Den 2015-09-15 kl. 16:52, skrev [email protected]:
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
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