On 15-09-16 08:39 AM, ?????? ??????? wrote: > Halloo ! Colleagues ! Can anyone hear me ? > > >> ???????????, 14 ???????? 2015, 14:20 +03:00 ?? ?????? ??????? >> <1393419 at mail.ru>: >> >> 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. You won't be able to connect them without additional work. PJMEDIA has no support for DTLS, which is used in WebRTC to negotiate the keying material used for SRTP. As a result you will get no media flow. Something has to act as a gateway to fulfill all the WebRTC requirements. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
