[Asterisk-Users] codec quality

[Steven Langley]

Hi there

I am using Meetme and have been testing with 2 different codecs - GSM and
g.711 - these seem to be the only 2 free codecs which are supported by my
soft phones (built using the RTC Client API). All users will be using this
same softphone when communicating.

The quality of g.711 (ulaw) I have found to be good, but it uses too much
bandwidth. Although it sends less data, the quality of GSM is not great - it
is quite fuzzy and not pleasant. Is there any way to improve the quality of
this codec? Or perhaps it is just an inferior codec to others which transmit
at 13kbps or less (such as g.729 or ilbc)? Skype uses ilbc and the quality
seems really good.

Lastly, what is the overhead that is added onto the audio packets? For
instance, GSM (13 kbps) sends at about 40 kbps and g.711 (64 kbps) sends at
about 80 kbps?

Many thanks

Steven

 

 

 


  

 

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