Im using asterisk v13.25. I’m connected over CSIPSimple to server. I’m using SIP.
Sometimes I’m on mobile internet where the ping can be over 150ms constantly what realy degrade the call.
From another side I’m in another country where the server is, and it happend that the route between me and server is with higher ping.
Is there any trick with config how to do this?
Can I do anything with parameter jitter?
Thx for help.
What actually matters is jitter, not round trip time. (Obviously large round trip times are not good, but there is no way you can travel back in time to reduce them.)
Mobile internet, in particular, is not designed for real time speech. That is why they still provide a voice service, which is. The best thing is to use the service intended for voice.
Yes I know, but sometimes there is no alternative, that why I’m asking for help.
How to change jitter value?
What is the optimal in my case, and now to calculate?
Use RTCP debugging to find the actual values.
However, note that jitter buffering VoIP in an intermediate switch is not normal, and I don’t know how well it is supported. Normally the phones should be doing the the jitter buffering. That also means that, even if it works, there will be little experience as how to best configure it.
Remember that typical jitter buffer sizes for streaming video are several seconds, which would be totally unacceptable for real time voice, but what is needed to get reliable audio over the sort of channel you are talking about.
Thx…I will try and announce the result.