asterisk users - May 2014 - One Way Audio with WebRTC (with external asterisk)

Hi,

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

Hi Gary

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 
asterisk boxes.
I hope this helps to resolve your issue.

*Thanks & Regards,*
Amit Patkar


On 5/21/2014 2:26 PM, Gary Shergill wrote:
> Hi,
>
> 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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