The switch is receiving audio frames larger than it should for the negotiated codec, which causes it to analyze the timing of received frames and correct the issue. To stop the switch from trying to fix the timing you should be able to uncomment the rtp-autofix-timing line in your related sofia profile, but I cant speak to whether that will fix your audio.<br>
<br>There is a logfile.conf.xml to alter any of the settings of how the switch logs to file. For debugging SIP to log file you should do it outside of the switch console, for instance with ngrep.<br><br><div class="gmail_quote">
On Wed, Dec 22, 2010 at 5:09 PM, Serge S. Yuriev <span dir="ltr"><<a href="mailto:me@nevian.org">me@nevian.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello,<br>
<br>
2010-12-20 16:51:13.960502 [WARNING] mod_sofia.c:1036 Asynchronous PTIME<br>
not supported, changing our end from 20 to 60<br>
<br>
I'm getting this warning and client hears chopped sound :(<br>
That is "Our end"?<br>
<br>
Call flow is Cisco (sip) FS (sip) YATE (h323) MeraTransitSoftSwitch<br>
All but MVTS under my control.<br>
<br>
I doesn't see any clue in logs and can't reproduce this with my testing<br>
via Blink/PortSIP/PhonerLite/Twinkle or Grandstream devices<br>
<br>
Which debug/logs I should take? Any ideas?<br>
<br>
Thanks a lot.<br>
<br>
btw how I can save debug into log not only console?<br>
--<br>
wbr,<br>
Serge<br>
<br>
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