Hi, Joshua! I am very grateful to you for taking the opportunity to reply! It seems I cope with the problem. Thank you very much, Sincerely, Herman. >???????, 19 ???????? 2015, 12:14 -03:00 ?? Joshua Colp <jcolp at digium.com>: > > >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 > >_______________________________________________ >Visit our blog: http://blog.pjsip.org > >pjsip mailing list >pjsip at lists.pjsip.org >http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20150920/36312835/attachment.html>
