I have been kind of tracking IAX2 calls and trying to measure performance with a given "iax2 set jitter" command. My default is 250ms..
When a call is in progress I'll be watching it at the console with "iax2 show channels" Here are my stats from one particular call: 66.225.202.72 benshaw 00001/16413 00048/00035 00489ms 0221ms ILBC 66.225.202.72 benshaw 00001/16413 00137/00125 00487ms 0270ms ILBC 66.225.202.72 benshaw 00001/16413 00137/00125 00491ms 0269ms ILBC 66.225.202.72 benshaw 00001/16413 00141/00129 00487ms 0241ms ILBC 66.225.202.72 benshaw 00001/16413 00141/00129 00480ms 0235ms ILBC 66.225.202.72 benshaw 00001/16413 00143/00131 00480ms 0256ms ILBC 66.225.202.72 benshaw 00001/16413 00144/00132 00492ms 0268ms ILBC 66.225.202.72 benshaw 00001/16413 00152/00140 00487ms 0472ms ILBC 66.225.202.72 benshaw 00001/16413 00154/00142 00507ms 0473ms ILBC Now I figured the guy would be coming up to my office shooting but when I asked him how the call was he said "perfect." -- now he knows he's on a VOIP call but he had no idea of the jitter and lag here... So I suppose my question is "huh?" How can I have such poor jitter and yet have this guy (not a techie) claim the call was perfect? Neither he nor the guy on the other end (PSTN through NuFone) had any issues about the quality. I don't want to look a gift horse in the mouth, so to speak, but I would like to know how to measure call quality; I thought jitter was a pretty good indicator. Regards, Andrew _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
