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

Nanang Izzuddin nanang at pjsip.org
Thu Apr 10 09:21:23 EDT 2008


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 ]
>
>
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