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billybigpotatoe
Participant
January 30, 2012
Question

Audio quality

  • January 30, 2012
  • 1 reply
  • 1303 views

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);

  application.legService.setLegStateSendRecv(info.legID);

 

  return true;

}

All advice greatly appreciated.

D.

This topic has been closed for replies.

1 reply

A
Anonymous
January 30, 2012

Hello,

         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

-Pankaj

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billybigpotatoeAuthor
Participant
January 30, 2012

Do you have any recomendations regarding codecs to use between SIP and FMG?

D.

A
Anonymous
January 30, 2012

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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