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

[Kristian Kielhofner]

On 4/24/07, John Todd <jtodd at loligo.com> wrote:
> At 9:21 AM -0400 2007/4/24, Andrew Kohlsmith wrote:
> >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...
> >
> >-A.
>
> Olle -
>    A manager command would be useful as well.  I'm a big believer in
> getting things into the Dialplan, too.  Any way a patch could be made
> for the two or three lines to create a variable while you're in there?
>


  I like the idea but I'd like to keep it in the dialplan.

  For now just something that could log the IPs of the actual RTP
endpoints in a standard "SIP trapezoid" type call setup would be a
great start.  Perhaps we need a new variable for this - ${RTPIP} or
similar?


-- 
Kristian Kielhofner