[Freeswitch-users] h323 - sip call is not working

[Nikolay Kondratyev]

Mmm. I doubt that it is an audio level problem.

Look at the picture that wireshark "voip calls -> player" shows for that
wheezy rbt.

Here it is:



 

This graph shows that FS really plays "stutter" tone instead of normal rbt.
Or am I mistaken?

 

And wireshark analysis of that rtp stream is strange.it shows 22 _seconds_
for that rtp stream..

Here is the text summary of wireshark rtp stream analysis of the stream from
FS to IPO:

Max delta = 50,65 ms at packet no. 1147 

Max jitter = 22556,56 ms. Mean jitter = 238,60 ms.

Max skew = 355392,29 ms.

Total RTP packets = 1538   (expected 1538)   Lost RTP packets = 0 (0,00%)
Sequence errors = 0 

Duration 36,14 s (513134 ms clock drift, corresponding to 121603 Hz
(+1420,03%)

 

I think the problem is on FS site.

 

Thanks and regards,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Monday, February 15, 2010 6:40 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

It looks like audio level issue .. can you lower the gain on IPO ?

T.

On Mon, Feb 15, 2010 at 4:26 PM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Sent again..

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Monday, February 15, 2010 4:50 PM


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

 

 

On Mon, Feb 15, 2010 at 10:46 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Tihomir, 

I've just sent the trace to your gmail address.


didn't get anything...
 

Nikolay.

 

 


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