[Asterisk-Dev] Input and testing.

[Dan Austin]

This is cool, and something my environment needs.  We don't use
SIP, so I spent this afternoon seeing if I could add this to
the ooH323c based chan_h323.

It turned out to be relatively easy and works like a champ
using the general level settings.  I have a slightly tricky
environment, so testing User and Peer level settings is difficult.

Still it works for me, and fits a decent need.  This thread and
the entry on Mantis doesn't seem to be generating much excitement.
Is there any value adding a diff for chan_h323 to bug-id 5162?

Dan
 

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Brian West
Sent: Sunday, September 04, 2005 11:30 PM
To: Asterisk Developers Mailing List
Subject: [Asterisk-Dev] Input and testing.

[patch] http://www.bkw.org/rtp.diff

Before you ask why please compare some various options:
http://www.packetizer.com/voip/diagnostics/bandcalc.html

This is something asterisk should have had from day one along with a
jitter
buffer in rtp.  Please test and provide feedback before I post it to the
bug
tracker.

/b


_______________________________________________
Asterisk-Dev mailing list
Asterisk-Dev at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-dev
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev