Re: [Iaxclient-devel] Caller audio problem

[daniel h. <de...@to...>]

Steve Kann a =E9crit :

> daniel huhardeaux wrote:
>
>> Hi everybody,
>>
>> my setup: Linux Debian unstable, CVS from 1st november. Asterisk CVS=20
>> from 1st november (was the same with 09/25/04). Also tried the=20
>> iaxComm binary from 02/28/04 and the latest from 10/22/04.
>>
>> IAXComm -- Asterisk -- FWD IAX -- Echo test (613)
>>
>> My audio is ugly. It's noisy, incomprehensible. Sometimes (2% maxi)=20
>> you understand few words with a delay from 3~4 sec. The called party=20
>> audio (introduction) is perfect! If I call the demo from * on my box,=20
>> perfect to.
>>
>> I tried with Ulaw, ILBC (from asterisk), GSM, as preferred codec,=20
>> frames from 160 to 480. No Changes. On my laptop I have X-lite=20
>> running under wine as well as GnomeMeeting: both are working perfectly.
>>
>> If someone have an idea on what could be wrong, welcome.
>
>
>
> do you have the jitter buffer enabled on your asterisk box?

No. I tried with yes, dropcount=3D3 maxjitterbuffer=3D500=20
maxexcessbuffer=3D100: same result

>
> Which way is the audio poor?

alway caller -> called But I never tryied to call my EP so lets say it's=20
audio from my notebook to other party

>
> What happens if you call FWD directly?

I connect my iaxComm direct to FWD and stop *: same result. I call my=20
ATA286  BT SIP: I have perfect audio from ATA to iaxComm, other way same=20
as described. I also try removing trunk=3Dyes in iax.conf for FWD but=20
change nothing.

Anyway, all actual setup is working well with SIP EP. I was out of the=20
office last week, configuration was like this:

iaxComm -- Router/FW (Linux Debian) -- internet -- FW/My asterisk box --=20
SIP EP and had the same result. So seems that problem is coming from my=20
iaxComm

>
> -SteveK

Thanks for your help.
--=20
Daniel