CODEWITHSUNDEEP
video-streamingwebrtc
Annamalai
Annamalai

Reputation: 11

WebRTC Connectivity / Streaming Issue in different networks

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 test via different networks(mobile 3G networks/Railwire type of network), the call is not established and no stream is visible. 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 49.0.2623.87m.

If I check the candidates emitting in different networks, its state on Peer A, is moving from 'new','checking', and finally 'failed'. On Peer B it stops at 'checking' state.Shall anybody share your ideas on this to resolve without using WebRTC Gateway.

Upvotes: 0

Views: 490

Answers (1)

Palash Borhan Uddin
Palash Borhan Uddin

Reputation: 394

Check if you have Relay and Server-reflexive candidates in SDP and those are exchanged correctly between two attendee.

Upvotes: 0

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