Measuring latency


#1

Is there a way to measure/quantify latecy on each component of the network? (phone1 -> router -> asterisk -> router -> phone2)

I used several softphones and I noticed a prety big latecy between the end points. The same is with echo test, as I can judge around 200ms (human appreciation).

Several tests with different softphones concluded to the same results (X-Lite, Linphone, linphonec).

Because is not the router (average 5-6ms) and RTP is enabled in asterisk, I start to believe the problem is with the sound conversion in these phones (computers). So this is the reason of my question, the need to determine with certitude where the delay occurs and to find a fix.

Can anyone post some of personal experience?
The problem I have with the hardphones is that they don’t meet my needs for the application (or at least I’m not aware of). I need them to work in paging and intercom modes, “push to talk”, autoanswer and eventualy wireless. This was the reason to focus on softphones so I would be able to modify the code to fit my needs.

Or maybe I’m missing something else…

Thanks.


#2

Try to run ethereal on your Asterisk Server if he’s in the middle of the path. In the Statistic menu, under RTP you will be able to get the jitter at every packet.

There is also some statistic for the SIP protocol but this as nothing to do with call quality.