Howto:Network VoIP Readiness Test

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Contents

Applies To

This information applies to

  • all innovaphone VoIP Gateways and Phones


More Information

Problem Details

The VoIP as application on a network requires a stable connecton able to transport big number of UDP packets between two VoIP endpoints in both directions with minimal delay, jitter and packet loss.

Especially on WAN connections, a lack of available bandwidth, jitter or packet loss can affect voice quality on VoIP calls.

To test the connection between two VoIP endpoints with WAN connection between, it is not enough to send a few small ICMP ping packets. Furthermore a bidirectional stream of UDP packets with defined payload and interval must be sent over longer period of time.


Iperf

For this purpose can be used an open source tool Iperf.

To test the connection between to sites put a PC with Iperf on each. Iperf works in Server-Client mode, so on one host the Iperf must be started in the server mode, on the second PC in the client mode.

Download Iperf binaries for your operating system an make sure you can start Iperf from the command line:

C:\temp>iperf -v
iperf version 2.0.5 (08 Jul 2010) pthreads

Now we will test a connection between two sites by simulating one G711 call. To test more calls add the parameter --parallel 4 to server and client command lines.

On the server host execute following command:

iperf --server --udp --len 300 --tos 184 -fk --interval 5

On the client host use following command:

iperf -c SERVER_IP_OR_HOSTNAME --udp --len 300 --bandwidth 67000 --dualtest --tradeoff --tos 184 -fk --interval 5 --time 60 --listenport 5002

With this prameters iperf will generate a bi-directional stream of UDP packets for 60 seconds and consume bandwidth of 67 kBit/s (RTP stream with G711 30ms frame size payload, for other codecs bandwidth usage please refer to www.bandcalc.com)

The resulting statistics should look like this:

[ ID] Interval    Transfer    Bandwidth      Jitter   Lost/Total Datagrams
[ 3] 0.0- 5.0 sec 39.6 KBytes 64.8 Kbits/sec 0.205 ms 1/ 136 (0.74%)
[ 3] 5.0-10.0 sec 39.8 KBytes 65.3 Kbits/sec 0.217 ms 0/ 136 (0%)
[ 3] 10.0-15.0 sec 39.6 KBytes 64.8 Kbits/sec 0.222 ms 0/ 135 (0%)
[ 3] 15.0-20.0 sec 39.6 KBytes 64.8 Kbits/sec 0.240 ms 0/ 135 (0%)
[ 3] 20.0-25.0 sec 39.8 KBytes 65.3 Kbits/sec 0.348 ms 0/ 136 (0%)
[ 3] 25.0-30.0 sec 39.6 KBytes 64.8 Kbits/sec 0.261 ms 0/ 135 (0%)
[ 3] 30.0-35.0 sec 39.8 KBytes 65.3 Kbits/sec 0.197 ms 0/ 136 (0%)
[ 3] 35.0-40.0 sec 39.6 KBytes 64.8 Kbits/sec 0.218 ms 0/ 135 (0%)
[ 3] 40.0-45.0 sec 39.6 KBytes 64.8 Kbits/sec 0.394 ms 0/ 135 (0%)
[ 3] 45.0-50.0 sec 39.8 KBytes 65.3 Kbits/sec 0.231 ms 0/ 136 (0%)
[ 3] 50.0-55.0 sec 39.6 KBytes 64.8 Kbits/sec 0.320 ms 0/ 135 (0%)
[ 3] 0.0-60.0 sec 476 KBytes 65.0 Kbits/sec 0.250 ms 1/ 1626 (0.062%)

For good voice quality there should be no packet loss and jitter should not higher than few milliseconds.

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