Does my IP-Link (WAN-Link) satisfy the requirements to do VoIP ? (kb2385)
The information in this article applies to:
- SwyxWare from version 4
Since version 4.0x of SwyxWare it's possible to connect different sites via the Swyx LinkManager to do phone calls via IP, instead of expensive PSTN calls. Additionally SwyxIt!-users do login to the SwyxServer via low bandwidth VPN connections.
Because the speechquality depends on the quality of this link, Swyx does provide a tool to measure this link regarding applicability for phone calls.
Most important facts regarding voice quality are delay/jitter and packet loss.
These parameters are measured with the linktest.exe tool. The results will be written into an Excel file (*.csv). Swyx does also provide a set of Excel macros to analyze the results of the linktest.exe tool and create some graphical output of the measurements.
Basically linktest.exe must be started at both ends of the link you want to test. After startup the linktest-tool will generate typical voice traffic and does measure at the same time the delay, jitter and packet loss. These values will be written into a *.csv-file
Handling the linktest-tool
The linktest-tool is a console application. Parameters are given at the command line. After the start of the linktest-tool in a Windows console it does check the necessary registry settings to enable QoS. If something is missing, it will insert the settings. It may be necessary to reboot the machine. This will be mentioned by the tool and it is a must to do this! The linktest-tool does run on Windows 2000 and following.
The linktest-tool does accept lots of parameters via the command line, but only one parameter is necessary. With this parameter you will tell the linktest-tool where the second instance of the tool does run (the IP-address of the machine). You have the call linktest this way:
LinkTest -R 192.168.100.10
In this sample the remote machine of the second instance of the linktest-tool is 192.168.100.10. After starting the linktest-tool on both sides of the link it will start to generate IP-traffic for 15 minutes. After these 15 minutes the tool will stop automatically, or you can stop it by pressing simple any key.
If the measurement was successful you will find the saved *.csv-file inside the same directory the linktest-tool was started. The filename is build out of the current date and time.
If you like to use longer or shorter intervalls you can do this by adding the parameter -d minutes. Minutes is the time of the measurements. You will get a complete list of possible parameters on calling the linktest-tool with the parameter -?.
Sometimes it's useful to start the linktest-tool at specific times of a day to get an overview of traffic behaviour during the day. In most cases you want to collect all the measurement into one file. You will do this by using the parameter -f LogFileName. All written values are appended to the Logfile on using the -f parameter.
The measurements inside the *.csv-file can be analyzed with the Excel-macros from the file LQAnalyser.xlt. Please doubleclick the LQAnalyser.xlt file, which will start Excel. Depending on the security settings regarding macros inside Excel you will see a dialog asking for activation of macros. Please acknowledge the use of macros, otherwise the analyzation will fail.
Now you will find a new item in the menu list named Link Quality Analyser. Please click this item and chose Load Data afterwards. Inside the FileOpen-Dialog select the *.csv-file with the stored measurements inside.
On opening the file the macros will create two sheets. Sheet one is named Table of results and does contain on column with packet loss in percent and one column with jitter in percent. On the right side there are all the columns with the measurements of each interval. By default each interval is 5 minutes long. The startup-time of each interval is shown on top of the column.
Column D does devide the roudtriptime of the IP-traffic in intervals of 10 milliseconds each up to a maximum of 490 milliseconds. At the right side you will find the number of packets send/received in each interval. At the left side in column A the sum of all packets inside each interval in mentioned in percent of all packets. The same is done in column B for the jitter (in percent).
|"A" - Packets in %||"B" - Jitter in %||"D" - Intervall||M1||M2||M3|
Column B shows the jitter in percent. VoIP applications do collect received packets inside jitter buffers. The application does wait some time to fill up the jitter buffer to eliminate the different delay of the received packets. This is necessary, as the playback devices will need the packets in a very regular manner with fix intervals between the packets. The jitter is now given in percent for each interval to show how many packets can be collected inside the jitter buffer before playback. Looking at the above samples does show, that 96,84% of all packets can be inserted into the jitterbuffer, if this jitterbuffer has at least a size of 70-80 milliseconds. If the jitter buffers size is increased, you will be able to receive more and more packets, but at the same time the delay will increase, as the packets are collected in the jitter buffer before playback. SwyxIt! does use a jitter buffer of max. 210 ms at this time. If the packets are received on a very regular base, SwyxIt! will decrease the size of the jitter buffer automatically to decrease the delay.
The packet loss should not get bigger than 5% of all audio packets. Also the delay should not increase in a way, that more than 5% are lost, because they arrive to late to put them into the jitter buffer. Loosing more than 5% of packets will have a very negative impact on the voice quality.
The second sheet in Excel is named Charts of results. This will show you a graphical representation of the measurements. The diagram Average Delay shows the average delay of all packets. Please check, that at least 95% of all packets do have a delay of max. 210ms. The diagram Lost packets per session does show the number of lost packets in percent per session of measurement (each session is by default 5 minutes long). Please check, that the lost packets of the sessions does not vary too much, otherwise voice quality will change from time to time.
- Link Test Tool
The third-party contact information included in this article is provided to help you find the technical support you need. This contact information is subject to change without notice. Swyx in no way guarantees the accuracy of this third-party contact information nor is responsible for it's content.