[asterisk-dev] chan_sip RTCP

[John Todd]

On Dec 2, 2008, at 7:12 PM, Gregory Boehnlein wrote:

> Hello,
> 	I have been having a conversation with someone that is considering
> the possibility of sponsoring some paid development supporting  
> collection
> and logging of RTCP statistics. I have not been paying attention to
> developments in 1.6 and trunk as of late, so I'm not sure what the  
> current
> status of the implementation is. A quick google turned up this:
>
> http://www.johnlange.ca/tech-tips/asterisk/logging-asterisk-rtcp-statistics/
>
> Not sure if that is still the case. Anyone care to comment?


The RTCP stats still aren't 100% reliable at this point as far as I  
know, though they do seem to track local stats with some decent margin  
(more testing would be required on this, though.)  A structural  
problem seems to be evident if your media flows are re-INVITEd during  
a call, as the RTPAUDIOQOS stats only catch the data for one RTP  
stream.  Follow this thread:

   http://lists.digium.com/pipermail/asterisk-dev/2008-November/035061.html

I'm happy to be corrected if someone says that they're on-target these  
days.  I'd also be interested in hearing what anyone is using as a  
test rig to evaluate performance of RTCP statistics - can it be made  
available to everyone?

JT

---
John Todd
jtodd at digium.com        +1-256-428-6083
Asterisk Open Source Community Director