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.

Halloo ! Colleagues ! Can anyone hear me ? >Понедельник, 14 сентября 2015, 14:20 +03:00 от Герман Чичикин <1393419@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. > >Can anyone explain how to overcome this problem? > > >Thanks! > >Herman. > > >_______________________________________________ >Visit our blog: http://blog.pjsip.org > >pjsip mailing list >pjsip@lists.pjsip.org >http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Yes, I'm not qualified to answer though. On Wed, Sep 16, 2015 at 4:39 AM, Герман Чичикин <1393419@mail.ru> wrote: > Halloo ! Colleagues ! Can anyone hear me ? > > > >Понедельник, 14 сентября 2015, 14:20 +03:00 от Герман Чичикин < > 1393419@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. > > > >Can anyone explain how to overcome this problem? > > > > > >Thanks! > > > >Herman. > > > > > >_______________________________________________ > >Visit our blog: http://blog.pjsip.org > > > >pjsip mailing list > >pjsip@lists.pjsip.org > >http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >

On 15-09-16 08:39 AM, Герман Чичикин wrote: > Halloo ! Colleagues ! Can anyone hear me ? > > >> Понедельник, 14 сентября 2015, 14:20 +03:00 от Герман Чичикин >> <1393419@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

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. >Суббота, 19 сентября 2015, 12:14 -03:00 от Joshua Colp <jcolp@digium.com>: > > >On 15-09-16 08:39 AM, Герман Чичикин wrote: >> Halloo ! Colleagues ! Can anyone hear me ? >> >> >>> Понедельник, 14 сентября 2015, 14:20 +03:00 от Герман Чичикин >>> < 1393419@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@lists.pjsip.org >http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org