28 Nov
2012
28 Nov
'12
8:14 p.m.
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