小米2s手机 在连接成功后 挂断会APP无响应
安卓版本5.0.2

相关LOG:

01-12 22:05:01.186 22905-24148/com.dds.webrtc I/org.webrtc.Logging: VideoCapturerAndroid: stopCapture
01-12 22:05:01.186 22905-24151/com.dds.webrtc I/org.webrtc.Logging: VideoCapturerAndroid: stopCaptureOnCameraThread
    VideoCapturerAndroid: Stop preview.
01-12 22:05:01.226 22905-24151/com.dds.webrtc I/org.webrtc.Logging: VideoCapturerAndroid: Release camera.
01-12 22:05:01.296 22905-24148/com.dds.webrtc I/org.webrtc.Logging: VideoCapturerAndroid: stopCapture done
01-12 22:05:01.306 22905-22905/com.dds.webrtc V/dds_webrtc: 关闭通道流
01-12 22:05:01.306 22905-24149/com.dds.webrtc D/MyLog: PeerConnection.Observer::onIceConnectionChange:CLOSED
01-12 22:05:01.306 22905-24149/com.dds.webrtc V/dds_webrtc: ice 状态改变 CLOSED
01-12 22:05:01.306 22905-24149/com.dds.webrtc D/MyLog: PeerConnection.Observer::onSignalingChange:CLOSED
**01-12 22:05:01.316 22905-24202/com.dds.webrtc W/art: Native thread exiting without having called DetachCurrentThread (maybe it's going to use a pthread_key_create destructor?):** Thread[41,tid=24202,Native,Thread*=0xa4563400,peer=0x1307b080,"IncomingVideoSt - 24202"]
01-12 22:05:01.326 22905-24148/com.dds.webrtc D/AudioTrackJni: StopPlayout@[tid=24148]
01-12 22:05:01.326 22905-24148/com.dds.webrtc I/org.webrtc.Logging: WebRtcAudioTrack: stopPlayout
01-12 22:05:01.336 22905-24148/com.dds.webrtc D/AudioRecordJni: StopRecording@[tid=24148]
01-12 22:05:01.336 22905-24148/com.dds.webrtc I/org.webrtc.Logging: WebRtcAudioRecord: stopRecording
01-12 22:06:51.153 22905-24135/com.dds.webrtc D/MyLog: WebSocketClient::onMessage
01-12 22:06:51.153 22905-24135/com.dds.webrtc D/MyLog: _remove_peer:{"eventName":"_remove_peer","data":{"socketId":"373d7007-f0c8-4d80-a87f-197ce9a52e1a"}}
01-12 22:06:51.153 22905-24135/com.dds.webrtc V/dds_webrtc: 关闭通道流
01-12 22:08:42.031 22905-24147/com.dds.webrtc D/MyLog: WebSocketClient::onClose:
01-12 22:08:42.041 22905-24147/com.dds.webrtc E/dds_JavaWebSocket: onClose:

I have deployed my Node js server on firebase and get the Url and replaced WSS inside app. I'm getting this error:

E/dds_JavaWebSocket: java.lang.IllegalArgumentException: unknown scheme: https
E/dds_JavaWebSocket: onClose:unknown scheme: https

Localy it is working fine when I run node js server locally and replace local ip inside app.

怎么编译报错呀

有么有大佬有办法呀？

我给服务器发送的IceCandidate之后
PC端那边总是显示以下错误是什么原因：
Uncaught (in promise) DOMException: Failed to execute 'addIceCandidate' on 'RTCPeerConnection': Error processing ICE candidate

是我的peerPeerConnection创建错误了吗

windows怎么弄啊

hello @ddssingsong
i've cloned both android & server side code. Android project is imported on Android Studio properly. But how can I start the server? I've installed NodeJS on my system (Ubuntu 18.04 LTS).

You mentioned this "Deploy turn and stun services : https://github.com/coturn/coturn.git", then start "server.js".

BUT how can I deploy "turn" & "stun" services?

Kindly help.

your sample code is not working

加不上群

please provide project with newer libjingle_peerconnection_so, because many methods deprecated in newer so file

多人通话怎么操作？CallMultiActivity似乎是空的activity，怎么调用邀请多人通话呢？看ReadMe有实现这个功能，但是不知道怎么去使用，谢谢！

Hi,

This issue happens frequently on both of self-compiled lib or official lib,

D/dds_test: addRemoteIceCandidate 2222 W/libEGL: EGLNativeWindowType 0xc6f39808 disconnect failed W/System.err: at java.util.ArrayList$Itr.next(ArrayList.java:860) at com.dds.skywebrtc.engine.webrtc.Peer.drainCandidates(Peer.java:298) at com.dds.skywebrtc.engine.webrtc.Peer.onSetSuccess(Peer.java:260) D/dds_test: receiveIceCandidate--Note5 E/rtc: # # Fatal error in: gen/sdk/android/generated_metrics_jni/../../../../../../../../usr/local/google/home/sakal/code/webrtc-aar-release/src/sdk/android/src/jni/jni_generator_helper.h, line 94 # last system error: 0 # Check failed: !env->ExceptionCheck() # D/dds_test: addRemoteIceCandidate D/dds_test: addRemoteIceCandidate 2222 D/SurfaceView: destroy() Surface(name=SurfaceView - com.dds.webrtc.debug/com.dds.core.voip.CallSingleActivity@a6429e@0[13507])/@0x34ab17f org.webrtc.SurfaceViewRenderer{a6429e V.E...... ........ 0,0-1440,2792} D/dds_test: receiveIceCandidate--Note5 A/libc: Fatal signal 6 (SIGABRT), code -6 (SI_TKILL) in tid 15343 (signaling_threa), pid 13507 (ds.webrtc.debug) D/dds_test: addRemoteIceCandidate addRemoteIceCandidate 2222

Or

E/stun_port.cc: (line 284): Port[556c6c00:audio:1:0:local:Net[lo:127.0.0.0/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22 W/port.cc: (line 1246): Conn[23fed000:audio:Net[lo:127.0.0.0/8:Loopback:id=1]:7QYlvr0f:1:0:local:udp:127.0.0.1:47610->uu7x8KeF:1:2122260223:local:udp:192.168.28.135:47115|C--W|-|0|0|9114193830701055486|-]: Failed to send STUN ping err=-1 id=2b596b4b37527533782f6976 I/port.cc: (line 1756): Conn[23fed000:audio:Net[lo:127.0.0.0/8:Loopback:id=1]:7QYlvr0f:1:0:local:udp:127.0.0.1:47610->uu7x8KeF:1:2122260223:local:udp:192.168.28.135:47115|C--W|-|0|0|9114193830701055486|-]: Sent STUN ping, id=2b596b4b37527533782f6976, use_candidate=0, nomination=0 W/System.err: java.util.ConcurrentModificationException W/libEGL: EGLNativeWindowType 0x702b695010 disconnect failed D/SurfaceView: destroy() Surface(name=SurfaceView - com.dds.webrtc.debug/com.dds.core.voip.CallSingleActivity@e1f20cc@0[20124])/@0x9074f15 org.webrtc.SurfaceViewRenderer{e1f20cc V.E...... ........ 0,0-1440,2960} D/SurfaceView: onWindowVisibilityChanged(0) true org.webrtc.SurfaceViewRenderer{e1f20cc V.E...... ......I. 0,0-1440,2960} of ViewRootImpl@ca09235[CallSingleActivity] W/System.err: at java.util.ArrayList$Itr.next(ArrayList.java:860) at com.dds.skywebrtc.engine.webrtc.Peer.drainCandidates(Peer.java:298) at com.dds.skywebrtc.engine.webrtc.Peer.onSetSuccess(Peer.java:260) E/rtc: # # Fatal error in: gen/sdk/android/generated_base_jni/jni/../../../../../../../sdk/android/src/jni/jni_generator_helper.h, line 85 # last system error: 0 # Check failed: !env->ExceptionCheck() # A/libc: Fatal signal 6 (SIGABRT), code -6 (SI_TKILL) in tid 21306 (signaling_threa), pid 20124 (ds.webrtc.debug) E/ds.webrtc.debu: No package ID ff found for ID 0xffffffff. I/org.webrtc.Logging: EglRenderer: Initializing EglRenderer

Further info of this can be seen in enclosed log,

SIGARBT6 Error from Caller in drainCandidate function.log

Server Running perfectly fine on IP = "52.167.201.188";

Tested on webrtc_android app:-
webSocket.connect(); returns connectFlag=false & socket not opening in webrtc_android app.So that login was unsuccessful after input the user name & tapping enter.

You can replace the IP like this in Android webrtc_android app.

`public class Urls {

// private final static String IP = "192.168.2.111";
public final static String IP = "52.167.201.188";`
in Android webrtc_android app & Test.

Please guide.

Thanks in advance

请问如何截取手机端的预览图片呢？

Hi, ddssingsong

Thank you for your lib,
I'm using the Nodejs version of the library both devices are connected but they have just shown their local camera and none of them didn't get any frame from remote

I sent some logs below

4003 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
2021-02-08 23:40:33.665 6569-7312/com.dds.webrtc I/org.webrtc.Logging: CameraStatistics: Camera fps: 15.
2021-02-08 23:40:34.206 6569-7307/com.dds.webrtc E/stun_port.cc: (line 561): sendto : [0x00000016] Invalid argument
2021-02-08 23:40:34.255 6569-7307/com.dds.webrtc I/turn_port.cc: (line 1315): Port[edad0c10:audio:1:0:relay:Net[wlan0:10.0.2.0/24:Wifi:id=5]]: TURN allocate request sent, id=693378736554596b79463138
2021-02-08 23:40:34.262 6569-7307/com.dds.webrtc I/turn_port.cc: (line 1315): Port[edad2010:audio:1:0:relay:Net[radio0:10.0.2.0/24:Cellular:id=3]]: TURN allocate request sent, id=66324333753856717a306456
2021-02-08 23:40:34.272 6569-7307/com.dds.webrtc E/turn_port.cc: (line 820): Port[edacfc10:audio:1:0:relay:Net[lo:127.0.0.0/8:Loopback:id=1]]: Failed to send TURN message, error: 22
2021-02-08 23:40:34.272 6569-7307/com.dds.webrtc I/turn_port.cc: (line 1315): Port[edacfc10:audio:1:0:relay:Net[lo:127.0.0.0/8:Loopback:id=1]]: TURN allocate request sent, id=4e44676f627a416635486655
2021-02-08 23:40:35.666 6569-7312/com.dds.webrtc I/org.webrtc.Logging: CameraStatistics: Camera fps: 15.
2021-02-08 23:40:36.945 6569-7304/com.dds.webrtc I/org.webrtc.Logging: EglRenderer: local_view_renderDuration: 4001 ms. Frames received: 60. Dropped: 0. Rendered: 60. Render fps: 15.0. Average render time: 2409 us. Average swapBuffer time: 1972 us.
2021-02-08 23:40:36.996 6569-7305/com.dds.webrtc I/org.webrtc.Logging: EglRenderer: remote_view_renderDuration: 4002 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.
2021-02-08 23:40:37.274 6569-7308/com.dds.webrtc W/warn_current_thread_is_deadlocked.cc: (line 19): Probable deadlock:
2021-02-08 23:40:37.274 6569-7308/com.dds.webrtc W/stacktrace.cc: (line 176): Could not translate address to symbolic information for address 4294967294 at stack depth 10
2021-02-08 23:40:37.274 6569-7308/com.dds.webrtc W/warn_current_thread_is_deadlocked.cc: [1/2] (line 20): #00 pc 005a5cc4 /data/app/~~d8tlQXuILbywpc-9LMZsxw==/com.dds.webrtc-Ez-RdrAmWn7CXk469PhY9w==/lib/x86/libjingle_peerconnection_so.so
#1 pc 003e2652 /data/app/~~d8tlQXuILbywpc-9LMZsxw==/com.dds.webrtc-Ez-RdrAmWn7CXk469PhY9w==/lib/x86/libjingle_peerconnection_so.so
#2 pc 003e251e /data/app/~~d8tlQXuILbywpc-9LMZsxw==/com.dds.webrtc-Ez-RdrAmWn7CXk469PhY9w==/lib/x86/libjingle_peerconnection_so.so
#3 pc 0058c94f /data/app/~~d8tlQXuILbywpc-9LMZsxw==/com.dds.webrtc-Ez-RdrAmWn7CXk469PhY9w==/lib/x86/libjingle_peerconnection_so.so
#4 pc 005858fc /data/app/~~d8tlQXuILbywpc-9LMZsxw==/com.dds.webrtc-Ez-RdrAmWn7CXk469PhY9w==/lib/x86/libjingle_peerconnection_so.so
#5 pc 005a05cd /data/app/~~d8tlQXuILbywpc-9LMZsxw==/com.dds.webrtc-Ez-RdrAmWn7CXk469PhY9w==/lib/x86/libjingle_peerconnection_so.so
#6 pc 005a052a /data/app/~~d8tlQXuILbywpc-9LMZsxw==/com.dds.webrtc-Ez-RdrAmWn7CXk469PhY9w==/lib/x86/libjingle_peerconnection_so.so
#7 pc 005a03c0 /data/app/~~d
2021-02-08 23:40:37.274 6569-7308/com.dds.webrtc W/warn_current_thread_is_deadlocked.cc: [2/2] 8tlQXuILbywpc-9LMZsxw==/com.dds.webrtc-Ez-RdrAmWn7CXk469PhY9w==/lib/x86/libjingle_peerconnection_so.so
#8 pc 000e6973 /apex/com.android.runtime/lib/bionic/libc.so
#9 pc 00078566 /apex/com.android.runtime/lib/bionic/libc.so
2021-02-08 23:40:37.669 6569-7312/com.dds.webrtc I/org.webrtc.Logging: CameraStatistics: Camera fps: 15.
2021-02-08 23:40:39.670 6569-7312/com.dds.webrtc I/org.webrtc.Logging: CameraStatistics: Camera fps: 15.
2021-02-08 23:40:40.169 6569-7311/com.dds.webrtc I/audio_device_buffer.cc: (line 431): [PLAY: 10008msec, 44kHz] callbacks: 1001, samples: 441441, rate: 44109, rate diff: 0%, level: 0
2021-02-08 23:40:40.946 6569-7304/com.dds.webrtc I/org.webrtc.Logging: EglRenderer: local_view_renderDuration: 4000 ms. Frames received: 59. Dropped: 0. Rendered: 59. Render fps: 14.7. Average render time: 2723 us. Average swapBuffer time: 2237 us.

Hello, are you still under development?
I want your reply. thanks.

本地步骤了java版本的服务器，要android要如何跟浏览器互通？是要在nodejs版本的吗？

Thanks for library.
I' m testing now your WebRTC demo project. Every thing is god, code well organized, but the fps is too low. How can I increase it?

问下libjingle_peerconnection_so.so这个文件怎么来的，有源码吗

I follow the blog https://blog.csdn.net/u011077027/article/details/86225524
But unable to connect from phone
Please guide me

12-17 16:07:30.289 886 886 E DEBUG : failed to readlink /proc/538/fd/94: No such file or directory
12-17 16:07:30.427 886 886 F DEBUG : *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** ***
12-17 16:07:30.428 886 886 F DEBUG : Build fingerprint: 'HUAWEI/FLA-AL20/HWFLA-H:9/HUAWEIFLA-AL20/9.1.0.137C00:user/release-keys'
12-17 16:07:30.428 886 886 F DEBUG : Revision: '0'
12-17 16:07:30.428 886 886 F DEBUG : ABI: 'arm'
12-17 16:07:30.428 886 886 F DEBUG : Happend: 'Thu Dec 17 16:07:30 2020
12-17 16:07:30.428 886 886 F DEBUG : '
12-17 16:07:30.429 886 886 F DEBUG : SYSVMTYPE: Art
12-17 16:07:30.429 886 886 F DEBUG : APPVMTYPE: Art
12-17 16:07:30.429 886 886 F DEBUG : pid: 32373, tid: 538, name: signaling_threa >>> com.dds.webrtc.debug <<<
12-17 16:07:30.429 886 886 F DEBUG : signal 6 (SIGABRT), code -6 (SI_TKILL), fault addr --------
12-17 16:07:30.429 886 886 F DEBUG : r0 00000000 r1 0000021a r2 00000006 r3 00000008
12-17 16:07:30.429 886 886 F DEBUG : r4 00007e75 r5 0000021a r6 d036a7cc r7 0000010c
12-17 16:07:30.429 886 886 F DEBUG : r8 d036a80c r9 ffffffff r10 cdab92cd r11 d253a7a8
12-17 16:07:30.429 886 886 F DEBUG : ip 00000041 sp d036a7b8 lr f1b6ee45 pc f1b66b6a
12-17 16:07:30.432 886 886 F DEBUG :
12-17 16:07:30.432 886 886 F DEBUG : backtrace:
12-17 16:07:30.432 886 886 F DEBUG : #00 pc 0001db6a /system/lib/libc.so (abort+58)
12-17 16:07:30.432 886 886 F DEBUG : #1 pc 001b95a9 /data/app/com.dds.webrtc.debug-EIKN1mbIjvqaFDukX3RLzA==/lib/arm/libjingle_peerconnection_so.so
12-17 16:07:30.432 886 886 F DEBUG : #2 pc 001b75d3 /data/app/com.dds.webrtc.debug-EIKN1mbIjvqaFDukX3RLzA==/lib/arm/libjingle_peerconnection_so.so
12-17 16:07:30.432 886 886 F DEBUG : #3 pc 0066b3f9 /data/app/com.dds.webrtc.debug-EIKN1mbIjvqaFDukX3RLzA==/lib/arm/libjingle_peerconnection_so.so
12-17 16:07:30.432 886 886 F DEBUG : #4 pc 0067f5cd /data/app/com.dds.webrtc.debug-EIKN1mbIjvqaFDukX3RLzA==/lib/arm/libjingle_peerconnection_so.so
12-17 16:07:30.432 886 886 F DEBUG : #5 pc 002a3e75 /data/app/com.dds.webrtc.debug-EIKN1mbIjvqaFDukX3RLzA==/lib/arm/libjingle_peerconnection_so.so
12-17 16:07:30.432 886 886 F DEBUG : #6 pc 002b341f /data/app/com.dds.webrtc.debug-EIKN1mbIjvqaFDukX3RLzA==/lib/arm/libjingle_peerconnection_so.so
12-17 16:07:30.432 886 886 F DEBUG : #7 pc 002b3303 /data/app/com.dds.webrtc.debug-EIKN1mbIjvqaFDukX3RLzA==/lib/arm/libjingle_peerconnection_so.so
12-17 16:07:30.432 886 886 F DEBUG : #8 pc 00065ea7 /system/lib/libc.so (__pthread_start(void*)+22)
12-17 16:07:30.432 886 886 F DEBUG : #9 pc 0001ed81 /system/lib/libc.so (__start_thread+32)

Thank you for the great lib, app is crashed and here stacktrace, Could you help me, what is the reason?

        at org.webrtc.PeerConnectionFactory.initialize(PeerConnectionFactory.java:291)
        at com.packagename.meeting.PeerConnectionHelper.createConnectionFactory(PeerConnectionHelper.java:221)
        at com.packagename.meeting.PeerConnectionHelper.lambda$onJoinToRoom$0$PeerConnectionHelper(PeerConnectionHelper.java:140)
        at com.packagename.meeting.-$$Lambda$PeerConnectionHelper$shvUap96rkuUOOTcV81kaXKQuAw.run(Unknown Source:6)
        at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1167)
        at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:641)
        at java.lang.Thread.run(Thread.java:923)```

Hi ,
Thanks for writing such a fantastic library, I want to improve my streaming video quality and to that in the callSession class i am setting width and height to be that of the host device, yet the quality is not that good, by that i mean the remote video is quite grainy and I would like to improve that if its possible. Is using latest version of webRTC going to help?, Generally my network connection is above 25Mbps, so i dont think network bandwidth is the issue

Where is the code of many-to-many voice and video conferencing?

com.dds.java.voip.CallMultiActivity.java have no codes.

您好，
我在我自己的项目中直接依赖最新的webrtc sdk：rg.webrtc:google-webrtc:1.0.28513
在很多手机上收集到的sdp中video codec都是不包含H264，只有使用高通芯片的手机可以支持H264；
但是如果导入您这个项目的libwebrtc module并且依赖它，sdp信息中就会支持H264，并且也可以找到Hwdecoder，对于高通、海思、MTK、Nvidia、MX.IMG.MSVDX都是支持的；
请问您在libwebrtc module中使用的是哪一个版本webrtc sdk，您是否有做过修改，或者是否有对libjingle_peerconnection_so及java层code做过修改，您是怎么解决对不同手机支持H264硬解的问题的；

ice 的 stun and turn 无法使用,导致无法透传,只能局域网内进行视频

多人通话的时候，对方用后置摄像头会出现镜像的问题，
如何设置

当接入视频时，对方的画面看不到显示黑屏，而且很容易闪退。

Audio sound is low. How to improve that?

俩台Android设备测试 1对1 和会议 声音都特别特别小。不知道哪里出了问题。

我想问一下，如果想做音视频美颜，和消除噪音、回声那些，应该从哪里入手呢

E/stun_port.cc: (line 284): Port[b923cb28:audio:1:0:local:Net[lo:127.0.0.0/8:Loopback:id=1]]: UDP send of 96 bytes failed with error 22
09-23 18:48:40.416 2420-8900/com.yunxin.webrtc W/port.cc: (line 1246): Conn[b962fc58:audio:Net[lo:127.0.0.0/8:Loopback:id=1]:3+huZJtc:1:0:local:udp:127.0.0.1:58095->FMAsLrZK:1:1686052607:stun:udp:1.193.36.223:55499|C--I|-|0|0|7241540810644667902|-]: Failed to send STUN ping err=-1 id=73724d7458506f784a543263
09-23 18:48:40.416 2420-8900/com.yunxin.webrtc I/port.cc: (line 1756): Conn[b962fc58:audio:Net[lo:127.0.0.0/8:Loopback:id=1]:3+huZJtc:1:0:local:udp:127.0.0.1:58095->FMAsLrZK:1:1686052607:stun:udp:1.193.36.223:55499|C--I|-|0|0|7241540810644667902|-]: Sent STUN ping, id=73724d7458506f784a543263, use_candidate=0, nomination=0
09-23 18:48:40.466 2420-8900/com.yunxin.webrtc I/port.cc: (line 1756): Conn[b96fc4b0:audio:Net[rmnet_data0:2409:8944:280:da8::/64:Cellular:id=4]:xLy1eMHD:1:0:local:udp:[2409:8944:280:da8:b014:d82f:8f0f:eab2]:35070->r+Lqy4Mi:1:2122005759:local:udp:[::1]:39111|C--I|-|0|0|9113945341073183230|-]: Sent STUN ping, id=6234667873452b6d46663148, use_candidate=0, nomination=0
09-23 18:48:40.486 2420-8900/com.yunxin.webrtc E/stun_port.cc: (line 561): sendto : [0x00000016] Invalid argument
09-23 18:48:40.516 2420-8900/com.yunxin.webrtc I/port.cc: (line 1756): Conn[b96fda40:audio:Net[rmnet_data0:10.152.174.88/30:Cellular:id=3]:x0AxrwmI:1:0:local:udp:10.152.174.90:53213->7KDriU1s:1:2121932543:local:udp:127.0.0.1:47844|C--I|-|0|0|9113630880747503102|-]: Sent STUN ping, id=794a496765566d77734d4471, use_candidate=0, nomination=0
09-23 18:48:40.536 2420-8900/com.yunxin.webrtc I/turn_port.cc: (line 1315): Port[b929ef30:audio:1:0:relay:Net[rmnet_data0:10.152.174.88/30:Cellular:id=3]]: TURN allocate request sent, id=416d637661626d6846366e42

帮我看一下吧,谢谢谢谢

Hello,
Thanks for this beautiful library & code sample.

I'm using your NodeJS server and on client side, I'm using the sample code which uses NodeJS server.

I can send text message from one peer to another during Audio/Video call (between 2 peers) using DataChannel but problem occurs while conference call. Assume 3 peers are connected to conference call. When I need to send message to other 2 Peers; one of these two peer receives the message, other one doesn't.

can you please, work on this message sending issue? I'd be helpful...

Thanks

在局域网下通信并且在同一个网段 视频通话只有本地画面没有远端画面 对讲也是没声音的 用的master的代码 服务器java版本
turn/stun服务器地址用demo里原来的 但因为是内网环境 所以无法使用 是否跟turn服务器有关?

Hello,
Thanks for the library. I used "libskywebrtc2" on my project; also used your NodeJS server as signalling server.

I need to implement a feature like this ... if two users (User A & B) established a video call, one user (assume, User A) can switch off his video streaming, so that other user (User B) can not see his (User A) video. "appr.tc" has this kind of feature too.

How can I achieve this using your library.
Thanks

Hello,
Thanks for the library.
I'd like to improve the video quality (during video call). how can I improve the quality?

First of all, thanks to @ddssingsong for the awesome work. I've some queries regarding IceServers (mainly for understanding).

`private static MyIceServer[] iceServers = {

        new MyIceServer("stun:stun.l.google.com:19302"),

        // 测试地址0
        new MyIceServer("stun:global.stun.twilio.com:3478?transport=udp"),
        new MyIceServer("turn:global.turn.twilio.com:3478?transport=udp",
                "79fdd6b3c57147c5cc44944344c69d85624b63ec30624b8674ddc67b145e3f3c",
                "xjfTOLkVmDtvFDrDKvpacXU7YofAwPg6P6TXKiztVGw"),
        new MyIceServer("turn:global.turn.twilio.com:3478?transport=tcp",
                "79fdd6b3c57147c5cc44944344c69d85624b63ec30624b8674ddc67b145e3f3c",
                "xjfTOLkVmDtvFDrDKvpacXU7YofAwPg6P6TXKiztVGw"),

        // 测试地址1
        new MyIceServer("stun:118.25.25.147:3478?transport=udp"),
        new MyIceServer("turn:118.25.25.147:3478?transport=udp",
                "ddssingsong",
                "123456"),
        new MyIceServer("turn:118.25.25.147:3478?transport=tcp",
                "ddssingsong",
                "123456"),
        // 测试地址2
        new MyIceServer("turn:157.255.51.168:3478?transport=udp",
                "ddssingsong",
                "123456"),
        new MyIceServer("turn:157.255.51.168:3478?transport=tcp",
                "ddssingsong",
                "123456")

};`

I've few questions, kindly answer.

1. Why IceServers are needed?
2. Why do You use 3 测试地址 (3 Test Address)?
3. Google' "stun" server is used. Why? for this case, why "turn" server didn't use?
4. Twilio' "stun" server and "turn" server (for UDP & TCP) is used. Why?
5. "118.25.25.147" who's server is that? For this, "stun" and "turn" server (for UDP & TCP) is used. Why?
6. "157.255.51.168" who's server is that? For this, only "turn" server (for UDP & TCP) is used. Why? why "stun" server didn't use?

I'll be very grateful, if You answer all those 6 question.
Thanks, again.

搭建好麻烦

大佬您好，我在局域网部署好了您提供的服务端代码，并且关闭了服务端电脑的防火墙，在浏览器中能正常视频，但是我在Android代码中把signal = "wss://47.254.34.146/wss";这里面的ip地址换成我的服务器内网ip（192.168.2.102），并且Android手机也连入了同一个局域网，却不能连接，由于我是内网使用，所以并没有部署turnserver，Android端日志显示websocket连接异常，具体的错误是java.net.ConnectException: failed to connect to /192.168.2.102 (port 443) after 90000ms: isConnected failed: EHOSTUNREACH (No route to host)，您能给我点指导吗？

Hi @ddssingsong ,

As per the title: does this source codes support ABS and do you have any plan to include QoS, Security (DTLS, SRTP) into this source codes?
I'm willing to support if you do so in term of testing and fine-tune if I can do it...

视频通话过程中回音和噪音比较严重，该怎么优化处理一下呢？

我想attach下源码 方便阅读

我看安卓上面一些协议在node服务端代码里面没有