[asterisk-users] Asterisk 13 WebRTC Status report

[Johan Wilfer]

Den 2015-09-15 kl. 16:52, skrev asandovalros at gmail.com:
> Hello Marek! I’ve been running on an issue with my Asterisk 12
> configuration for using WebRTC on a LAN environment for about a month! I
> really need some help …
>
> My calls from the browser are done fine. I get ringing, they can be
> answered and never drop. The thing is that there is no audio on any
> side! But I don’t get any error or warning from JavaScript nor the
> Asterisk CLI. I’m using Asterisk 12 + jsSIP.
>
> If you could help me solving this I would be eternally greatful 😃 It’s
> for my grade project …
> These are my files:
> sip.conf: http://pastebin.com/kWwXpi4V
> http.conf: http://pastebin.com/ZwJWiiwf
> SIP debugging for client REGISTER: http://pastebin.com/GNZETtQb
> SIP debugging for extension call (Hello-World recording):
> http://pastebin.com/0PxjLwBb
>
> I followed these tutorials. If you have any other useful resource, I’d
> be glad if you could share it:
> http://stackoverflow.com/questions/26254980/websocket-connection-fails-with-asterisk-11
> http://blog.gmc.uy/2014/04/asterisk-12-ubuntu-1204-pjproject-srtp.html
>
> Furthermore, if I want to have a local Asterisk configuration, which
> should be the IP address for the http.conf + DTLS certificates?? I tried
> with localhost but RTP packets redirect to my eth IP.
>
> Thanks in advance!!!!!!!!!!

In asterisk you have "rtp set debug on" to see if you get rtp packets.
On your client you can start wireshark and look if RTP packets flow in 
both directions.

If you have RTP traffic, maybe you didn't attach the incoming media to 
an audio/video tag in your html. For example:

html: <video id="remoteView" autoplay></video>
In the event-handler for 'addstream' for the call, you have to attach 
the stream to #remoteView.

/Johan