[asterisk-users] Latency woes, qos the fix?

[Stephen Reese]

> 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.