A little more testing handled.  If I prevent native bridging
the RTP frame size is honored in the default case and in the
peer case.  I don't have a way to test a user, but I suspect
that case works as well.

The failure in native bridging makes sense.  In that case
the endpoint will use what-ever packetization they are configured
for, if they have any options.

Dan

-----Original Message-----
From: Vishal Phirke [mailto:[EMAIL PROTECTED] 
Sent: Thursday, September 22, 2005 3:25 PM
To: Dan Austin; [email protected]
Subject: Re: [ooh323c-devel] Configurable packetization in Asterisk

Configurable packetization in AsteriskHi Dan,

This will be a really useful thing for channel driver. Earlier there was
no
way to set the packetsize and hence, even if channel driver negotiated a
particular packetsize, there wasn't a way to enforce that with rtp.

What I think is really required, that we should provide configs such as:

allow=mulaw(rx:240,tx:20), alaw(rx:120:tx:20)

Does SIP/SDP allow negotiating packetsizes???

Regards,
-Vishal


----- Original Message ----- 
From: Dan Austin
To: [email protected]
Sent: Wednesday, September 21, 2005 9:29 PM
Subject: [ooh323c-devel] Configurable packetization in Asterisk


I noticed this bug on Mantis, http://bugs.digium.com/view.php?id=5162 ,
and
thought
it would be slick if the chan_h323 based on ooh323c could support it.  I
have spent part
of the day trying to code it up, and seem to have made some progress.
The global settings works.  User seems to quietly fail, and I have not
tested the peer
yet.  I think the user setting is failing due to media re-invites, since
an
incoming
call triggers find_user(), but no user is matched.
I don't know if the bug will end up being added to the Asterisk CVS, but
would anyone
be interested in this?
Dan




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