[SR-Users] Call from SipML5 -> PSTN

[Konstantin M.]

Jeremy, it is doesn't work at all. I've made a lot of changes to that
patched asterisk to make it working and no luck.
However, ast11 has fully supported webrtc, but I heard no voice during a
call.
Another issue is - sipml5 is sending a malformed Contact field, and
asterisk is trying to contact to invalid destination and finally closing a
call.

2012/11/28 Jeremy Jongepier <jeremy at autostatic.com>

> On 11/28/2012 05:48 PM, Raj Roy Ghandhi wrote:
>
>> Dear Peter,
>> Thansk for your fast response. I highly appreciate it.
>> Is there any way that I can convert the RTP/SAVPF into general media
>> profile that PSTN GW support ? So that I can get that call working/
>>
>> Best Regards,
>> Roy.
>>
>>
> I think at the moment only a patched Asterisk might be able to do this:
> http://code.google.com/p/**sipml5/wiki/Asterisk<http://code.google.com/p/sipml5/wiki/Asterisk>
> But I haven't got this to work yet because I get the feeling the patch
> Doubango provides is incomplete.
> Or maybe webrtc2sip by the same makers: http://code.google.com/p/**
> webrtc2sip/ <http://code.google.com/p/webrtc2sip/>
> I've tried setting up a webrtc2sip server today but it crashes after a few
> minutes, is horrible to set up (the web GUI is lightyears behind Siremis)
> and for the moment I can't get it to work because I can't find any proper
> documentation on how to set it up.
>
> If anyone could build webrtc support into rtpproxy or any other media
> proxy that can work together with Kamailio I'd be more than happy to test
> it.
>
> Regards,
>
> Jeremy
>
>
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