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

Mon Feb 15 01:46:00 PST 2010

Tihomir, 

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

Nikolay.

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir
Culjaga
Sent: Friday, February 12, 2010 11:46 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] h323 - sip call is not working

On Fri, Feb 12, 2010 at 3:29 PM, Brian West <brian at freeswitch.org> wrote:

This is a rather broad assumption.  I have seen 40ms, 60ms and even 80ms in
the wild.  It all depends on what you want to do.  It lowers overhead and
increases efficiency on the wire.

/b

On Feb 12, 2010, at 5:24 AM, Nikolay Kondratyev wrote:

By the way, I know that one can use different packetization times for the
same codec, but I've never heard, that somebody really uses 30 ms for
G711Alaw. Always 20ms.

everything above 60 ms is a nonsense ... and ugly :) It screws your voice
quality not even thinking VBD (voice band data) over that line :).

Anyhow, Nikolay, your problem is broken IPO h323 stack and the know avaya
"flexibility" when interoping with other vendor equipments. Here IPO is
unable to negotiate a different framing size than the default and sadly this
is the core of the problem.

Please, can you send me two tcpdump captures of calls between IPO and FS:

1.	a capture with fast start & h245tunneling=true
2.	a captire with fast start & h245tunelling=true + h245inSetup

I just want to be sure of something.

T.

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