WebRTC Audio/Video Calls not working with Different and Mobile Networks

[Annamalai]

Hi,

I have developed a simple P2P audio and video call application using WebRTC recent adapter.js, 'Adapter.js shimming chrome!' version. When the application is tested with in the same (cable) network, it works fine and the video,audio call gets connected most of the times. If i tested via different networks, like mobile 3G networks / Railwire type of network, the call is not established. No errors were shown specific on the logger added, except the candidate state mismatch. I used the following servers and technologies for my application development :

STUN : stun:stun.l.google.com:19302

TURN : turn:192.158.29.39:3478, username : "JZEOEt2V3Qb0y27GRntt2u2PAYA=", credential : "28224511:1379330808"

(Also tested with our own TURN server created using Google resource at https://code.google.com/archive/p/rfc5766-turn-server/)

Signaling : Node.js and Socket.IO, 

Chrome browser Version 49.0.2623.87 m.

If i check the candidates emitting and its state, its reaching 'ICEConnectionStateFailed' on Peer A and its still 'ICEConnectionState:Gathering' on the Peer B side, for the error case with different networks. Shall anybody share your ideas on this to resolve without using WebRTC Gateway.

[Honghai Zhang]

We are aware of an issue that the ICEConnection may reach a failed state at the beginning of the call if a connection is created and deleted quickly before any other connections are created. The error you described might be caused this issue. 

In your application, if you do not let ICEConnectionStateFailed end your call, it will fix the problem. 

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