It seems to be related to this FAQ: http://trac.pjsip.org/repos/wiki/FAQ#tx-timing Regards, nanang On 10/04/2008, Alexey Trizno <xpg934 at mail.ru> wrote: > Hi! > > I try to do outgoing calls (in intranet, 100mbit/s) with the pjsua_vc8d.exe (winxp) > The incoming sound stream accurate and clear, but outgoing sound stream with awful jitter from 100 to 125 and even above. > Changing sound devices, Disable EAC and other - does not help. > On other workstation jitter = 0 (TX and RX). > Why so it turns out? X-Lite, sipXtapi (with PortAudio) and others work without problems with a sound on my workstation. > > 10:27:56.336 pjsua_app.c > [CONFIRMED] To: sip:999 at asterisk;tag=as43ab2cae > Call time: 00h:00m:12s, 1st res in 62 ms, conn in 62ms > SRTP status: Not active Crypto-suite: (null) > #0 PCMA @8KHz, sendrecv, peer=192.168.0.252:11780 > RX pt=8, stat last update: 00h:00m:04.016s ago > total 628pkt 100.4KB (125.6KB +IP hdr) @avg=63.9Kbps/79.9Kbps > pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) > (msec) min avg max last > loss period: 0.000 0.000 0.000 0.000 > jitter : 0.000 0.015 0.250 0.000 > TX pt=8, ptime=20ms, stat last update: 00h:00m:02.547s ago > total 622pkt 99.4KB (124.3KB +IP hdr) @avg 63.3Kbps/79.1Kbps > pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) > (msec) min avg max last > loss period: 0.000 0.000 0.000 0.000 > jitter : 110.000 110.062 110.125 110.125 > RTT msec : 9.002 9.002 9.002 9.002 > > pjsip compiled using default settings in VS2005 > > -- > Best regards, > Alexey [ xpg934 at mail.ru ] > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >
