[Asterisk-Users] RTP Packetization

[Dan Austin]

Sorry for the top post, i've only got a few seconds to respond.

The Async patch (-I) has nothing to do with packetization.  The
poster that added that information to the bug notes under 5162
was confused.

As to why it is not working, you said you set it on a peer.
Did that 'peer' call Asterisk, or did another device on
Asterisk call it?

Is the second device also using 30ms?  Do you have re-invites
enabled?  A re-invite to/from a device not told to use 30ms
won't use 30ms.

I use type 'friend' and get 30ms to/from my endpoints, and
since my server is primarily for MeetMe, I do not have reinvite
enabled.

Dan

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Patrick
Neubauer
Sent: Friday, May 19, 2006 7:27 AM
To: Asterisk Users Mailing List
Subject: [Asterisk-Users] RTP Packetization

Hi all,

I need to be able to adjust packet sizes and found the patch at 
http://bugs.digium.com/view.php?id=5162

Thus, I checked out and compiled 
http://svn.digium.com/view/asterisk-old/team/group/5162_rtp_packetizatio
n

I added the line "packetization = 30" for one peer in my sip.conf and 
started asterisk with the "-I" switch for async RTP.
That's all it takes according to the 5162 issue page. Nevertheless, 
asterisk still keeps sending it 20ms packets, even though a "sip show 
peer foobar" shows Packetization: 30.

What could be wrong? What about that ztdummy thing for internal timing? 
Is this necessary to run asterisk properly? Is it important for 
packetization?

Regards, Patrick
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