[Freeswitch-users] Asynchronous PTIME

[Saeed Ahmed]

Thanks Jan, I tried that, but still with ptime: 40 the voice is choppy :(


On Thu, Dec 30, 2010 at 6:06 PM, Jan Riedinger <riedinger at sns.eu> wrote:

>  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> 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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>
>
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