Добрый день, On Thu, 23 Dec 2010 11:48:50 -0500 Chris Burns <chris@cloudtel.com> wrote: Thank you so much for answers!

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.

So it's not wrong signaling problem it's somewhere in RTP, right?

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.

How I can analyse side "bad" RTP from? WireShark?

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.

Ok, got it.

On Wed, Dec 22, 2010 at 5:09 PM, Serge S. Yuriev <me@nevian.org> wrote:

...

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? -- wbr, Serge

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-- Serge S. Yuriev Lead VoIP engineer

If the VoIP industry develops like the PSTN industry, you will *never* see the end of this crap. The only FAX modems I've seen which fully comply with the ITU specs are my own ones, in spandsp. All the others did some wrong things in the early days of the specs. The competitors learned to tolerate each other's errors. After that nobody ever bothered to fix those errors. The same can be said almost every other group of ITU specs. Steve On 12/25/2010 07:16 AM, Brian West wrote:

sigh... You could also try setting your codec to XXX@60i

But in the end if the device is broken make them fix it... if we don't start making people fix broken stuff we'll never see the end of this crap.

/b

On Dec 24, 2010, at 5:02 PM, Jan Riedinger 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

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@sns.eu <mailto:riedinger@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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Saeed Ahmed wrote:

Thanks Jan, I tried that, but still with ptime: 40 the voice is choppy :( I just upgraded my FS v1.0.6 to today's git version and experienced this issue. After reading through the posts on this thread, I did a search for rtp-autofix-timing variable and found it is commented under conf/sip_profiles/internal.xml file. I removed the comment and did a sofia profile internal restart reloadxml under fs_cli and the problem went away. So far, so good. Thanks to anyone who suggested this option.

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