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We are developing an app using AppRTC. Audio and Video call from iOS-iOS and Android-Android are working fine but whenever we try to call from android to iOS or iOS to android, nothing happens after the call is accepted.

We have tried using the same Video codec (H264) on both android and iOS but the issue still persists.

Any assistance in this matter is highly appreciated.

App is based on webRTC vs websocket . Android studio 2.3.2 last version.

I already use:

https protocol , autoplay , android version 7.0 (min > android 5.0) . App works on all supported browsers only android webview generate error.

This is first lines of errors log in logcat (android studio last ver):

E/chromium: [ERROR:audio_manager_android.cc(264)] Unable to select audio device! E/cr_VideoCapture: allocate: manager.openCamera: SecurityException: validateConnectLocked:1112: Caller "com.testwebrtc.nikola.myapplication" cannot open camera "1" without camera permission at android.hardware.camera2.CameraManager.throwAsPublicException(CameraManager.java:628) at android.hardware.camera2.CameraManager.openCameraDeviceUserAsync(CameraManager.java:347) at android.hardware.camera2.CameraManager.openCamera(CameraManager.java:450) at org.chromium.media.VideoCaptureCamera2.startCapture(VideoCaptureCamera2.java:661)

Other variant for error:

[ERROR:web_contents_delegate.cc(199)] WebContentsDelegate::CheckMediaAccessPermission: Not supported.

This is error log from chrome/webview (from errorCallBack - getUserMedia ):

I am using this library: https://bintray.com/google/webrtc/google-webrtc

What I want to achieve (at least, at the beginning of my project) is render video locally. I am using this tutorial (which is the only one around the Internet) https://vivekc.xyz/getting-started-with-webrtc-for-android-daab1e268ff4. Unfortunately, the last line of code is not up-to-date anymore. The constructor needs a callback which I have no idea how to implement:

localVideoTrack.addRenderer(new VideoRenderer(i420Frame -> { // no idea what to put here }));

My code is exactly the same as in the posted tutorial. This is the very first step to make familiar with WebRTC technology in Android which I cannot figure out. My camera is capturing the video because I can see it in my log:

I/org.webrtc.Logging: CameraStatistics: Camera fps: 28.

The main issue is that I have no idea how to pass it to my SurfaceViewRenderer through a callback. Did anyone meet that problem? I'll really appreciate any help or suggestions.

Here is the official example app which is the only source but it is done differently than one in the tutorial, it's much more complicated: https://webrtc.googlesource.com/src/+/master/examples/androidapp/src/org/appspot/apprtc

The index.html file:

My question in github: https://github.com/webrtc/apprtc/issues/615 I can't config apprtc for signal server, just call video ok via wifi but via mobile network has no luck. Please view my config, I can't find any example for constands.py in anywhere. Here is my config:

I'm pretty new to forum and I really appreciate the passion that all of you use to solve the questions.

So I'm here because I'm struggling to integrate the GoogleWebRTC in my Xamarin Form project in the iOS part (I'm been able to make a native objC test app ), I've try to integrate it via CocoaPods with Sharpie Pod and bind it, but I'm not be able to resolve all the errors created in the bind on ApiDefinitions.cs.

After some time I found this project https://github.com/valentingrigorean/apprtc-ios-xamarin that has successfully bind the library and after some bug fix ( editing WebRTCBinding.csproj and remove the -lstdc++.6 linker and removed some errors in code) but when I try to add that binding library into my project and try to run return me :

I use official sample to create offer SDP in Android Chrome, we can find a=rtpmap:100 H264/90000 that meant it can support H264.

But if I build AppRTC(official Android sample) and use official prebuilt libraries version 1.0.25821, call createOffer then receive SDP in SdpObserver::onCreateSuccess, the SDP did not contain H264.

My test device is Oppo R15 (with MTK Helio P60, Android 8.1).

So, why WebRTC only support H264 in Chrome but not in native application with some Android devices?

I am working on video call app using Apprtc.Follow below mention libraries.

1. https://github.com/njovy/AppRTCDemo
2. https://github.com/Piasy/AppRTC-Android

When I change url to my custom server instead of apprtc server then video call is disconnected after 1 minute. I have lost connection with server.

To avoid the connection lost with server, we need to ping server in regular interval approx 30 sec.

But above mention AppRTC project are using jar file(autobanh.jar) to websocket connection, but in library sendPing mentod is private so not accessible.

Question 1 - There is nay way to ping websocket server.

Try after replace websocet library I had changed websocket library with below mention libraries

1. https://github.com/Koredotcom/android-kore-sdk/tree/master/BotsSDK/korebotsdklib/src/main/java/kore/botssdk/autobahn
2. https://github.com/martindale/soundtrack.io-android/tree/master/src/de/tavendo/autobahn

After recplacing websocket library, Now I am able to access sendPing method. But still I have connection lost after 60 seconds during video call.

Ping Method-

I build a cordova app,and want to use apprtc in app.

Environment: