Difference between revisions of "VoIP call quality"
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{{czerwony| If we dont get answer is}} <br><br> | {{czerwony| If we dont get answer is}} <br><br> | ||
| − | {{czerwony| If you do not receive a reply to SIPPING / | + | {{czerwony| If you do not receive a reply to SIPPING / IAXPING / SSLPING means that the VoIP call is not going to happen. This is because routers are not configured correctly to work in VoIP}} <br><br> |
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Pktloss value should be equal to "0" other value means that part of my package did not reach server(PABX). | Pktloss value should be equal to "0" other value means that part of my package did not reach server(PABX). | ||
<br> | <br> | ||
| + | '''Example''' | ||
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| + | [[File:Wynik_qos.gif]] | ||
| + | '''These results provide very good call quality''' | ||
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| + | [[File:Log_Qos_-_bad_Quality.gif]] | ||
| + | '''These results provide very poor call quality''' | ||
Latest revision as of 10:20, 8 September 2014
Contents
QoS Internet connection
The level of quality of access consists of the following factors:
Throughput
When used in the context of communication networks, such as Ethernet or packet radio, throughput or network throughput is the rate of successful message delivery over a communication channel. The data these messages belong to may be delivered over a physical or logical link, or it can pass through a certain network node. Throughput is usually measured in bits per second (bit/s or bps), and sometimes in data packets per second or data packets per time slot. Each connection VoIP bandwidth is symmetrical, ie both ways (transmission / reception). Using this test, we can also estimate how many simultaneous connections we can perform.
Delay
The delay of a network specifies how long it takes for a bit of data to travel across the network from one node or endpoint to another. It is typically measured in multiples or fractions of seconds. Delay may differ slightly, depending on the location of the specific pair of communicating nodes. Delay is measured using the "ping" command
ping <destination address>
Ping is a computer network administration utility used to test the reachability of a host on an Internet Protocol (IP) network and to measure the round-trip time for messages sent from the originating host to a destination computer. The correct response to this command does not guarantee proper operation of VoIP.
If you do not receive a ping response does not mean that the VoIP call is not going to happen. This is because some routers, for safety reasons, they have disabled the response to this command.
sipping/iaxping/sslping <destination address>
This is proper commend using in SIP traffic to measure packet time "end to end". This command allows diagnosis of two important parameters:
- If get response is the correct configuration of all devices on the road (mainly routers).
- Time to respond packets it is not to be "greater than 150ms"
If we dont get answer is
If you do not receive a reply to SIPPING / IAXPING / SSLPING means that the VoIP call is not going to happen. This is because routers are not configured correctly to work in VoIP
Jitter
Illustrates the difference in delays between packets, and counts the lost packets.The value of 'jitter' for the next packet should have the same value. Larger deviations of the values mean lower quality connection. Pktloss value should be equal to 0 other value means that part of my package did not reach server(PABX).
Packet loss
Pktloss value should be equal to "0" other value means that part of my package did not reach server(PABX).
Example
These results provide very good call quality
These results provide very poor call quality
