[asterisk-dev] AMI command for checking SIP channel QoS (proposal)

[Olle E. Johansson]

6 nov 2009 kl. 09.34 skrev Klaus Darilion:

> Hi Olle!
>
> Great that you are working on QoS again.
>
> It would be great if you could make a command to retrieve the QoS data
> of ALL ongoing SIP calls, maybe channel: SIP/* ?

Let's start somewhere and then extend when we have something working.  
Seems like there's a lot of issues in our RTCP implementation.

The addition of an extra IP makes me worried. If we get RTCP from  
another IP, then it's another SSRC and RTP session and another set of  
statistics. Maybe we should have a chain of reports per call or  
something... Do we restart RTCP when we reinvite audio, do we send end  
report before we hand over audio to go p2p? What's the intended  
behaviour?

I noticed while testing yesterday that the RTT was from the last  
report from the other side. Maybe we should have an average calculated?

/O