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
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 > > > >I have the exact same problem with my Grandstream phones (Snom's are fine) and others have reported it as well with GS phones but this is the first time I have heard of this problem in X-Lite.. All I can suggest is that you use the latest version of x-lite and see if it helps, other than that I have not been able to find the answer.. Good luck, and if you find a solution let us know.. Later..
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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
On Fri, 2004-01-02 at 11:35, Nicolas Gudino wrote:> 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.What is the ping times between your 2 asterisk servers? In the archive I have documented before that IAX jitter buffer sometimes has problems on short ping time links. At the time we where on a private T1 with 4ms ping times. We re enabled our jitter buffer now that we are on a DSL connection and our ping time is between 56 and 70 ms.> 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 > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- Steven Critchfield <critch@basesys.com>
I am having the same problem, but only with one specific user, so I believe it is network related. Anyone that can point me in the specific direction of what would cause this?> -----Original Message----- > From: WipeOut [mailto:wipe_out@users.sourceforge.net] > Sent: Monday, January 05, 2004 10:22 AM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] one way choppy sound problem ! > > Michael Van Donselaar wrote: > > >On Mon, 5 Jan 2004 13:29:06 +0100, "Dawid Mielnik" > <D.Mielnik@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. Theyall> 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 regularintervals> of > >>time, at about 1-2 seconds !? > >> > >> > > > >Are the PSTN interface and a network card sharing an interrupt? Ihad> similar > >problems with my X100P and a thunderlan dual ethernet card shringIRQs> (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'tdo> the > >trick with GSM conversion. DIAX and iaxComm, since they use the > iaxclient > >library, need to use GSM. > > > > > I have the same choppy sound problem on my server, my card is not > sharing an interrupt and I am using G711 which is not hittng the P2400> at all.. It seems there is a gremlin.. :) > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users