Processing the addition of a new track for MID=video (added to streams=[janus]).
I/webrtcvideoengine.cc: (line 1335): SetSink: ssrc:363841697 (ptr)
I/CameraManagerGlobal: do not need hide aux camera, device number: 2
D/PCRTCClient: ==onAddTrack mediaStreams size:1
E/rtpsender.cc: (line 468): Tried to get DTMF sender from video sender.
D/PCRTCClient: ==onAddTrack mediaStreams size:1
W/peerconnection.cc: (line 1911): CreateAnswer: offer_to_receive_audio is not supported with Unified Plan semantics. Use the RtpTransceiver API instead.
W/peerconnection.cc: (line 1916): CreateAnswer: offer_to_receive_video is not supported with Unified Plan semantics. Use the RtpTransceiver API instead.
I/peerconnection.cc: (line 4852): Local and Remote descriptions must be applied to get the SSL Role of the session.
I/chatty: uid=10153(u0_a153) signaling_threa identical 1 line
I/peerconnection.cc: (line 4852): Local and Remote descriptions must be applied to get the SSL Role of the session.
D/PCRTCClient: ==onAddStream tracks size:1
D/PCRTCClient: Change media description from: m=video 9 UDP/TLS/RTP/SAVPF 96 98 97 99 to m=video 9 UDP/TLS/RTP/SAVPF 96 98 97 99
D/PCRTCClient: Remote SDP set succesfully
D/PCRTCClient: Set local SDP from ANSWER
I/webrtcsdp.cc: (line 1742): AddSctpDataCodec: Got SCTP Port Number 5000
I/dtlssrtptransport.cc: (line 60): Setting RTCP Transport on audio transport 0x0
(line 65): Setting RTP Transport on audio transport 0x73bf381700
I/p2ptransportchannel.cc: (line 388): Set ICE ufrag: P3eV pwd: u8YRL96zzZPXxAMVH3KA0R0q on transport audio
I/dtlstransport.cc: (line 376): DtlsTransport[audio|1|__]: DTLS setup complete.
I/rtptransceiver.cc: (line 174): Changing transceiver (MID=audio) current direction from <not set> to kRecvOnly.
(line 174): Changing transceiver (MID=video) current direction from <not set> to kRecvOnly.
I/channel.cc: (line 529): Channel enabled
(line 778): Changing voice state, recv=0 send=0
(line 529): Channel enabled
I/channel.cc: (line 908): Changing video state, send=0
I/peerconnection.cc: (line 3585): Session: 1242318263477639705 Old state: kHaveRemoteOffer New state: kStable
D/PCRTCClient: SignalingState: STABLE
I/channel.cc: (line 786): Setting local voice description
I/webrtcvoiceengine.cc: (line 1360): WebRtcVoiceMediaChannel::SetRecvParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:0:1], AudioCodec[104:ISAC:32000:0:1], AudioCodec[9:G722:8000:0:1], AudioCodec[0:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:1], AudioCodec[113:telephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 4}]}
(line 1529): Setting receive voice codecs.
I/audio_receive_stream.cc: (line 329): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3245288059, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), sync_group: janus}
I/call.cc: (line 1047): UpdateAggregateNetworkState: aggregate_state=down
I/send_side_congestion_controller.cc: (line 308): SignalNetworkState Down
I/audio_receive_stream.cc: (line 130): ~AudioReceiveStream: 3245288059
I/neteq_impl.cc: (line 114): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=50, enable_fast_accelerate=false, enable_muted_state= true
I/audio_coding_module.cc: (line 463): Created
I/audio_receive_stream.cc: (line 106): AudioReceiveStream: 3245288059
(line 329): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3245288059, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: [{uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 4}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), sync_group: janus}
I/call.cc: (line 1047): UpdateAggregateNetworkState: aggregate_state=down
I/send_side_congestion_controller.cc: (line 308): SignalNetworkState Down
I/audio_device_module.cc: (line 272): Playing
(line 193): InitPlayout
(line 207): PlayoutIsInitialized
I/audio_track_jni.cc: (line 77): InitPlayout
I/WebRtcAudioTrackExternal: initPlayout(sampleRate=48000, channels=1)
byteBuffer.capacity: 960
I/audio_track_jni.cc: (line 189): OnCacheDirectBufferAddress
(line 194): direct buffer capacity: 960
(line 198): frames_per_buffer: 480
I/WebRtcAudioTrackExternal: AudioTrack.getMinBufferSize: 3848
createAudioTrackOnLollipopOrHigher
I/WebRtcAudioTrackExternal: nativeOutputSampleRate: 48000
I/Camera2Session: Camera opened.
D/AudioTrack: OFFLOAD 0,mNotificationFrames 0,mStreamType =-1,mOriginalSampleRate 48000,mAfSampleRate 48000,mTransfer 3
Client defaulted notificationFrames to 962 for frameCount 1924
I/Camera2Session: Camera capture session configured.
I/Camera2Session: Using video stabilization.
I/Camera2Session: Auto-focus is not available.
I/WebRtcAudioTrackExternal: AudioTrack: session ID: 913, channels: 1, sample rate: 48000, max gain: 1.0
I/WebRtcAudioTrackExternal: AudioTrack: buffer size in frames: 1924
I/WebRtcAudioTrackExternal: AudioTrack: buffer capacity in frames: 1924
I/audio_device_module.cc: (line 200): output: 0
(line 238): StartPlayout
(line 272): Playing
I/audio_track_jni.cc: (line 99): StartPlayout
I/VolumeLogger: start@[name=worker_thread - 17795, id=1821]
I/VolumeLogger: audio mode is: MODE_IN_COMMUNICATION
W/GrallocMapperPassthrough: buffer descriptor with invalid usage bits 0x2080000
I/WebRtcAudioTrackExternal: startPlayout
D/android.media.AudioTrack: [HSM] AudioTrace play() uid: 10153, pid: 17709
I/Camera2Session: Camera device successfully started.
I/CameraCapturer: Create session done. Switch state: IDLE
I/nativecapturerobserver.cc: (line 58): NativeCapturerObserver_nativeCapturerStarted
I/SurfaceTextureHelper: Setting listener to org.webrtc.-$$Lambda$Camera2Session$CaptureSessionCallback$UDvzHNj8-cAJE1WNByx98pxD9vA@da50f08
I/audio_device_module.cc: (line 245): output: 0
I/audio_device_buffer.cc: (line 100): StartPlayout
I/channel.cc: (line 778): Changing voice state, recv=1 send=0
I/channel.cc: (line 921): Setting local video description
I/WebRtcAudioTrackExternal: AudioTrackThread@[name=AudioTrackJavaThread, id=1831]
I/webrtcvideoengine.cc: (line 990): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[97:rtx], VideoCodec[99:rtx]], extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: urn:3gpp:video-orientation, id: 13}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 12}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 4}]}
I/audio_device_buffer.cc: (line 314): Size of playout buffer: 480
I/webrtcvideoengine.cc: (line 1005): Changing recv codecs from {VideoCodec[96:VP8], VideoCodec[98:VP9]} to {VideoCodec[96:VP8], VideoCodec[98:VP9]}
I/webrtcvideoengine.cc: (line 2424): MaybeRecreateWebRtcFlexfecStream (recv) because of SetRecvParameters
(line 2429): RecreateWebRtcVideoStream (recv) because of SetRecvParameters
I/call.cc: (line 1047): UpdateAggregateNetworkState: aggregate_state=down
I/send_side_congestion_controller.cc: (line 308): SignalNetworkState Down
I/video_receive_stream.cc: (line 153): ~VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 96, payload_name: VP8, codec_params: {}}, {decoder: (VideoDecoder), payload_type: 98, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 363841697, local_ssrc: 1, rtcp_mode: RtcpMode::kCompound, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 102, red_type: 100, rtx_ssrc: 2079154895, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: janus, target_delay_ms: 0}
I/VolumeLogger: VOICE_CALL stream volume: 1 (max=5)
I/receive_statistics_proxy.cc: (line 487): Frames decoded 0
WebRTC.Video.DroppedFrames.Receiver 0
I/video_receive_stream.cc: [1/2] (line 113): VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 96, payload_name: VP8, codec_params: {}}, {decoder: (VideoDecoder), payload_type: 98, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 363841697, local_ssrc: 1, rtcp_mode: RtcpMode::kCompound, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: -1, red_type: -1, rtx_ssrc: 2079154895, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), }, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 12}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 13},
[2/2] {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 4}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: janus, target_delay_ms: 0}
I/call.cc: (line 1047): UpdateAggregateNetworkState: aggregate_state=down
I/send_side_congestion_controller.cc: (line 308): SignalNetworkState Down
I/channel.cc: (line 908): Changing video state, send=0
I/peerconnection.cc: (line 3537): Changing IceConnectionState 0 => 1
W/jseptransportcontroller.cc: (line 303): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
W/jseptransportcontroller.cc: (line 303): Not adding candidate because the JsepTransport doesn't exist. Ignore it.
I/NetworkMonitor: Start monitoring with native observer 497126715584
W/GrallocMapperPassthrough: buffer descriptor with invalid usage bits 0x2080000
I/HwCust: Constructor found for class android.net.HwCustConnectivityManagerImpl
D/HwCust: Create obj success use class android.net.HwCustConnectivityManagerImpl
W/GrallocMapperPassthrough: buffer descriptor with invalid usage bits 0x2080000
W/GrallocMapperPassthrough: buffer descriptor with invalid usage bits 0x2080000
I/androidnetworkmonitor.cc: (line 362): Android network monitor found 1 networks
(line 327): Network connected: NetInfo[name rmnet0; handle 700096105182; type 3; address 10.154.222.237 2409:893c:46:ebe3:68e6:7260:eae2:7061]
I/basicportallocator.cc: (line 342): Start getting ports with prune_turn_ports disabled
W/androidnetworkmonitor.cc: (line 376): Get an unknown type for the interface rmnet_ims
I/basicportallocator.cc: (line 851): Network manager has started
I/network.cc: (line 476): Network change was observed
D/PCRTCClient: IceConnectionState: CHECKING
W/androidnetworkmonitor.cc: (line 376): Get an unknown type for the interface rmnet_ims
D/PCRTCClient: IceGatheringState: GATHERING
I/basicportallocator.cc: (line 753): Allocate ports on 7 networks
W/physicalsocketserver.cc: (line 209): Binding socket to network address 2409:8807:828b:123a:1:1:cb68:45ac failed; result: -3
E/basicpacketsocketfactory.cc: (line 52): UDP bind failed with error 0
D/PCRTCClient: Local SDP set succesfully
D/PCRTCClient: Add 0 remote candidates
D/CallRTCClient: Sending ANSWER, delay=21124ms
W/GrallocMapperPassthrough: buffer descriptor with invalid usage bits 0x2080000
I/basicportallocator.cc: (line 1446): Net[rmnet_ims:2409:8807:828b:x:x:x:x:x/64:Unknown:id=5]: Allocation Phase=Udp
I/port.cc: (line 319): Port[0x73bf1b8300::1:0:local:Net[rmnet_ims:2409:8807:828b:x:x:x:x:x/64:Unknown:id=5]]: Port created with network cost 50
W/physicalsocketserver.cc: (line 209): Binding socket to network address 2409:8807:828b:123a:1:1:cb68:45ac failed; result: -3
E/basicpacketsocketfactory.cc: (line 52): UDP bind failed with error 0
W/stunport.cc: (line 208): Port[0x73bf1b8300::1:0:local:Net[rmnet_ims:2409:8807:828b:x:x:x:x:x/64:Unknown:id=5]]: UDP socket creation failed
I/basicportallocator.cc: (line 1446): Net[rmnet0:2409:893c:46:x:x:x:x:x/64:Cellular:id=4]: Allocation Phase=Udp
I/port.cc: (line 319): Port[0x73bf1b8300::1:0:local:Net[rmnet0:2409:893c:46:x:x:x:x:x/64:Cellular:id=4]]: Port created with network cost 900
I/basicportallocator.cc: (line 1517): AllocationSequence: UDPPort will be handling the STUN candidate generation.
(line 873): Adding allocated port for audio
(line 892): Port[0x73bf1b8300:audio:1:0:local:Net[rmnet0:2409:893c:46:x:x:x:x:x/64:Cellular:id=4]]: Added port to allocator
I/basicportallocator.cc: (line 909): Port[0x73bf1b8300:audio:1:0:local:Net[rmnet0:2409:893c:46:x:x:x:x:x/64:Cellular:id=4]]: Gathered candidate: Cand[:2845846792:1:udp:2122197247:[2409:893c:46:x:x:x:x:x]:34470:local::0:P3eV:u8YRL96zzZPXxAMVH3KA0R0q:4:900:0]
(line 956): Port[0x73bf1b8300:audio:1:0:local:Net[rmnet0:2409:893c:46:x:x:x:x:x/64:Cellular:id=4]]: Port ready.
W/physicalsocketserver.cc: (line 562): Socket::OPT_DSCP not supported.
I/p2ptransportchannel.cc: (line 734): Port[0x73bf1b8300:audio:1:0:local:Net[rmnet0:2409:893c:46:x:x:x:x:x/64:Cellular:id=4]]: SetOption(5, 0) failed: 0
I/stunport.cc: (line 389): Port[0x73bf1b8300:audio:1:0:local:Net[rmnet0:2409:893c:46:x:x:x:x:x/64:Cellular:id=4]]: Starting STUN host lookup for stun.freeswitch.org:3478
D/HwRTBlurUtils: check blur style for HwToast-Toast, themeResId : 0x7f0e00a6, context : org.appspot.apprtc.EchoTestActivity3@f8162e1, Nhwext : 6, get Blur : disable with , android.graphics.drawable.NinePatchDrawable@2c16a26
D/CallRTCClient: ========onIceCandidate=======
I/basicportallocator.cc: (line 1446): Net[rmnet0:10.154.222.x/32:Cellular:id=3]: Allocation Phase=Udp
I/port.cc: (line 319): Port[0x73bf1b8800::1:0:local:Net[rmnet0:10.154.222.x/32:Cellular:id=3]]: Port created with network cost 900
I/basicportallocator.cc: (line 1517): AllocationSequence: UDPPort will be handling the STUN candidate generation.
(line 873): Adding allocated port for audio
I/basicportallocator.cc: (line 892): Port[0x73bf1b8800:audio:1:0:local:Net[rmnet0:10.154.222.x/32:Cellular:id=3]]: Added port to allocator
I/basicportallocator.cc: (line 909): Port[0x73bf1b8800:audio:1:0:local:Net[rmnet0:10.154.222.x/32:Cellular:id=3]]: Gathered candidate: Cand[:3618465998:1:udp:2122129151:10.154.222.x:43295:local::0:P3eV:u8YRL96zzZPXxAMVH3KA0R0q:3:900:0]
I/basicportallocator.cc: (line 956): Port[0x73bf1b8800:audio:1:0:local:Net[rmnet0:10.154.222.x/32:Cellular:id=3]]: Port ready.
W/physicalsocketserver.cc: (line 562): Socket::OPT_DSCP not supported.
I/p2ptransportchannel.cc: (line 734): Port[0x73bf1b8800:audio:1:0:local:Net[rmnet0:10.154.222.x/32:Cellular:id=3]]: SetOption(5, 0) failed: 0
I/port.cc: (line 1072): Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--W|-|0|0|8655919583267012094|-]: Connection created
I/p2ptransportchannel.cc: (line 1097): Channel[audio|1|__]: Created connection with origin: 2, total: 1
I/p2ptransportchannel.cc: (line 1730): Channel[audio|1|__]: Transport channel state changed from 0 to 2
I/jseptransportcontroller.cc: (line 1166): audio Transport 1 state changed. Check if state is complete.
I/p2ptransportchannel.cc: (line 1311): Channel[audio|1|__]: Have a pingable connection for the first time; starting to ping.
I/stunport.cc: (line 389): Port[0x73bf1b8800:audio:1:0:local:Net[rmnet0:10.154.222.x/32:Cellular:id=3]]: Starting STUN host lookup for stun.freeswitch.org:3478
D/CallRTCClient: ========onIceCandidate=======
W/GrallocMapperPassthrough: buffer descriptor with invalid usage bits 0x2080000
I/basicportallocator.cc: (line 1446): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=2]: Allocation Phase=Udp
I/port.cc: (line 319): Port[0x73bf1b8d00::1:0:local:Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=2]]: Port created with network cost 0
I/basicportallocator.cc: (line 1517): AllocationSequence: UDPPort will be handling the STUN candidate generation.
(line 873): Adding allocated port for audio
(line 892): Port[0x73bf1b8d00:audio:1:0:local:Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=2]]: Added port to allocator
(line 909): Port[0x73bf1b8d00:audio:1:0:local:Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=2]]: Gathered candidate: Cand[:559267639:1:udp:2122071295:[0:0:0:x:x:x:x:x]:33555:local::0:P3eV:u8YRL96zzZPXxAMVH3KA0R0q:2:0:0]
(line 956): Port[0x73bf1b8d00:audio:1:0:local:Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=2]]: Port ready.
W/physicalsocketserver.cc: (line 562): Socket::OPT_DSCP not supported.
I/p2ptransportchannel.cc: (line 734): Port[0x73bf1b8d00:audio:1:0:local:Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=2]]: SetOption(5, 0) failed: 0
I/stunport.cc: (line 389): Port[0x73bf1b8d00:audio:1:0:local:Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=2]]: Starting STUN host lookup for stun.freeswitch.org:3478
I/AppRTCUtils: randomPoz:34 charSetLen:62
randomPoz:14 charSetLen:62
randomPoz:27 charSetLen:62
D/CallRTCClient: ========onIceCandidate=======
I/AppRTCUtils: randomPoz:24 charSetLen:62
randomPoz:3 charSetLen:62
randomPoz:8 charSetLen:62
randomPoz:0 charSetLen:62
randomPoz:33 charSetLen:62
I/AppRTCUtils: randomPoz:32 charSetLen:62
W/libEGL: EGLNativeWindowType 0x73a3ee1010 disconnect failed
I/AppRTCUtils: randomPoz:21 charSetLen:62
randomPoz:57 charSetLen:62
I/AppRTCUtils: randomPoz:45 charSetLen:62
I/basicportallocator.cc: (line 1446): Net[lo:127.0.0.x/8:Loopback:id=1]: Allocation Phase=Udp
I/port.cc: (line 319): Port[0x73bf1b9700::1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: Port created with network cost 0
I/basicportallocator.cc: (line 1517): AllocationSequence: UDPPort will be handling the STUN candidate generation.
(line 873): Adding allocated port for audio
(line 892): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: Added port to allocator
(line 909): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: Gathered candidate: Cand[:1510613869:1:udp:2121998079:127.0.0.x:49388:local::0:P3eV:u8YRL96zzZPXxAMVH3KA0R0q:1:0:0]
(line 956): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: Port ready.
W/physicalsocketserver.cc: (line 562): Socket::OPT_DSCP not supported.
I/p2ptransportchannel.cc: (line 734): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: SetOption(5, 0) failed: 0
I/port.cc: (line 1072): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--W|-|0|0|8655919583266749950|-]: Connection created
I/p2ptransportchannel.cc: (line 1097): Channel[audio|1|__]: Created connection with origin: 2, total: 2
I/stunport.cc: (line 389): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: Starting STUN host lookup for stun.freeswitch.org:3478
I/basicportallocator.cc: (line 1446): Net[any:0.0.0.x/0:Wildcard:id=0]: Allocation Phase=Udp
I/port.cc: (line 319): Port[0x73bf1b9c00::1:0:local:Net[any:0.0.0.x/0:Wildcard:id=0]]: Port created with network cost 999
D/CallRTCClient: ========onIceCandidate=======
I/basicportallocator.cc: (line 1517): AllocationSequence: UDPPort will be handling the STUN candidate generation.
(line 873): Adding allocated port for audio
(line 892): Port[0x73bf1b9c00:audio:1:0:local:Net[any:0.0.0.x/0:Wildcard:id=0]]: Added port to allocator
I/basicportallocator.cc: (line 909): Port[0x73bf1b9c00:audio:1:0:local:Net[any:0.0.0.x/0:Wildcard:id=0]]: Gathered candidate: Cand[:3618465998:1:udp:2113937151:10.154.222.x:37222:local::0:P3eV:u8YRL96zzZPXxAMVH3KA0R0q:0:999:0]
(line 976): Candidate not signaled yet because it is from the any address port: Cand[:3618465998:1:udp:2113937151:10.154.222.x:37222:local::0:P3eV:u8YRL96zzZPXxAMVH3KA0R0q:0:999:0]
I/stunport.cc: (line 389): Port[0x73bf1b9c00:audio:1:0:local:Net[any:0.0.0.x/0:Wildcard:id=0]]: Starting STUN host lookup for stun.freeswitch.org:3478
I/basicportallocator.cc: (line 1446): Net[any:0:0:0:x:x:x:x:x/0:Wildcard:id=0]: Allocation Phase=Udp
I/port.cc: (line 319): Port[0x73bf1ba100::1:0:local:Net[any:0:0:0:x:x:x:x:x/0:Wildcard:id=0]]: Port created with network cost 999
I/basicportallocator.cc: (line 1517): AllocationSequence: UDPPort will be handling the STUN candidate generation.
(line 873): Adding allocated port for audio
(line 892): Port[0x73bf1ba100:audio:1:0:local:Net[any:0:0:0:x:x:x:x:x/0:Wildcard:id=0]]: Added port to allocator
I/basicportallocator.cc: (line 909): Port[0x73bf1ba100:audio:1:0:local:Net[any:0:0:0:x:x:x:x:x/0:Wildcard:id=0]]: Gathered candidate: Cand[:2845846792:1:udp:2113939711:[2409:893c:46:x:x:x:x:x]:38795:local::0:P3eV:u8YRL96zzZPXxAMVH3KA0R0q:0:999:0]
I/basicportallocator.cc: (line 976): Candidate not signaled yet because it is from the any address port: Cand[:2845846792:1:udp:2113939711:[2409:893c:46:x:x:x:x:x]:38795:local::0:P3eV:u8YRL96zzZPXxAMVH3KA0R0q:0:999:0]
I/stunport.cc: (line 389): Port[0x73bf1ba100:audio:1:0:local:Net[any:0:0:0:x:x:x:x:x/0:Wildcard:id=0]]: Starting STUN host lookup for stun.freeswitch.org:3478
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--W|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=557271497a4f754a44576a4e
D/WebSocketChannelClient: C->WSS: {"janus":"message","body":{"request":"accept"},"session_id":6886760364035320,"handle_id":5790718854272780,"transaction":"nulliObYDIAhgV5t","jsep":{"type":"answer","sdp":"v=0\r\no=- 1242318263477639705 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video data\r\na=msid-semantic: WMS\r\nm=audio 9 UDP\/TLS\/RTP\/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:P3eV\r\na=ice-pwd:u8YRL96zzZPXxAMVH3KA0R0q\r\na=ice-options:trickle renomination\r\na=fingerprint:sha-256 8A:D8:4D:BB:E9:89:47:6F:8A:31:CB:A3:78:B0:10:A7:57:D7:50:BB:5A:75:9D:70:DD:BE:2E:D6:61:A5:7E:12\r\na=setup:active\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=recvonly\r\na=rtcp-mux\r\na=rtpmap:111 opus\/48000\/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC\/16000\r\na=rtpmap:104 ISAC\/32000\r\na=rtpmap:9 G722\/8000\r\na=rtpmap:0 PCMU\/8000\r\na=rtpmap:8 PCMA\/8000\r\na=rtpmap:106 CN\/32000\r\na=rtpmap:105 CN\/16000\r\na=rtpmap:13 CN\/8000\r\na=rtpmap:110 telephone-event\/48000\r\na=rtpmap:112 telephone-event\/32000\r\na=rtpmap:113 telephone-event\/16000\r\na=rtpmap:126 telephone-event\/8000\r\nm=video 9 UDP\/TLS\/RTP\/SAVPF 96 98 97 99\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:P3eV\r\na=ice-pwd:u8YRL96zzZPXxAMVH3KA0R0q\r\na=ice-options:trickle renomination\r\na=fingerprint:sha-256 8A:D8:4D:BB:E9:89:47:6F:8A:31:CB:A3:78:B0:10:A7:57:D7:50:BB:5A:75:9D:70:DD:BE:2E:D6:61:A5:7E:12\r\na=setup:active\r\na=mid:video\r\na=extmap:14 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:2 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/abs-send-time\r\na=extmap:13 urn:3gpp:video-orientation\r\na=extmap:3 http:\/\/www.ietf.org\/id\/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:12 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/playout-delay\r\na=extmap:11 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/video-content-type\r\na=extmap:7 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/video-timing\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=recvonly\r\na=rtcp-mux\r\na=rtpmap:96 VP8\/90000\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtpmap:98 VP9\/90000\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=rtpmap:97 rtx\/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:99 rtx\/90000\r\na=fmtp:99 apt=98\r\nm=application 9 UDP\/DTLS\/SCTP webrtc-datachannel\r\nc=IN IP4 0.0.0.0\r\nb=AS:30\r\na=ice-ufrag:P3eV\r\na=ice-pwd:u8YRL96zzZPXxAMVH3KA0R0q\r\na=ice-options:trickle renomination\r\na=fingerprint:sha-256 8A:D8:4D:BB:E9:89:47:6F:8A:31:CB:A3:78:B0:10:A7:57:D7:50:BB:5A:75:9D:70:DD:BE:2E:D6:61:A5:7E:12\r\na=setup:active\r\na=mid:data\r\na=sctp-port:5000\r\n"}}
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--W|-|0|0|8655919583266749950|-]: Sent STUN ping, id=557271497a4f754a44576a4e, use_candidate=0, nomination=0
W/GrallocMapperPassthrough: buffer descriptor with invalid usage bits 0x2080000
I/zygote64: Do partial code cache collection, code=125KB, data=101KB
After code cache collection, code=125KB, data=101KB
Increasing code cache capacity to 512KB
I/AppRTCUtils: randomPoz:56 charSetLen:62
randomPoz:3 charSetLen:62
randomPoz:9 charSetLen:62
randomPoz:41 charSetLen:62
randomPoz:60 charSetLen:62
randomPoz:54 charSetLen:62
randomPoz:3 charSetLen:62
randomPoz:33 charSetLen:62
randomPoz:29 charSetLen:62
randomPoz:60 charSetLen:62
randomPoz:54 charSetLen:62
W/VRSystemServiceManager: vr service is not alive
I/AppRTCUtils: randomPoz:18 charSetLen:62
D/WSRTCClient: trickleCandidate transactionID: null4DJp82Dhd82S
D/WSRTCClient: C->WSS: {"janus":"trickle","candidate":{"candidate":"candidate:2845846792 1 udp 2122197247 2409:893c:46:ebe3:68e6:7260:eae2:7061 34470 typ host generation 0 ufrag P3eV network-id 4 network-cost 900","sdpMid":"audio","sdpMLineIndex":0},"transaction":"null4DJp82Dhd82S","session_id":6886760364035320,"handle_id":5790718854272780}
D/WebSocketChannelClient: C->WSS: {"janus":"trickle","candidate":{"candidate":"candidate:2845846792 1 udp 2122197247 2409:893c:46:ebe3:68e6:7260:eae2:7061 34470 typ host generation 0 ufrag P3eV network-id 4 network-cost 900","sdpMid":"audio","sdpMLineIndex":0},"transaction":"null4DJp82Dhd82S","session_id":6886760364035320,"handle_id":5790718854272780}
I/AppRTCUtils: randomPoz:44 charSetLen:62
randomPoz:46 charSetLen:62
randomPoz:2 charSetLen:62
randomPoz:6 charSetLen:62
randomPoz:30 charSetLen:62
randomPoz:20 charSetLen:62
randomPoz:6 charSetLen:62
randomPoz:3 charSetLen:62
randomPoz:3 charSetLen:62
randomPoz:1 charSetLen:62
randomPoz:0 charSetLen:62
randomPoz:42 charSetLen:62
D/WSRTCClient: trickleCandidate transactionID: nullsuCGeUGDDBAq
D/WSRTCClient: C->WSS: {"janus":"trickle","candidate":{"candidate":"candidate:3618465998 1 udp 2122129151 10.154.222.237 43295 typ host generation 0 ufrag P3eV network-id 3 network-cost 900","sdpMid":"audio","sdpMLineIndex":0},"transaction":"nullsuCGeUGDDBAq","session_id":6886760364035320,"handle_id":5790718854272780}
D/WebSocketChannelClient: C->WSS: {"janus":"trickle","candidate":{"candidate":"candidate:3618465998 1 udp 2122129151 10.154.222.237 43295 typ host generation 0 ufrag P3eV network-id 3 network-cost 900","sdpMid":"audio","sdpMLineIndex":0},"transaction":"nullsuCGeUGDDBAq","session_id":6886760364035320,"handle_id":5790718854272780}
I/AppRTCUtils: randomPoz:12 charSetLen:62
randomPoz:26 charSetLen:62
randomPoz:55 charSetLen:62
randomPoz:57 charSetLen:62
randomPoz:28 charSetLen:62
D/OpenGLRenderer: HWUI Binary is enabled
I/AppRTCUtils: randomPoz:7 charSetLen:62
randomPoz:0 charSetLen:62
randomPoz:20 charSetLen:62
randomPoz:16 charSetLen:62
I/AppRTCUtils: randomPoz:18 charSetLen:62
randomPoz:38 charSetLen:62
randomPoz:9 charSetLen:62
D/WSRTCClient: trickleCandidate transactionID: nullMa35cHAUQSmJ
C->WSS: {"janus":"trickle","candidate":{"candidate":"candidate:559267639 1 udp 2122071295 ::1 33555 typ host generation 0 ufrag P3eV network-id 2","sdpMid":"audio","sdpMLineIndex":0},"transaction":"nullMa35cHAUQSmJ","session_id":6886760364035320,"handle_id":5790718854272780}
D/WebSocketChannelClient: C->WSS: {"janus":"trickle","candidate":{"candidate":"candidate:559267639 1 udp 2122071295 ::1 33555 typ host generation 0 ufrag P3eV network-id 2","sdpMid":"audio","sdpMLineIndex":0},"transaction":"nullMa35cHAUQSmJ","session_id":6886760364035320,"handle_id":5790718854272780}
I/AppRTCUtils: randomPoz:59 charSetLen:62
randomPoz:50 charSetLen:62
randomPoz:7 charSetLen:62
randomPoz:21 charSetLen:62
randomPoz:56 charSetLen:62
randomPoz:31 charSetLen:62
randomPoz:53 charSetLen:62
I/AppRTCUtils: randomPoz:30 charSetLen:62
randomPoz:21 charSetLen:62
randomPoz:3 charSetLen:62
randomPoz:38 charSetLen:62
randomPoz:35 charSetLen:62
D/WSRTCClient: trickleCandidate transactionID: null7yHV4f1eVDmj
D/WSRTCClient: C->WSS: {"janus":"trickle","candidate":{"candidate":"candidate:1510613869 1 udp 2121998079 127.0.0.1 49388 typ host generation 0 ufrag P3eV network-id 1","sdpMid":"audio","sdpMLineIndex":0},"transaction":"null7yHV4f1eVDmj","session_id":6886760364035320,"handle_id":5790718854272780}
D/WebSocketChannelClient: C->WSS: {"janus":"trickle","candidate":{"candidate":"candidate:1510613869 1 udp 2121998079 127.0.0.1 49388 typ host generation 0 ufrag P3eV network-id 1","sdpMid":"audio","sdpMLineIndex":0},"transaction":"null7yHV4f1eVDmj","session_id":6886760364035320,"handle_id":5790718854272780}
I/NetworkMonitorAutoDetect: Network becomes available: 163
D/OpenGLRenderer: HWUI Binary is enabled
I/androidnetworkmonitor.cc: (line 327): Network connected: NetInfo[name rmnet0; handle 700096105182; type 3; address 10.154.222.237 2409:893c:46:ebe3:68e6:7260:eae2:7061]
I/network.cc: (line 476): Network change was observed
W/androidnetworkmonitor.cc: (line 376): Get an unknown type for the interface rmnet_ims
I/NetworkMonitorAutoDetect: capabilities changed: [ Transports: CELLULAR Capabilities: SUPL&INTERNET&NOT_RESTRICTED&TRUSTED&NOT_VPN&VALIDATED&FOREGROUND LinkUpBandwidth>=51200Kbps LinkDnBandwidth>=102400Kbps Specifier: <0>]
I/androidnetworkmonitor.cc: (line 327): Network connected: NetInfo[name rmnet0; handle 700096105182; type 3; address 10.154.222.237 2409:893c:46:ebe3:68e6:7260:eae2:7061]
I/network.cc: (line 476): Network change was observed
W/androidnetworkmonitor.cc: (line 376): Get an unknown type for the interface rmnet_ims
I/NetworkMonitorAutoDetect: link properties changed: {InterfaceName: rmnet0 LinkAddresses: [*] Routes: [*] DnsAddresses: [211.137.191.27,218.201.96.131,2409:803c:2000:2::27,2409:803c:2000:4::131,] Domains: null MTU: 1400 TcpBufferSizes: 524288,4194304,8388608,262144,524288,1048576}
W/ViewRootImpl[Toast]: EGLdebug relayoutWindow Surface isSurface(name=null) (appName =org.appspot.apprtc) ( mSurfaceControllerIsValid =false) (mNativeObject =0)/@0xa2b360a,viewVisibility is0
I/androidnetworkmonitor.cc: (line 327): Network connected: NetInfo[name rmnet0; handle 700096105182; type 3; address 10.154.222.237 2409:893c:46:ebe3:68e6:7260:eae2:7061]
I/network.cc: (line 476): Network change was observed
W/androidnetworkmonitor.cc: (line 376): Get an unknown type for the interface rmnet_ims
W/ViewRootImpl[Toast]: EGLdebug relayoutWindow Surface isSurface(name=null) (appName =org.appspot.apprtc) ( mSurfaceControllerIsValid =true) (mNativeObject =73a3ee1000)/@0xa2b360a,relayoutResult is7
EGLdebug Surface isSurface(name=null) (appName =org.appspot.apprtc) ( mSurfaceControllerIsValid =true) (mNativeObject =73a3ee1000)/@0xa2b360a
D/mali_winsys: EGLint new_window_surface(egl_winsys_display *, void *, EGLSurface, EGLConfig, egl_winsys_surface **, egl_color_buffer_format *, EGLBoolean) returns 0x3000
I/basicportallocator.cc: (line 1446): Net[rmnet_ims:2409:8807:828b:x:x:x:x:x/64:Unknown:id=5]: Allocation Phase=Relay
I/basicportallocator.cc: (line 1446): Net[rmnet0:2409:893c:46:x:x:x:x:x/64:Cellular:id=4]: Allocation Phase=Relay
I/basicportallocator.cc: (line 1446): Net[rmnet0:10.154.222.x/32:Cellular:id=3]: Allocation Phase=Relay
I/basicportallocator.cc: (line 1446): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=2]: Allocation Phase=Relay
I/port.cc: (line 1697): Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--W|-|0|0|8655919583267012094|-]: Sent STUN ping, id=7338594871705371424a7051, use_candidate=0, nomination=0
I/basicportallocator.cc: (line 1446): Net[lo:127.0.0.x/8:Loopback:id=1]: Allocation Phase=Relay
(line 1446): Net[any:0.0.0.x/0:Wildcard:id=0]: Allocation Phase=Relay
I/basicportallocator.cc: (line 1446): Net[any:0:0:0:x:x:x:x:x/0:Wildcard:id=0]: Allocation Phase=Relay
W/stunport.cc: (line 400): Port[0x73bf1ba100:audio:1:0:local:Net[any:0:0:0:x:x:x:x:x/0:Wildcard:id=0]]: StunPort: stun host lookup received error 0
I/basicportallocator.cc: (line 1044): Port[0x73bf1ba100:audio:1:0:local:Net[any:0:0:0:x:x:x:x:x/0:Wildcard:id=0]]: Port completed gathering candidates.
W/thread.cc: (line 271): Waiting for the thread to join, but blocking calls have been disallowed
I/chatty: uid=10153(u0_a153) network_thread identical 1 line
W/thread.cc: (line 271): Waiting for the thread to join, but blocking calls have been disallowed
W/stunport.cc: (line 400): Port[0x73bf1b8300:audio:1:0:local:Net[rmnet0:2409:893c:46:x:x:x:x:x/64:Cellular:id=4]]: StunPort: stun host lookup received error 0
I/basicportallocator.cc: (line 1044): Port[0x73bf1b8300:audio:1:0:local:Net[rmnet0:2409:893c:46:x:x:x:x:x/64:Cellular:id=4]]: Port completed gathering candidates.
W/thread.cc: (line 271): Waiting for the thread to join, but blocking calls have been disallowed
E/stunport.cc: (line 529): sendto : [0x00000016] Invalid argument
W/thread.cc: (line 271): Waiting for the thread to join, but blocking calls have been disallowed
W/stunport.cc: (line 400): Port[0x73bf1b8d00:audio:1:0:local:Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=2]]: StunPort: stun host lookup received error 0
I/basicportallocator.cc: (line 1044): Port[0x73bf1b8d00:audio:1:0:local:Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=2]]: Port completed gathering candidates.
W/thread.cc: (line 271): Waiting for the thread to join, but blocking calls have been disallowed
I/basicportallocator.cc: (line 1446): Net[rmnet_ims:2409:8807:828b:x:x:x:x:x/64:Unknown:id=5]: Allocation Phase=Tcp
I/basicportallocator.cc: (line 1446): Net[rmnet0:2409:893c:46:x:x:x:x:x/64:Cellular:id=4]: Allocation Phase=Tcp
I/basicportallocator.cc: (line 1446): Net[rmnet0:10.154.222.x/32:Cellular:id=3]: Allocation Phase=Tcp
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=6358325334342f41386a6d62
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=6358325334342f41386a6d62, use_candidate=0, nomination=0
I/basicportallocator.cc: (line 1446): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=2]: Allocation Phase=Tcp
I/basicportallocator.cc: (line 1446): Net[lo:127.0.0.x/8:Loopback:id=1]: Allocation Phase=Tcp
I/basicportallocator.cc: (line 1446): Net[any:0.0.0.x/0:Wildcard:id=0]: Allocation Phase=Tcp
(line 1446): Net[any:0:0:0:x:x:x:x:x/0:Wildcard:id=0]: Allocation Phase=Tcp
I/AudioTrack: audioTrack continuously zero data counter >= 20, send pause/stop state to pg, counter: 20
I/port.cc: (line 1697): Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583267012094|-]: Sent STUN ping, id=77792b37674f347677726468, use_candidate=0, nomination=0
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=696a41397a714e783857797a
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=696a41397a714e783857797a, use_candidate=0, nomination=0
I/port.cc: (line 1697): Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583267012094|-]: Sent STUN ping, id=4b373454454f4d5661616349, use_candidate=0, nomination=0
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=2b526659664675734d424135
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=2b526659664675734d424135, use_candidate=0, nomination=0
E/stunport.cc: (line 529): sendto : [0x00000016] Invalid argument
W/GrallocMapperPassthrough: buffer descriptor with invalid usage bits 0x2080000
I/SurfaceEglRenderer: fullscreen_video_view: Reporting first rendered frame.
I/SurfaceEglRenderer: fullscreen_video_view: Reporting frame resolution changed to 1280x720 with rotation 270
W/GrallocMapperPassthrough: buffer descriptor with invalid usage bits 0x2080000
I/SurfaceViewRenderer: fullscreen_video_view: onMeasure(). New size: 1080x1920
I/EglRenderer: fullscreen_video_viewsetLayoutAspectRatio: 0.5625
I/port.cc: (line 1697): Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583267012094|-]: Sent STUN ping, id=35735630456a323343584b74, use_candidate=0, nomination=0
I/basicportallocator.cc: (line 909): Port[0x73bf1b8800:audio:1:0:local:Net[rmnet0:10.154.222.x/32:Cellular:id=3]]: Gathered candidate: Cand[:842163049:1:udp:1685921535:117.136.94.x:5218:stun:10.154.222.237:43295:P3eV:u8YRL96zzZPXxAMVH3KA0R0q:3:900:0]
(line 1044): Port[0x73bf1b8800:audio:1:0:local:Net[rmnet0:10.154.222.x/32:Cellular:id=3]]: Port completed gathering candidates.
D/CallRTCClient: ========onIceCandidate=======
I/AppRTCUtils: randomPoz:47 charSetLen:62
randomPoz:42 charSetLen:62
randomPoz:37 charSetLen:62
randomPoz:47 charSetLen:62
randomPoz:19 charSetLen:62
randomPoz:13 charSetLen:62
I/AppRTCUtils: randomPoz:56 charSetLen:62
randomPoz:28 charSetLen:62
randomPoz:2 charSetLen:62
randomPoz:56 charSetLen:62
randomPoz:24 charSetLen:62
randomPoz:31 charSetLen:62
D/WSRTCClient: trickleCandidate transactionID: nullvqlvTN4cC4Yf
C->WSS: {"janus":"trickle","candidate":{"candidate":"candidate:842163049 1 udp 1685921535 117.136.94.100 5218 typ srflx raddr 10.154.222.237 rport 43295 generation 0 ufrag P3eV network-id 3 network-cost 900","sdpMid":"audio","sdpMLineIndex":0},"transaction":"nullvqlvTN4cC4Yf","session_id":6886760364035320,"handle_id":5790718854272780}
D/WebSocketChannelClient: C->WSS: {"janus":"trickle","candidate":{"candidate":"candidate:842163049 1 udp 1685921535 117.136.94.100 5218 typ srflx raddr 10.154.222.237 rport 43295 generation 0 ufrag P3eV network-id 3 network-cost 900","sdpMid":"audio","sdpMLineIndex":0},"transaction":"nullvqlvTN4cC4Yf","session_id":6886760364035320,"handle_id":5790718854272780}
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=65466f4a2b63392b64575a6a
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=65466f4a2b63392b64575a6a, use_candidate=0, nomination=0
I/port.cc: (line 1697): Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583267012094|-]: Sent STUN ping, id=306f556f694c5a664f666658, use_candidate=0, nomination=0
I/port.cc: (line 1227): Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583267012094|-]: Received STUN ping, id=1e6a30347dbb0e8fc73629ca
I/port.cc: (line 799): Port[0x73bf1b8800:audio:1:0:local:Net[rmnet0:10.154.222.x/32:Cellular:id=3]]: Sent STUN ping response, to=188.213.167.x:45290, id=1e6a30347dbb0e8fc73629ca
I/p2ptransportchannel.cc: (line 956): Not switching the selected connection on controlled side yet: Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|CR-I|-|1|0|8655919583267012094|-]
I/p2ptransportchannel.cc: (line 2273): Selecting connection for triggered check: Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|CR-I|-|1|0|8655919583267012094|-]
I/port.cc: (line 1697): Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|CR-I|-|1|0|8655919583267012094|-]: Sent STUN ping, id=43487173613677746b4d676b, use_candidate=0, nomination=0
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=7171755a7259547479464a37
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=7171755a7259547479464a37, use_candidate=0, nomination=0
I/port.cc: (line 1227): Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|CR-I|-|1|0|8655919583267012094|-]: Received STUN ping, id=f64413354cbf2036de5c5319
I/port.cc: (line 799): Port[0x73bf1b8800:audio:1:0:local:Net[rmnet0:10.154.222.x/32:Cellular:id=3]]: Sent STUN ping response, to=188.213.167.x:45290, id=f64413354cbf2036de5c5319
I/p2ptransportchannel.cc: (line 2273): Selecting connection for triggered check: Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|CR-I|-|1|0|8655919583267012094|-]
I/port.cc: (line 1697): Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|CR-I|-|1|0|8655919583267012094|-]: Sent STUN ping, id=5a52527062587133337a6a50, use_candidate=0, nomination=0
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=7a70354b63366f5875577031
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=7a70354b63366f5875577031, use_candidate=0, nomination=0
I/port.cc: (line 1643): Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|CR-I|-|1|0|8655919583267012094|-]: Received STUN ping response, id=4b373454454f4d5661616349, code=0, rtt=413, pings_since_last_response=7338594871705371424a7051 77792b37674f347677726468 4b373454454f4d5661616349 35735630456a323343584b74 306f556f694c5a664f666658 ... 2 more
(line 1841): Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:A1XHutCR:1:0:local:udp:10.154.222.x:43295->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|CRWS|-|1|0|8655919583267012094|413]: Updating local candidate type to prflx.
I/basicportallocator.cc: (line 1247): Signaling candidates from the any address ports.
(line 1150): All candidates gathered for audio:1:0
I/p2ptransportchannel.cc: (line 782): P2PTransportChannel: audio, component 1 gathering complete, but using continual gathering so not changing gathering state.
(line 267): Switching selected connection due to: candidate pair state changed
(line 1688): Channel[audio|1|R_]: New selected connection: Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:vP/w82/4:1:0:prflx:udp:117.136.94.x:11023->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|CRWS|S|1|0|7961553800857125375|413]
I/dtlstransport.cc: (line 791): DtlsTransport[audio|1|__]: configuring DTLS handshake timeout 826 based on ICE RTT 413
I/opensslstreamadapter.cc: (line 766): BeginSSL with peer.
I/dtlstransport.cc: (line 704): DtlsTransport[audio|1|__]: DtlsTransport: Started DTLS handshake
I/srtptransport.cc: (line 349): The params in SRTP transport are reset.
I/port.cc: (line 932): Port[0x73bf1ba100:audio:1:0:local:Net[any:0:0:0:x:x:x:x:x/0:Wildcard:id=0]]: Port deleted
I/basicportallocator.cc: (line 1258): Port[0x73bf1ba100:audio:1:0:local:Net[any:0:0:0:x:x:x:x:x/0:Wildcard:id=0]]: Removed port from allocator (5 remaining)
E/stunport.cc: (line 529): sendto : [0x00000016] Invalid argument
I/opensslstreamadapter.cc: (line 1087): Accepted peer certificate.
D/WebSocketChannelClient: WSS->C: {
"janus": "ack",
"session_id": 6886760364035320,
"transaction": "null4DJp82Dhd82S"
}
D/WebSocketChannelClient: WSS->C: {
"janus": "ack",
"session_id": 6886760364035320,
"transaction": "nullsuCGeUGDDBAq"
}
WSS->C: {
"janus": "ack",
"session_id": 6886760364035320,
"transaction": "nullMa35cHAUQSmJ"
}
I/WSRTCClient: Got wsmsg:{
"janus": "ack",
"session_id": 6886760364035320,
"transaction": "null4DJp82Dhd82S"
}
I/回环: ========================={"janus":"ack","session_id":6886760364035320,"transaction":"null4DJp82Dhd82S"}
W/ViewRootImpl[Toast]: EGLdebug relayoutWindow Surface isSurface(name=null) (appName =org.appspot.apprtc) ( mSurfaceControllerIsValid =false) (mNativeObject =0)/@0xd2b5809,viewVisibility is0
W/ViewRootImpl[Toast]: EGLdebug relayoutWindow Surface isSurface(name=null) (appName =org.appspot.apprtc) ( mSurfaceControllerIsValid =true) (mNativeObject =73a3ee1000)/@0xd2b5809,relayoutResult is7
E/stunport.cc: (line 529): sendto : [0x00000016] Invalid argument
I/CameraStatistics: Camera fps: 26.
W/androidnetworkmonitor.cc: (line 376): Get an unknown type for the interface rmnet_ims
I/port.cc: (line 1796): Conn[0x73bf0c1800:audio:Net[rmnet0:10.154.222.x/32:Cellular:id=3]:vP/w82/4:1:0:prflx:udp:117.136.94.x:11023->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|CRWS|S|1|0|7961553800857125375|416]: Updating local candidate type to srflx.
I/basicportallocator.cc: (line 909): Port[0x73bf1b9c00:audio:1:0:local:Net[any:0.0.0.x/0:Wildcard:id=0]]: Gathered candidate: Cand[:842163049:1:udp:1677729535:117.136.94.x:5219:stun:10.154.222.237:37222:P3eV:u8YRL96zzZPXxAMVH3KA0R0q:0:999:0]
W/basicportallocator.cc: (line 914): Discarding candidate because port is already done gathering.
I/basicportallocator.cc: (line 1044): Port[0x73bf1b9c00:audio:1:0:local:Net[any:0.0.0.x/0:Wildcard:id=0]]: Port completed gathering candidates.
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=5a73387a2f356830414a6f30
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=5a73387a2f356830414a6f30, use_candidate=0, nomination=0
I/probe_controller.cc: (line 288): kWaitingForProbingResult: timeout
I/EglRenderer: fullscreen_video_viewDuration: 4006 ms. Frames received: 70. Dropped: 0. Rendered: 70. Render fps: 17.5. Average render time: 3948 ?s. Average swapBuffer time: 2253 ?s.
I/EglRenderer: pip_video_viewDuration: 4012 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=5a374a666c396431596e7a71
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=5a374a666c396431596e7a71, use_candidate=0, nomination=0
I/rtp_receiver_audio.cc: (line 71): Received first audio RTP packet
I/rtp_video_stream_receiver.cc: (line 293): Packet received on SSRC: 363841697 with payload type: 96, timestamp: 0, sequence number: 1, arrival time: 1606963774531
I/decoder_database.cc: (line 143): Initializing decoder with payload type '96'.
I/SurfaceEglRenderer: pip_video_view: Reporting first rendered frame.
pip_video_view: Reporting frame resolution changed to 640x480 with rotation 0
I/SurfaceViewRenderer: pip_video_view: updateSurfaceSize. Layout size: 1032x432, frame size: 640x480, requested surface size: 640x267, old surface size: 0x0
I/AudioTrack: audioTrack send start state to pg
I/SurfaceViewRenderer: pip_video_view: onMeasure(). New size: 576x432
I/SurfaceEglRenderer: pip_video_view: surfaceChanged: format: 4 size: 640x267
I/EglRenderer: pip_video_viewsetLayoutAspectRatio: 1.3333334
I/SurfaceViewRenderer: pip_video_view: updateSurfaceSize. Layout size: 576x432, frame size: 640x480, requested surface size: 576x432, old surface size: 640x267
I/SurfaceEglRenderer: pip_video_view: surfaceChanged: format: 4 size: 576x432
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=614e6243347168354556624a
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=614e6243347168354556624a, use_candidate=0, nomination=0
W/video_render_frames.cc: (line 61): Frame scheduled out of order, render_time=39278118, latest=39278140
W/libEGL: EGLNativeWindowType 0x73a3ee1010 disconnect failed
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=4e2b52414939514779695034
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=4e2b52414939514779695034, use_candidate=0, nomination=0
E/stunport.cc: (line 529): sendto : [0x00000016] Invalid argument
I/CameraStatistics: Camera fps: 30.
W/androidnetworkmonitor.cc: (line 376): Get an unknown type for the interface rmnet_ims
D/WebSocketChannelClient: WSS->C: {
"janus": "slowlink",
"session_id": 6886760364035320,
"sender": 5790718854272780,
"media": "video",
"uplink": true,
"lost": 6
}
I/WSRTCClient: Got wsmsg:{
"janus": "slowlink",
"session_id": 6886760364035320,
"sender": 5790718854272780,
"media": "video",
"uplink": true,
"lost": 6
}
I/回环: ========================={"janus":"slowlink","session_id":6886760364035320,"sender":5790718854272780,"media":"video","uplink":true,"lost":6}
D/WSRTCClient: Got a slowlink event on session 6886760364035320
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=43464b4f3152343256425347
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=43464b4f3152343256425347, use_candidate=0, nomination=0
W/nack_module.cc: (line 243): Sequence number 41 removed from NACK list due to max retries.
W/nack_module.cc: (line 243): Sequence number 44 removed from NACK list due to max retries.
I/AudioTrack: audioTrack continuously zero data counter >= 20, send pause/stop state to pg, counter: 20
W/nack_module.cc: (line 243): Sequence number 79 removed from NACK list due to max retries.
(line 243): Sequence number 83 removed from NACK list due to max retries.
(line 243): Sequence number 86 removed from NACK list due to max retries.
W/nack_module.cc: (line 243): Sequence number 93 removed from NACK list due to max retries.
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=616330533163316965667358
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=616330533163316965667358, use_candidate=0, nomination=0
D/WebSocketChannelClient: WSS->C: {
"janus": "slowlink",
"session_id": 6886760364035320,
"sender": 5790718854272780,
"media": "video",
"uplink": true,
"lost": 4
}
I/WSRTCClient: Got wsmsg:{
"janus": "slowlink",
"session_id": 6886760364035320,
"sender": 5790718854272780,
"media": "video",
"uplink": true,
"lost": 4
}
I/回环: ========================={"janus":"slowlink","session_id":6886760364035320,"sender":5790718854272780,"media":"video","uplink":true,"lost":4}
D/WSRTCClient: Got a slowlink event on session 6886760364035320
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=57356775424a4a2f7051526e
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=57356775424a4a2f7051526e, use_candidate=0, nomination=0
I/CameraStatistics: Camera fps: 30.
W/androidnetworkmonitor.cc: (line 376): Get an unknown type for the interface rmnet_ims
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=795651737977456456413830
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=795651737977456456413830, use_candidate=0, nomination=0
I/EglRenderer: pip_video_viewDuration: 4005 ms. Frames received: 21. Dropped: 3. Rendered: 18. Render fps: 4.5. Average render time: 10929 ?s. Average swapBuffer time: 2836 ?s.
I/EglRenderer: fullscreen_video_viewDuration: 4014 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 29.9. Average render time: 4027 ?s. Average swapBuffer time: 2301 ?s.
W/video_receive_stream.cc: (line 454): No decodable frame in 3000 ms, requesting keyframe.
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=53495561704247425034366b
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=53495561704247425034366b, use_candidate=0, nomination=0
I/AppRTCUtils: randomPoz:48 charSetLen:62
randomPoz:27 charSetLen:62
randomPoz:42 charSetLen:62
I/AppRTCUtils: randomPoz:0 charSetLen:62
randomPoz:21 charSetLen:62
randomPoz:14 charSetLen:62
randomPoz:0 charSetLen:62
randomPoz:57 charSetLen:62
randomPoz:30 charSetLen:62
randomPoz:0 charSetLen:62
randomPoz:40 charSetLen:62
randomPoz:38 charSetLen:62
D/WebSocketChannelClient: C->WSS: {"janus":"keepalive","session_id":6886760364035320,"transaction":"nullwbqAVOA5eAom"}
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=475867394d5036347759356b
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=475867394d5036347759356b, use_candidate=0, nomination=0
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=54764c5a6f6539513176634a
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=54764c5a6f6539513176634a, use_candidate=0, nomination=0
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=30596c68766233434b316378
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=30596c68766233434b316378, use_candidate=0, nomination=0
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=662f6c765846707030777363
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=662f6c765846707030777363, use_candidate=0, nomination=0
D/WebSocketChannelClient: WSS->C: {
"janus": "ack",
"session_id": 6886760364035320,
"transaction": "nullwbqAVOA5eAom"
}
I/WSRTCClient: Got wsmsg:{
"janus": "ack",
"session_id": 6886760364035320,
"transaction": "nullwbqAVOA5eAom"
}
I/回环: ========================={"janus":"ack","session_id":6886760364035320,"transaction":"nullwbqAVOA5eAom"}
I/WSRTCClient: Got an ack on session 6886760364035320
E/stunport.cc: (line 529): sendto : [0x00000016] Invalid argument
I/CameraStatistics: Camera fps: 30.
W/androidnetworkmonitor.cc: (line 376): Get an unknown type for the interface rmnet_ims
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=6e7777744452316e5a6c4770
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=6e7777744452316e5a6c4770, use_candidate=0, nomination=0
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=443734684c7878732b716959
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=443734684c7878732b716959, use_candidate=0, nomination=0
D/WebSocketChannelClient: WSS->C: {
"janus": "event",
"session_id": 6886760364035320,
"sender": 5790718854272780,
"plugindata": {
"plugin": "janus.plugin.videocall",
"data": {
"videocall": "event",
"result": {
"event": "hangup",
"username": "123456",
"reason": "We did the hangup"
}
}
}
}
I/WSRTCClient: Got wsmsg:{
"janus": "event",
"session_id": 6886760364035320,
"sender": 5790718854272780,
"plugindata": {
"plugin": "janus.plugin.videocall",
"data": {
"videocall": "event",
"result": {
"event": "hangup",
"username": "123456",
"reason": "We did the hangup"
}
}
}
}
I/回环: ========================={"janus":"event","session_id":6886760364035320,"sender":5790718854272780,"plugindata":{"plugin":"janus.plugin.videocall","data":{"videocall":"event","result":{"event":"hangup","username":"123456","reason":"We did the hangup"}}}}
D/WSRTCClient: Got a plugin event on session 6886760364035320
D/WebSocketChannelClient: WSS->C: {
"janus": "event",
"session_id": 6886760364035320,
"sender": 5790718854272780,
"plugindata": {
"plugin": "janus.plugin.videocall",
"data": {
"videocall": "event",
"result": {
"event": "hangup",
"username": "123456",
"reason": "Remote WebRTC hangup"
}
}
}
}
I/WSRTCClient: Got wsmsg:{
"janus": "event",
"session_id": 6886760364035320,
"sender": 5790718854272780,
"plugindata": {
"plugin": "janus.plugin.videocall",
"data": {
"videocall": "event",
"result": {
"event": "hangup",
"username": "123456",
"reason": "Remote WebRTC hangup"
}
}
}
}
I/回环: ========================={"janus":"event","session_id":6886760364035320,"sender":5790718854272780,"plugindata":{"plugin":"janus.plugin.videocall","data":{"videocall":"event","result":{"event":"hangup","username":"123456","reason":"Remote WebRTC hangup"}}}}
D/WSRTCClient: Got a plugin event on session 6886760364035320
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=373044326539335749794b48
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=373044326539335749794b48, use_candidate=0, nomination=0
D/WebSocketChannelClient: WSS->C: {
"janus": "hangup",
"session_id": 6886760364035320,
"sender": 5790718854272780,
"reason": "Close PC"
}
I/WSRTCClient: Got wsmsg:{
"janus": "hangup",
"session_id": 6886760364035320,
"sender": 5790718854272780,
"reason": "Close PC"
}
I/回环: ========================={"janus":"hangup","session_id":6886760364035320,"sender":5790718854272780,"reason":"Close PC"}
D/WSRTCClient: Got a hangup event on session 6886760364035320
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=5364464f69354d2b53336947
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=5364464f69354d2b53336947, use_candidate=0, nomination=0
I/CameraStatistics: Camera fps: 30.
I/audio_device_buffer.cc: (line 427): [REC : 10010msec, 48kHz] callbacks: 0, samples: 0, rate: 0, level: 0
(line 436): [PLAY: 10010msec, 48kHz] callbacks: 1005, samples: 482400, rate: 48192, level: 549
W/androidnetworkmonitor.cc: (line 376): Get an unknown type for the interface rmnet_ims
I/EglRenderer: fullscreen_video_viewDuration: 4004 ms. Frames received: 121. Dropped: 0. Rendered: 121. Render fps: 30.2. Average render time: 3958 ?s. Average swapBuffer time: 2303 ?s.
I/EglRenderer: pip_video_viewDuration: 4007 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=77493675657a754471717548
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=77493675657a754471717548, use_candidate=0, nomination=0
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=636a776d77713767742f2f64
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=636a776d77713767742f2f64, use_candidate=0, nomination=0
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=50674a63624336456a744f38
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=50674a63624336456a744f38, use_candidate=0, nomination=0
I/receive_statistics_proxy.cc: (line 521): Bad call (any) start: 39285917
(line 527): Bad call (fps) start: 39285917
E/stunport.cc: (line 273): Port[0x73bf1b9700:audio:1:0:local:Net[lo:127.0.0.x/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
W/port.cc: (line 1188): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Failed to send STUN ping err=-1 id=5463623679576d4871446761
I/port.cc: (line 1697): Conn[0x73bf0c2200:audio:Net[lo:127.0.0.x/8:Loopback:id=1]:STPry1IM:1:0:local:udp:127.0.0.x:49388->SpRXhopg:1:2015363327:local:udp:188.213.167.x:45290|C--I|-|0|0|8655919583266749950|-]: Sent STUN ping, id=5463623679576d4871446761, use_candidate=0, nomination=0
Disconnected from the target VM, address: 'localhost:8603', transport: 'socket'
