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Audio quality

Jan 30, 2012 8:37 AM

Tags: #audio #fms #stream #codec #fmg #workflow.xml #speex #main.asc

We are streaming conference calls from a PBX to the web using the FMG / FMS combination to convert the calls from SIP to RTMP. The RTMP stream is being passed to a CDN for delivery to the audience.


Our current deployement is resulting in very clipped / tinny audio - we have tried many configs on both the RTMP, Speex side and we have also tried serveral different Codecs (NM8k, NM22k and Speex) and volume options.


Please can someone advise regarding settings to produce the best quality audio.


We can output our SIP stream using the Speex codec - would it be possible to remove the need for any transcoding by FMG?


As we are passing the stream to a CDN we are doing all the processing in FMS e.g.


application.legService.onLeg = function(info) {

  application.legService.setWriteCodec(info.legID, "speex");

  application.legService.setWriteAudioGain(info.legID, 70);



  return true;



All advice greatly appreciated.



  • Currently Being Moderated
    Jan 30, 2012 8:53 AM   in reply to billybigpotatoe


             Getting flawless audio quality is definitly easy to achieve.


             Try out following steps: 

                  1) Stick to codec nm22k; this provides by far the best output quality with FMG.

                  2) In rtmp.xml; set <SilenceLevel> to 0 (inside <audio> tag)

                  3) Since your use case in only one way broadcast, On flash player set NetStream.bufferTime  to 0.1 (a non zero value); this will make sure that flash player isn't dropping audio packets. Usually, it is not required, but if client bandwidth is really choppy; this will help ensure quality.

                  4) Check in FMG core00.log; does FMG log errors indicating frequent packet drops ?


             I hope it helps



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  • Currently Being Moderated
    Jan 30, 2012 9:14 AM   in reply to billybigpotatoe

    Default codec pre-configured in sample sip.xml FMG sounds perfect (i.e. G711u); so no special reccomendations on codec between sip & FMG.

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  • Currently Being Moderated
    Jan 31, 2012 6:40 AM   in reply to billybigpotatoe

    Excessive gain causes harsh audio. Default audio gain is just fine in most cases. Try removing following call from your custom app:


      application.legService.setWriteAudioGain(info.legID, 70);

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  • Currently Being Moderated
    Jan 31, 2012 9:51 AM   in reply to billybigpotatoe

    To confirm, are you using G711u (default codec entry) in sip <profile> in sip.xml ?

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