Re: [SR-Users] Browser WebRTC transcoder

Hey Daniel, If you say so, you probably right... I did not try it because on the sipwise GitHub (https://github.com/sipwise/rtpengine) they mention: "Rtpengine does not (yet) support: Repacketization or transcodingPlayback of pre-recorded streams/announcementsRecording of media streamsZRTP" So I did not test it... I will give it a try. By the way, from where should I download the source code? Also, any "tricky" (common mistake like my last one on WebRTC TLS) I should care about before trying it? Thanks, Moacir To: sr-users(a)lists.sip-router.org From: miconda(a)gmail.com Date: Wed, 18 May 2016 21:18:01 +0200 Subject: Re: [SR-Users] Browser WebRTC transcoder Hello, kamailio+rtpengine should do this job quite well. Cheers, Daniel On 18/05/16 19:16, Moacir Ferreira wrote: A question for the community: What would be your best advice for a RTP proxy/transcoder to allow browser WebRTC calls to legacy VoIP? Moacir _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users