IAX2 Trunking Maximum Bandwidth Saving

All Viewers I am not so much familiar with Asterisk, But I have been trying to Reduce Bandwidth uses by IAX2 Trunking the result is great but not that much how much I accepted, now I am just sending Call through IAX2 Trunking something like this SIP>>>IAX2>>>IAX2>>>SIP then GSM Gateway but like this if we use g729 or G723 codec for each call its consuming 11.5-16Kbps but we need it to be under 8kbps with g729 codec or maximum 8kbps & I know its possible, but how I really don’t know, I Post here because I know there is so many Professional people who knows very wellfully how can I do it & they can help me, I need help please help me if anyone can, I just want to Save Bandwidth uses as much as I can by any technology, hope I will get some good responds soon from you guys, Thanks for your attention. Have a Nice Day

With Best Regards

The minimum rate for G.729, before overhead is 8kHz, so your requirement is not possible. The figures you quote suggest you have already enabled the options needed to minimise the overhead.

Thanks a lot David, but how much I can reduce maximum with G729 or G723 codec also please tell me after how much bandwidth I needed for run 32 Channels by IAX Trunking. I am waiting for your feedback, Thank You.


According to voip-info.org/wiki/view/Aste … width+iax2 you need 327.5kbs.

However, this doesn’t allow for the layer 1 and 2 overheads and it doesn’t allow for signalling traffic and things like DNS lookups.

You have probably done all that you can to minimise the overheads.

David, Thanks ones again for your fast replay, as I told you that I don’t have so much idea about asterisk, So can you help me to reduce the IP Overhead of asterisk? I need to reduce more bandwidth by iax trunking, I want to run 32 Channel maximum with 256Kbps if possible to make more less so it would be great for me, will you can help me out from this problem? We can disuse via sky Pe if you want I leave my Id Here asterdeve, I am waiting for your feedback, Thank You.


You can do your own calculations from the URL I gave you. There is only one codec listed that stands any chance of working, when you add L1 and L2 overheads. Even then it might still not be enough. Your customers are probably not going like it. It was invented many decades ago for military encrypted links and the speech quality is likely to consistent with what only the military would have tolerated in the 1970s.

Your 256k may include L1 overheads, but is unlikely to include L2 overheads. The values of those will depend on the technology used, and I probably wouldn’t know them, even if you told me what it was.

I believe you have set yourself an impossible task, for VoIP.

hello! i have a problem that looks like this but i am using AsterisNow with Freepbx 2.11 and would like to configure IAX2 trunking between two servers in the same LAN.
I followed this article http://www.tux89.com/telephonie/trunk-iax-entre-deux-systemes-asterisk/ but i am still getting this message: “All lines are busy”.If someone could help me i will be happy :frowning:

The problem on this thread would would cause poor audio quality, and protocol timeouts. It would not produce an immediate congestion error.

Please ask your question as a new thread on the proper forum, Asterisk Support, assuming that you are able to debug and configure this without using FreePBX. (For FreePBX use freepbx.org/forums/ )