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@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga
Sent: Monday, February 15, 2010
6:40 PM
To:
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@nstel.ru> wrote:
Sent again..
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga
Sent: Monday, February 15, 2010
4:50 PM
To: freeswitch-users@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@nstel.ru> wrote:
Tihomir,
I’ve just sent the trace to your gmail
address…
didn't get anything...
Nikolay.
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