We are a team of 5 people. One is in Croatia, One in Pakistan, One in India, One in Perth, One in Melbroune. Server is in Melbourne.
Asterisk 13.6 is in a virtual machine on a Datacenter in Perth.
The person in Perth has a wireless internet connection with a bit of packet loss (1-4%). The others relatively decent ADSL connections. The one in Croatia is the furthest from the PBX (ping time 400 ms to Perth approx) and can’t normally.
We’ve set a conference room which is something like our “office on the cloud”.
Line often brakes and crackles specially for the one on wireless internet and the one in croatia.
When we talk on skype, the sound quality is much better: no drops, no crackling, etc.
I’ve been looking into asterisk options and I can see that there are
- two channels sip and pjsip,
- several supported protocols: tcp, upd
- rtp can apparentely also go over tcp or udp? not too clear on this one
- a number of codecs
- jitter buffer (seems to be only for chan_sip?)
Can someone recommend what the best asterisk settings would be for best stability and reliability on unreliable / distant internet connections so the quality of sound is more or less like skype?.