[asterisk-users] Asterisk / Audiocodes annoying issue - Seeking Suggestions

[Cory Andrews]

I have installed some AudioCodes analog gateways in conjunction with * and
am having an annoying problem on the mp-118-fxs side. call quality is decent
but on the mp118 you hear yourself in your ear with around 1/10 to 1/4 of a
second delay. this is very annoying and makes it difficult to carry on a
conversation.

 

i have tried nearly every combination of codec including g711 a-law. i have
tried every codec with and without silence suppression. even though it is
now disabled, i have also increased the minimum buffer for jitter correction
to 70ms and the jitter setting to 7 with no improvement in quality. i have
tried setting the maximum echo buffer to the maximum amount 128ms but no
improvement (i don't believe this affects anything anyways because by
default if should go that far anyways).

 

these sites are connected via an mpls vpn setup and the connection is very
stable between sites. running a pingtest (ping every 50ms) i am getting
consistent pings of between 71-72ms over a period of 10 minutes. 

that is why i have reduced the minimum jitter buffer to 0 and have set it to
the very aggresive setting of 0 (remember i changed it to default of 70/7
with no improvement). it seems like the unit is trying to play back my voice
in my ear like a regular phone would, but my voice is coming back to me only
after it reaches the far end first. is there any way to prevent me from
hearing my own voice in my ear at all or to filter that out? i do not hear
an echo of the person coming from the MP114 FXO unit, i only hear my own
voice delayed to myself.

 

If anyone is using Audiocodes, and has corrected similar problems, please
shoot me an email.

 

Thanks

 

 

Cory Andrews

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