[asterisk-users] H323 situation

[Luis Silva]

Hi,
Still I can manage to have good incoming calls from h323. Can someone give
me a hand?

Regards,
LS 


	Date: Thu, 16 Jul 2009 15:46:43 +0100
	From: Luis Silva <luis.silva at dreamware.pt>
	Subject: [asterisk-users] H323 situation
	To: <asterisk-users at lists.digium.com>
	Message-ID: <00ab01ca0624$3c9f69b0$b5de3d10$@silva at dreamware.pt>
	Content-Type: text/plain; charset="us-ascii"

	Hi all,

	I have this installation:

	Asterisk 1.6.1.1  with h323 support, pwlib_v1_10_3 and
openh323_v1_18_0.

	I have a  problem that is, when a call comes from H323 and goes to a
Sip
	phone the asterisk sends two rtp streams to the sip. I checked this
with
	tcpdump, save the payload (voice is in G711u), one is the ringing
indication
	and the other is the voice coming from the user in h323 side. And
worst they
	go to the same port. This causes that in the sip phone there are
problems,
	when the call is answered sometimes we get the riging indication,
others a
	mix of the two with very bad sound quality and others(few) a god
audio call.


	The outgoing calls from sip to H323 are ok.

	I also tested an incoming call from a dahdi channel and from here
everything
	is ok, only one rtp stream and a good call.



	By the way I had other problem that I fixed, but don't know if it
was in the
	best way.

	The h323 box is a Cisco AS5300 (or 5350?) and when I was making
outgoing
	calls the AS disconnected all of them after 10 sec.

	 I investigated I noticed that the AS as a limitation to the G711
payload to
	20 ms, and asterisk was using 150 ms. I resolve this changing in
frame.c the
	codec value and recompile asterisk. There is simpler way to do this?
Like
	changing values in codec.conf?...



	Regards

	LS


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