Hi Again,
Apart from X-lite client I have also tried eStara, diax phone, iaxcomm and
some others. I have tried different codecs - GSM, aLAW uLAW. They all give
the same result. In the direction PSTN user ---> Softphone user the sound is
crystal clear (also tried on a dial-up connection), in the other direction
however the sound is a bit choppy. The chops occur at regular intervals of
time, at about 1-2 seconds !?
When analyzing *'s ethernet interface with tcpdump (raw tcpdump -i eth0) I
have noticed that the scrolling slows down during the times when chops occur
in the sound.
I have tested things using different softphones and different internet
connections (user side) - always yelding the same result. In other words
this is probably a problem on asterisk, either the hardware (ehternet
interface/E100p) or a swoftware bug, incoming RTP buffering maybe ?
Has anyone actually obtained a good quality sound in a similar setup ?
Internet 2 x E1
x-lite <-------> Asterisk -------> PSTN
Any help appreciated !
Best regards,
Dave
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nicolas
Gudino
Sent: Friday, January 02, 2004 6:35 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] one way choppy sound problem !
I have a similar problem, with GS phones, X-Lite or Kphone. I tried all
the codecs with the same result. Choppy sound in the direction SIP-Phone
-> pstn, but crystal clear sound the other way around. The only
difference in my case is that I have two asterisks servers connected
together via IAX2, the PSTN call is received in one asterisk, while the
sip phones are in the other asterisk. Ex:
pstn -> * --iax2--> * ->sip phone (GS, Xlite or Kphone)
If I use an Xlite in the same asterisk as the pstn line, the sound is
perfect in both ways. But when I answer the call in the second asterisk,
the sound from the sip phone to pstn is choppy, with or without silence
detection, and the sound from pstn to sip phone is perfect.
The asterisk server with the pstn line is an old pentium 133, maybe
thats the problem, I will try with a better machine and see how it goes.
On Fri, 2004-01-02 at 06:23, Dawid Mielnik wrote:
> Hi all,
>
> I have my asterisk setup as following:
>
> IP 2 x E1
> x-lite <-------> Asterisk -------> PSTN
>
>
> When I place a call from x-lite to PSTN, the quality of the sound in the
> direction x-lite -> PSTN is very bad. That is, the voice of the x-lite
user,
> heard by the PSTN user is choppy and makes communication not very
pleasant.
> The sound is choppy as if bits of data were lost. The strange thing is
that
> the x-lite user hears the PSTN user fine !
>
> In x-lite, I have swithed off sience detection (transmit silence - yes),
> this has improved the sound quality but did not eliminated the problem. I
> have fed a countinious sound into the microphone and still got chops in
the
> sound. I have also tried changing the codecs gsm, alaw, ulaw - but I get
the
> same problem with all of them. Maybe the problem lies somewhere in audio
> buffering settings on x-lite ?
>
> Has anyone ever had this sort of problem and managed to deal with it ? I
> would greatly appreciate your help !
>
> Best regards,
>
> Dave
>
>
> _______________________________________________
> Asterisk-Users mailing list
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>
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