[Freeswitch-users] Asynchronous PTIME

[Jan Riedinger]

You have to set it the paramater in the profile, which is used for the 
call, e. g. set in external.xml:
<param name="rtp-autofix-timing" value="false"/>

BR
    Jan



Am 29.12.2010 20:04, schrieb Saeed Ahmed:
> on which profile did you put that param?
>
> I've allnet phone if i use g729 with ptime:40 then i can't hear called 
> party properly, its choppy voice, i put autofix param in both internal 
> and external profiles, but it didn't help.
>
> if i use ptime: 20 then it works fine.
>
> On Sat, Dec 25, 2010 at 12:02 AM, Jan Riedinger <riedinger at sns.eu 
> <mailto:riedinger at sns.eu>> wrote:
>
>     Hi Serge,
>
>     I had similar problems, look for my thread "Problematic Behaviour
>     of FS
>     regarding ptime negotiation" in October. I could solve the problem by
>     setting "rtp-autofix-timing=false", which disabled the (too) smart
>     behaviour of FreeSwitch.
>
>     BR
>         Jan
>
>
>     Am 22.12.2010 23:09, schrieb Serge S. Yuriev:
>     > Hello,
>     >
>     > 2010-12-20 16:51:13.960502 [WARNING] mod_sofia.c:1036
>     Asynchronous PTIME
>     > not supported, changing our end from 20 to 60
>     >
>     > I'm getting this warning and client hears chopped sound :(
>     > That is "Our end"?
>     >
>     > Call flow is Cisco (sip) FS (sip) YATE (h323) MeraTransitSoftSwitch
>     > All but MVTS under my control.
>     >
>     > I doesn't see any clue in logs and can't reproduce this with my
>     testing
>     > via Blink/PortSIP/PhonerLite/Twinkle or Grandstream devices
>     >
>     > Which debug/logs I should take? Any ideas?
>     >
>     > Thanks a lot.
>     >
>     > btw how I can save debug into log not only console?
>
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