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