8 Jul
2014
8 Jul
'14
1:40 a.m.
Hello. I have Kamailio 4.1.3+rtpengine_rtpproxy-ng as module for rtpengine.
Kamailio installed as frontend (registration, auth, proxy ) of asterisksk
servers.
WEBRTC users registred at kamailio and asterisk works as media server.
When I try to call from Jssip from Firefox to chrome to way audio is fine.
When I call from chrome - I see rtp packets only from firefox. Not from
chrome.
At kamailio log when I call from chrome log the same as whe i try to call
from firefox (I can not see anithing wrong)
So somebody else have same issue and may be somebody know reasons of this.
Thanks for advice