[sr-dev] routing time of sip request

[Juha Heinanen]

Henning Westerholt writes:

 > ah, i think we get this from a custom header extensions added from
 > the UA  (FritzBox). It measures the time from the sended INVITE to
 > the first received  

this does not measure "call setup time", i.e., invite in-out from
proxy.

 > provisional response or 200 OK. Not sure if you've this UA in your user set. 
 > Another option that i got from the german version is that is possible to use 
 > some external monitoring, i.e. just do some sampling.

yes, you can use a monitoring tool that sends requests to proxy and then
calculates availability and blocking.

-- juha