About WEB RTS again

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 It seems there is a problem with WEBRTS.

I use opensips v2.1 as sip-proxy, clients: on one side is web-client ( sipml5, jssip demo webphones under fierfox or crome ), on the other side - pjsua. SIP - signaling is witout any problems. But when I call from web to softphone, pjsua is ringing, but when I "pick up" the phone call droppes, short beeps, session ends. I get "BY" with couse:488 (but both peers try to use g722).? When I call from pjsua to web, web-client lets me to pick up, connection is established, but no any media. If both clients is of the same type (web-web or pjsua - pjsua) then everything works fine, no problems. 

Can anyone explain how to overcome this problem?


Thanks!

Herman.




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