Recommended setup for bad internet lines

Asterisk Asterisk Support
1

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?.

2

This sort of question is extremely broad and difficult to answer.

I have a suggestion or two:

  • Use the PJSIP channel driver as this is where SIP development within Asterisk is currently focused. The chan_sip driver was demoted out of core support in Asterisk 12. Asterisk 11 is still a supported LTS, but 13 is the latest and PJSIP is the only core supported SIP channel technology within 13.
  • Whether to use SIP over TCP or UDP is less an Asterisk question. There is plenty of information to Google on this subject. Here is a stack overflow article.
  • I don’t think Asterisk supports sending RTP/media over TCP - that seems like a bad idea.
  • These are easy to change and test. I recommend simply testing them. You might start with GSM and ULAW/ALAW
  • You can setup a jitterbuffer with PJSIP using the JITTERBUFFER function. If jitter is an issue then I recommend simply trying it out.
3

Thanks, I appreciate your help