Two more information: 1. I've played with all suported codecs, same problems for all of them.
2. After aprox. 1 minute of conversation the delay problem doesn't occur, or better, it is very less(some miliseconds) than the begining(10 seconds) of a call. Any ideas!? Denis. Em Seg 17 Jan 2005 11:51, Denis Galvão - iSolve escreveu: > Hi Dan, Steve, Michael, Bruno and others. > > I will try to describe my VoIP environment below: > > SERVER: > - FC1 with Asterisk CVS-v1-0-11/04/04-23:47:17 > - iax.conf > [general] > bindport = 4569 > bindaddr = 0.0.0.0 > delayreject=yes > disallow=all > allow=ulaw > allow=alaw > allow=gsm > tos=lowdelay > jitterbuffer=no > dropcount=2 > maxjitterbuffer=100 > maxexccessbuffer=100 > mailboxdetail=yes > > [1001] > callerid="Ramal 1001" <1001> > context=from-internal > host=dynamic > mailbox=1001 > notransfer=yes > port=4569 > secret=**** > type=friend > username=1001 > > [1002] > callerid="Ramal 1002" <1002> > context=from-internal > host=dynamic > mailbox=1002 > notransfer=yes > port=4569 > secret=**** > type=friend > username=1002 > > CLIENT 1001: > - Windows XP > - DIAX 0.9.9g > - Firefly 1.9.6 Build 3944 > - USB Phone NTP200E - Compatible with ATCOM USB Phone > - AMD 1.8Ghz with 256Mb > > CLIENT 1002: > - Windows XP > - DIAX 0.9.9g > - Firefly 1.9.6 Build 3944 > - USB Phone NTP200E - Compatible with ATCOM USB Phone > - AMD 1.66Ghz with 256Mb > > > ADDITIONAL INFORMATION > - All machines are in the same network(192.168.*.*) no firewall in the > middle; > - With Firefly I have a VERY GOOD conversation, without any delay; > - With DIAX I have a one way delay of 10 sec. Only the person who recieve > the call get the delay, the person who make the call listen without > problems; > - Firefly in one side and DIAX in the other side, same delay problem; > - No problems with SIP; > - No problems(delay) with Linux clients runnig IaxComm 0.99pre11; > - Same problem with DIAX oldest DLL; > - Ping from clients to server: 0% packet loss and < 1ms; > - No problems calling PSTN, Voicemail, etc, just between DIAX clients; > > If you need something else, let me know! > > Thanks for your help! > > Denis Galvão. > > Em Dom 16 Jan 2005 19:52, Steve Kann escreveu: > > On Jan 16, 2005, at 2:53 PM, Dan wrote: > > > Hi Steve, > > > > > > ----- Original Message ----- From: "Steve Kann" <[EMAIL PROTECTED]> > > > > > >> On Jan 14, 2005, at 2:03 PM, Dan wrote: > > >>> Hi, > > >>> > > >>> \> Em Sex 14 Jan 2005 16:43, Dan escreveu: > > >>>>> > I dont have problems when calling PSTN extensions, and calling > > >>>>> > VoceMail, EchoTest, etc. The problem is related with the > > >>>>> > > >>>>> conversation > > >>>>> > > >>>>> > between two DIAX Softphones. > > >>>>> > > >>>>> Between 2 DIAX phone and the delay is in one direction only?? > > >>>> > > >>>> Yes. One direction only... Just who make the call get the delay. > > >>> > > >>> Then try > > >>> jitterbuffer=no > > >>> in iax.conf > > >>> to see if it solves this issue. > > >> > > >> Dan et. al, > > >> I think this might be a problem with native transfers, and needing > > >> to reset the jitterbuffer history when this happens, or something > > >> like this.. > > >> -SteveK > > > > > > But I have tried and I do don't have this problem here... > > > What can I do to make this happen here? > > > > I don't know... > > > > Maybe if we could get a packet trace of the situation that causes the > > problem? > > > > Maybe try notransfer or whatever the iax.conf parameter is, and see if > > that changes things. If it does, it points towards this being the > > problem. > > > > If the delay goes down after a couple of minutes after the transfer, > > this could be the problem. If it doesn't, there's something else > > really wrong.. > > > > (I'm assuming you're using the new JB code here..). Also, if you're > > using the new JB code, you should implement the stuff to get the > > network stats, so we can see if calculated jitter is substantially > > higher..) > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users