Difference between revisions of "Latency"
From Kolmisoft Wiki
Jump to navigationJump to search
| (3 intermediate revisions by 2 users not shown) | |||
| Line 1: | Line 1: | ||
The time it takes a conversation to travel from the speaker to the listener is the end-to-end delay, or latency. | The time it takes a conversation to travel from the speaker to the listener is the end-to-end delay, or latency. | ||
<br><br><br> | |||
End-to-end delay is actually made up of four components: | End-to-end delay is actually made up of four components: | ||
| Line 8: | Line 8: | ||
* Jitter buffer delay— The delay introduced by the receiver as it holds one or more datagrams to reduce variations in arrival times. | * Jitter buffer delay— The delay introduced by the receiver as it holds one or more datagrams to reduce variations in arrival times. | ||
<br> | |||
Many VoIP engineers don't know how much latency is too much. A simple answer is 150 ms. | Many VoIP engineers don't know how much latency is too much. A simple answer is 150 ms. | ||
Users begin noticing latency as a service level issue when roundtrip latency is greater than 250 milliseconds (ms). | |||
The International Telecommunications Union recommends that latency never exceed 150 ms one way (from speaker to listener). | |||
<br><br> | |||
It can be checked using ping or iperf commands. | |||
Example with ping:<br> | |||
[root@ns36850 ~]# ping 109.235.71.xxx | |||
PING 109.235.71.xxx(109.235.71.xxx) 56(84) bytes of data. | |||
64 bytes from 109.235.71.xxx: icmp_seq=1 ttl=56 time=40.6 ms | |||
64 bytes from 109.235.71.xxx: icmp_seq=2 ttl=56 time=40.6 ms | |||
64 bytes from 109.235.71.xxx: icmp_seq=3 ttl=56 time=40.7 ms | |||
<br><br> | <br><br> | ||
Source: [http://www.amazon.com/Taking-Charge-Project-Network-Business/dp/1587200929 Taking Charge of Your VoIP Project] | Source: [http://www.amazon.com/Taking-Charge-Project-Network-Business/dp/1587200929 Taking Charge of Your VoIP Project] | ||
Latest revision as of 13:54, 29 October 2011
The time it takes a conversation to travel from the speaker to the listener is the end-to-end delay, or latency.
End-to-end delay is actually made up of four components:
- Propagation delay— The time to travel across the network from end to end. It is based on the speed of light and the distance the signal must travel.
- Transport delay— The time to get through the network devices along the path. Networks with many firewalls, many routers, congestion, or slow WANs introduce more delay than an overprovisioned LAN on one floor of a building.
- Packetization delay— The time for the codec to digitize the analog signal and build frames—and undo it at the other end.
- Jitter buffer delay— The delay introduced by the receiver as it holds one or more datagrams to reduce variations in arrival times.
Many VoIP engineers don't know how much latency is too much. A simple answer is 150 ms.
Users begin noticing latency as a service level issue when roundtrip latency is greater than 250 milliseconds (ms).
The International Telecommunications Union recommends that latency never exceed 150 ms one way (from speaker to listener).
It can be checked using ping or iperf commands.
Example with ping:
[root@ns36850 ~]# ping 109.235.71.xxx PING 109.235.71.xxx(109.235.71.xxx) 56(84) bytes of data. 64 bytes from 109.235.71.xxx: icmp_seq=1 ttl=56 time=40.6 ms 64 bytes from 109.235.71.xxx: icmp_seq=2 ttl=56 time=40.6 ms 64 bytes from 109.235.71.xxx: icmp_seq=3 ttl=56 time=40.7 ms
