[pjsip] big jitter in TX part... why?

Alexey Trizno xpg934 at mail.ru
Thu Apr 10 08:07:07 EDT 2008


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 ]





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