[SR-Users] Browser WebRTC transcoder

[Moacir Ferreira]

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 at lists.sip-router.org
From: miconda at 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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Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
  


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