Hello list, before to lost my time, I'd like know if someone have a WebRTC working configuration on Asterisk 13.11.0 SIP or PJSIP channel. Thank you Regards
Hi. It have big audio delay because using extenral ICE servers. Better to use kamailio/opensips + rpenigne infront 2016-09-09 0:36 GMT+03:00 Annus Fictus <annusfictus at gmail.com>:> Hello list, > > before to lost my time, I'd like know if someone have a WebRTC working > configuration on Asterisk 13.11.0 SIP or PJSIP channel. > > Thank you > > Regards > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 > http://www.asterisk.org/community/astricon-user-conference > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160909/8d63c0b3/attachment.html>
using in production last asterisk 13 + pjsip bundled + pjsip patch for RTP/SAVPF (search pjsip conf) + sipml5 version from roginvs https://github.com/DoubangoTelecom/sipml5/pull/238 Dne 08/09/2016 v 23:36 Annus Fictus napsal(a):> Hello list, > > before to lost my time, I'd like know if someone have a WebRTC working > configuration on Asterisk 13.11.0 SIP or PJSIP channel. > > Thank you > > Regards > > >
Hello, I mean a working configuration (SIP o PJSIP) without patches or code corrections. Thank you Regards El 09/09/2016 a las 03:47, marek cervenka escribi?:> using in production > > last asterisk 13 + pjsip bundled + pjsip patch for RTP/SAVPF (search > pjsip conf) + sipml5 version from roginvs > > https://github.com/DoubangoTelecom/sipml5/pull/238 > > > Dne 08/09/2016 v 23:36 Annus Fictus napsal(a): >> Hello list, >> >> before to lost my time, I'd like know if someone have a WebRTC >> working configuration on Asterisk 13.11.0 SIP or PJSIP channel. >> >> Thank you >> >> Regards >> >> >> > >