[asterisk-users] One Way Audio with WebRTC (with external asterisk)
amit at avhan.com
Wed May 21 04:41:50 CDT 2014
You need to check if ICE / STUN is configured.
How are these extensions configured? If you are in private network, you
might have to disable DirectMedia / reInvite for calls going between 2
I hope this helps to resolve your issue.
*Thanks & Regards,*
On 5/21/2014 2:26 PM, Gary Shergill wrote:
> I've run into a slight issue when using WebRTC and two Asterisk boxes.
> I am using SIPml as the test WebRTC client.
> My two asterisk boxes, one of them is configured for WebRTC with websockets, etc (asteriskrtc.local) and the other is just a standard asterisk server (asteriskgary.local).
> Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to log in to the SIPml webpage and make a call from a SIP Phone to that WebRTC user, and vice versa, and all the media flows.
> When I try making a call from the other asterisk server (asteriskgary.local) to asteriskrtc.local (all routes are set up) I am seeing the following behaviour:
> - asteriskgary.local user, 1000, dials asteriskrtc.local number, 6901
> - 6901 sees the call and has the option to answer
> - 6901 answers the call
> - 6901 can hear 1000 talking
> - 1000 can not hear 6901
> The weird thing is, sometimes it works, sometimes it doesn't...
> I think it has something to do with the port destination changing when the call is answered but I'm not sure (wireshark suggests that, as it says "Port Unreachable").
> Has anyone tried this before and seen this issue? Or knows why it is and how to debug it? I can provide any logs required, I have some logs from when it works and doesn't.
> Thank you for your help.
> Kind Regards,
> Gary Shergill
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