On 11/28/2012 08:58 PM, Konstantin M. wrote:
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.
Hello Konstantin,
Thanks for the heads up. Those sound like issues that could be resolved.
No audio or one way audio is almost always either a codec or a NAT issue
and the malformed Contact field is something I think could be worked
around too.
Regards,
Jeremy
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