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@lists.sip-router.org
From: mico...@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
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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
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