On 07/07/14 06:40 PM, Yuriy Gorlichenko wrote:
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)
It would help to have the complete log messages from rtpengine. Make
sure you have a very recent version of Firefox, it used to have certain
WebRTC implementation problems.
cheers
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