Comments
1. Check for Latency (Delay of Transmission of VOIP Packets)
2. Check latency between asterisk and far end carrier is less than 250
mill seconds
3. Check your Transmission and for Jitter Issues.
4. Check for packet loss
5. Draw a network diagram and keep monitoring equipment at every end of
equipment
6. Check the Traffic (Meg) at each point and see if there is sudden change
of Traffic or not.
7. Also send a tone, do a physical loop back at far end and see if you hear the
same tone like it was sent
8. You can also check by sending PCM files across your network and see if that
helps.
Posted by william
Troubleshooting VoIP Call Quality Issues
Assuming you have a Windows PC and a PBX running Linux, there are two useful tools for troubleshooting call quality issues on VoIP -- Wireshark and Ping Plotter Freeware. Sending the results from these programs to your provider will help them diagnose the problem faster, but it doesn't guarantee the issue is something they can fix!
Wireshark captures and analyzes network traffic. It has compatible Linux and Windows versions, which makes it easy to capture the traffic on your Linux PBX, copy it over to Windows and view the results. This capture file can be used to identify almost any SIP problem, so being able to do these steps is very helpful! To do the copy, you will also need to install a program called pscp on your Windows PC.
Run the following to install Wireshark on your PBX (assuming it's running CentOS):
yum install wireshark
Run the following commands on your PBX to capture all traffic, SIP (signalling) and RTP (audio), between the PBX and your provider's server into file /root/my.cap. Based on the options used below, this file can get very large very fast if you have lots of simultaneous calls! Type Ctrl-C after making a phone call to stop the capture.
tshark host server.provider.com -w /root/my.cap
Run the following command to compress your capture file my.cap into a compressed my.cap.gz file:
gzip /root/my.cap
From your Windows PC, run the following to copy that capture file from your PBX's IP address to your Windows PC's C: drive. You will be prompted for the PBX's root password.
pscp root@192.168.1.2:/root/my.cap.gz C:
Then you can open C:my.cap.gz using the Windows version of Wireshark. The Statistics > VoIP Calls menu will let you see all of the calls and let you graph the SIP dialog (to diagnose problems registering or incoming calls that don't ring) or playback the call (to find the one with audio quality problems). The Statistics > RTP menu will let you see all of the RTP streams (audio) and the packet loss, jitter, etc associated with each one. If you see packet loss or high jitter during the calls you experienced call quality issues with, that might explain the problem!
Next, if you want to track down where the packet loss is occurring, you can install Ping Plotter Freeware on your Windows PC and run a continuous ping to a server on your provider's network. You might want to ask them which server to use, because the SIP proxy or website may not be ideal.
The Ping Plotter trace will help you identify if there was packet loss during the times you experienced call quality issues. You can also save the Ping Plotter capture and send that to your provider.
Remember, if the packet loss is at some router on the Internet between you and your provider, you might both be helpless in fixing the situation! Occasionally, your provider might be able to recommend a different server to connect to that is on a network that will bypass that troublesome router. You can also try using a different codec, since some are better at packet loss concealment than others.
I hope this helps you get started in troubleshooting your VoIP call quality issues. Resolving the issue will require a similar effort from your provider and their willingness to examine the useful data you've collected on the problem. The most important thing is to actually capture the problem occurring in a Wireshark capture, because the entire phone call can then be analyzed.
Good luck!
Posted by sagitraz
-