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

[Klaus Darilion]

Olle E. Johansson schrieb:
> 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?

How is it defined?

klaus