I have two asterisk systems connected via IAX2 trunks. I am using the second system (FreePBX13 and Asterisk13-22, all modules current) to host the conference bridge. When the bridge gets over 23 people the audio starts to get really choppy. The Memory and Computer usage both are quite low so I don’t think the system is being overworked. Am I missing something in my IAX2 trunk setup?
Bandwidth issues?
I would try SIP instead of IAX and see if the same thing happens there.
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That is a good idea. The two systems are local to each other if that changes things at all.
I found this in a wireshark capture, not sure if it is helpful
.11 is the first asterisk server and .15 is the FreePBX server that is housing the conference bridge
asterisk -rx 'core show uptime'
really stupid question - if I copy and run this command from root it is not recognized. please don’t shoot me
asterisk -rx 'core show uptime'
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So I switched to SIP and the conference calls seem to be great over 20 people. What do you think accounts for the difference between an IAX2 connection and a SIP connection? Just wondering. Thanks again for your advice
I was using IAX trunks between PBXs over the internet and people were complaining about terrible call quality and one day out of the blue it stopped working one way. Then I switched to SIP and the issues went away.
You will find many who like IAX and will recommend it, but I am no longer using it for above experiences. IAX is also end of life and hasn’t been touched or patched for years.
Very interesting. I’ve always defaulted to using Iax2 connections between my FreePBX servers. If I choose SIP connections is the user and peer details the same as if I were using an iax2 trunk or do they become altered in a SIP trunk?
Make sure you you have
trunk=yes
In the trunk definition,
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