[asterisk-dev] Extensions to RTPAUDIOQOS for SIP/other channels

Andrew Kohlsmith akohlsmith-asterisk at benshaw.com
Tue Apr 24 06:21:25 MST 2007

On Tuesday 24 April 2007 8:39 am, Olle E Johansson wrote:
> > So, two questions:
> >
> >  1) Does RTPAUDIOQOS currently change back to zero when transfers/
> > re-invites occur?  (I could test this myself, I know, but it would
> > require a lot more acrobatics and hopefully someone just knows.)
> >
> >  2) Would anyone object to the addition of the "far-end" IP address
> > for a given RTP stream being included in the RTCPAUDIOQOS string
> > somewhere, or available otherwise within the dialplan?  If there is
> > any consensus that this might be useful, we'll code up and/or pay
> > for the patch to be submitted to the community.  This seems like a
> > trivial patch.
> I was working on a manager update with this information, propably #3
> in your list.

In what sense, a manager event whenever a reinvite occurs, and which dumps the 
RTPAUDIOQOS information from the old peer, then zeroing the stats?

It seems like we need two QOS stats... one which is reset on every reinvite, 
and one which is per-call... 


More information about the asterisk-dev mailing list