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A
Anonymous
February 4, 2011
Answered

flash phone calls audio doesn't sound good

  • February 4, 2011
  • 1 reply
  • 1956 views

Hi i have setup Flash media server and FMG on a local server. configured it as per quick start quide( default settings) . i am able to do sip calls from sample flash phone as well as call from flashphone to flashphone rtmp. Problem is the audio is terrible . When calling through flashphone i set both clients( all in local) to use speex. the audio quality is choppy and has lots of noise....

am i missing somthing in the setup ?

This topic has been closed for replies.
Correct answer

Hi Pankaj,

thanks for your reply.

yes both computers are running same sample flahsphone connecting to the server and

Yes switching to nm22k codec has significant sound quality improvement. The scratchy and Noisy audio is now clear and understandable with slight delay. Any idea on why the sound quality on speex is not good? it should have better quality than the NM codecs.

We're really looking forward to using this with VOIP that is why we're pushing for speex as codec.

here is a link to my cloud instance: http://50.16.102.49:8134/flashphone/flashphone.html so you can see for yourself. and maybe you can tell me if that is all that i can expect from my setup, and then i could close this thread. use : rtmp://50.16.102.49/telephony

Any more advise you can give me on this will be helpful.

/mike


Mike,

          NellyMoser 22Khz is quite versatile; it is OK to use it as preferred codec for VoIP calls. If you don't find anything wrong, your setup should be just fine. Meanwhile, keep an eye for future upgrades and share your feedback/experiences.

-Pankaj

1 reply

A
Anonymous
February 10, 2011

Reducing value of maximum microphone gain from "Max Gain" option should help. The Max gain control shows up at the bottom of the Sample Flash Phone UI while it is in a call.

-Pankaj

A
Anonymous
February 10, 2011

Hello Pankaj,

Yes reducing the mic gain did help clear out the audio a bit, and i noticed that when i switch off the mic on the other line, the audio is a lot better, way better! But then again switching off the microphone on one end would defeat the purpose of 2 way calling. There's got to be something to make the audio more bearable.  

I was wondering how they got the audio in adobe connect to sound so clear since it uses the same server architecture(fms+fmg). Has anybody else experienced this? Is this a limitation for the dev edition? 

A
Anonymous
February 16, 2011

Hi,

Ok Im still having problems with the audio quality for 2 way rtmp calling using speex.

I already tried setting the quality values in speex.xml from 8-10. i also tried playing around with the complexity, but no luck.

Does anybody here have any idea on how to get clear audio for calls ? I cant figue out why the sound is so terrible. There is no way i can use that for calls.

I am willing to give access to the sample flashphone app, that is included with the FMG install, in my fms instance so that you guys can see for yourself what im talking about. keep in mind that setup is single server install (fms+FMG) and all settings are default( as per isntall guide) only thing i changed is quality in speex.xml, maybe after testing you can tell me if thats is all i can expect of the whole setup. 

If youre interested in helping out you can contact me here : anba@splitmedialabs.com  and i can send you the link and access via email.

Thanks in progress.

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