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

Leif Madsen leif.madsen at asteriskdocs.org
Thu Apr 26 22:18:41 MST 2007

On Thursday 26 April 2007 20:29:56 John Todd wrote:
> I'm torn if this should be a channel-specific item, or if it should
> be left in some type of RTP-based value in the core CHANNEL
> functions.  I'm thinking the latter, since H323, Gtalk/Jingle, and
> SIP all use RTP and could benefit if the value was not specific to
> each channel type.
> At this point, I'd be happy if the value was just added to the end of
> the RTPAUDIOQOS blob like this, which seems pretty trivial to
> implement:
> ssrc=946918584;themssrc=3607239124;lp=7;rxjitter=0.005826;rxcount=1253;txji

Also to note is that RTPAUDIOQOS variable doesn't really work correctly, and 
thus you should pull these stats from the CHANNEL() function. Just in case 
you weren't aware of stats being there.


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