hi,i have 2 asterisk servers that run at medium capacity ,we have expanded in the last month and it has been horrible, the call quality at times is so degraded to the point it becomes impossible to have a normal conversation.i have researched and found 2 topics that might relate to voip poor quality, jitter and packet loss but i have no idea how to fix them.Please if someone has experienced this before share your experiense.thank you in advancee!
In situations such as this, you need to check a few different things:
1.) RTP/RTCP information. This can help you determine how much packet loss you may be experiencing from the other side’s perspective. Also you might look to see if you are experiencing network jitter during those times (packet reordering/duplication).
2.) Server load information. Using htop, if it looks like you have saturation one or more CPU cores, you will probably have problems related to real time media handling manifesting themselves in the form of quality problem. If you are doing disk I/O intensive activities such as recording and playback, you might check iotop as well to figure out if you are saturating your disk io bandwidth as well.
3.) If none of these things are occurring, you might have network related conditions contributing to the problem. Use a tool such as nload or something similar to monitor how much ingress/egress bandwidth you are using, and see if it makes sense for what you’d suppose your bandwidth limitations are.
4.) If none of these things look suspicious, sometimes it helps to enable jitter buffering and/or PLC support in Asterisk. They can make a difference in load and packet loss and jitter scenarios.
Hope that helps
thank u for the reply i think the problem is with jittter .do u have any guide to how can i enable jitter buffers .thank u
I’m assuming pure SIP, but you didn’t say. Normally, jitter buffering is done in the phone and PSTN gateway, not in intermediate PABXes. If you need extra jitter buffering, you should really be addressing the underlying problem, which lies in your network. The network is overloaded and/or RTP traffic doesn’t have enough priority on the network. Asterisk cannot really address an overloaded network. It can give hints about priority, but the network has to be configured to honour those hints (public networks generally don’t).
if i priorityze voip on the router do u think that it might solve the issue ?
It might do, depending where the choke point is. Note you need to prioritise the RTP. This is not easy for a router to identify, unless you tag it with tos.