[asterisk-users] Network testing for VoIP

[Justin Sherrill]

I've noticed that if I have people on speakerphone at the two farthest
ends of our internal network, they will occasionally get a second or two
of feedback.  (sounds like jingle bells)  I'm figuring it's some very
slight amount of packet loss or jitter that isn't helped by the
speakerphone echo, magnified into feedback briefly before the phone is
able to shut it off.

Basic reports on ping time or throughput look good, but VoIP traffic is
a lot more finicky than that.  I'm not relishing the idea of checking
each individual port on each switch.

Does anyone have recommendations on a way to test the quality of the
path this voice traffic takes through the network?  

(Asterisk 1.8, Polycom phones)