> Alex is correct. Always check thereare no half-duplex links in your > path. If you have an older dsl/cable modem or router that only has a > 10M ethernet, it is probably half. Also make certain there are no hubs > in the path. Keep in mind that colissions ar NORMAl for a hlaf duplex > connection. TCP traffic simply retransmits, but voice (on asterisk) is > RTP/UDP and the packet gets dropped. Even if it were TCP there is no > time for a retransmit to be detected and resent. Using ehternet to > detect the collision it does get resent, but there comes your jitter - > which has much worse effects than simply latency. > > As far as measuring latency, doing a sip show peer andlooking at the > qualify times is a GUIDELINE. It is my no means a correct indication, > the real time can be much lower. I have noticed various ATA on the > same networks as Polycom phones wil have sub 20ms times and the > Polycoms will be <50ms. Yet all is as it should be and working great. > > Generally QOS will help with packet loss and jitter. > > Hope this helps.
You were both right I was just double checking. I fired up a soft phone on a desktop that has relatively low ping rates and experienced similar response times ns1*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.29 5060 Unmonitored vitel-inbound/rsreese 64.2.142.116 5060 Unmonitored 102/102 68.156.63.118 D N 56558 OK (145 ms) 101/101 68.156.63.118 D N 1038 OK (135 ms) Thank you both for your insight. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
