[Asterisk-Users] one way choppy sound problem !

[Michael Van Donselaar]

On Mon, 5 Jan 2004 13:29:06 +0100, "Dawid Mielnik" <D.Mielnik at elka.pw.edu.pl>
wrote:

>
>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 !?

Are the PSTN interface and a network card sharing an interrupt?  I had similar
problems with my X100P and a thunderlan dual ethernet card shring IRQs (also
would make one of the ethernet ports fails until reboot)

Are you still using the P133?  I tried using a P120, but it wouldn't do the
trick with GSM conversion.  DIAX and iaxComm, since they use the iaxclient
library, need to use GSM.
>
>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: asterisk-users-admin at lists.digium.com
>[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Nicolas
>Gudino
>Sent: Friday, January 02, 2004 6:35 PM
>To: asterisk-users at lists.digium.com
>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
>>
>>
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>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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