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We are facing a strange issue in webrtc call i.e. in a connected webrtc audio only call when some one upgrade the call(add video) the audio tracks will drops from orignator side.

steps to reproduce the problem
1.make a audio only call between two peers A and B.
2.updgrade call to video by calling getUserMedia again from peer A.
3.call established.
4.A can hear audio and view video.
5.B cant hear audio.

What is the expected result?
onaddstream(e) e.stream should contain both audio and video tracks

What do you see instead?
only video track is there at B's side (recipient)

What version of the product are you using? On what operating system? Chrome 51/WIndows7

Please find the webrtc dump from below link Webrtc dump

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Mantu Nigam
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  • Please show actual code so we can find flaws in it. Also, this smells like Chrome, but please say which browser you are having problems with, as the solution [will differ atm](http://stackoverflow.com/questions/35504214/how-to-addtrack-in-mediastream-in-webrtc/35515536#35515536) unfortunately. – jib Aug 08 '16 at 13:29
  • Yes we are using chrome 51.We have tried the same code with node websocket signaling server and we are getting both of the tacks but not in actual solution. Please find the link of code https://github.com/mantu1987/idemo/blob/master/webtc.txt – Mantu Nigam Aug 10 '16 at 07:16

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