Problem to make end2end calls with 2000ms delay


I am trying to test out, how much delay is possible to have with the freePBX system.

When I am using a roundtrip delay of 1000ms, it is possible to call the advertisement number and also to make end2end calls from one user to the other.

In case of 2000ms roundtrip delay I suddenly face problems, the call to the advertisement number works fine, but not the en2end call.

My VOIP client is giving an error, like “connection type not supported”.

Does anybody know if that “timeout” problem is related to settings in PBX or in the Client and whichg settings should be checked/modified ?


VoIP is very sensitive to round trip delay. 2000ms is way too much. I assume this delay is on a WAN link since 2000ms on a LAN is not very likely.

Maybe google is down again today. :wink: I did a search for “VoIP delay” and found a considerable amount of material related to latency.

Here is a document that you light look at

I already found out that it must be related to the FreePBX VOIP server, with the TekSIP server call setup time between end2end user takes max 3 seconds in case of 2000ms roundtrip delay.

With FreePBX it takes more than 10 seconds in case of 1000ms roundtrip delay.

Ok it might be also related to the server HW.

Are there any config parameter that I need to optimize in the FreePBX system ?


I suppose you’re talking about “ping” times…with times such as these, real-time, full-duplex conversation is going to be impossible. If not from the technical standpoint, it will be from the user standpoint. Consider this…with a 2000 ms delay…I make a comment…it’s going to be 2 seconds before the other party hears is, then another 2 before you could hear a reply. Echo cancellation would be a nightmare. Even if you did get a call set up, I don’t think you could carry on a conversationon.

Actually that is not true. Even with a RTT of 2000 ms it is possible to establish a end2end VOIP connection and the voice quality is very good. OK sure you will have that delay of 2 seconds, but no problems with the echo cancellation. At least with the TekSip server. I just still wonder why the FreePBX server is not able to even handle the end2end call setup. Let me know if you know about some secret parameter that could be optimised ?

FreePBX really has no control over this. It’s an Asterisk issue.

I am sure that one of the timeouts is just set too low by default.

FreePBX provides the SIP settings module to adjust the parameters.

All of the “general” parameters listed can be tuned for Asterisk from FreePBX SIP settings module.

registerytimeout, registryattempts and rtptimeout would be great places to start.

SIP T1 and T2 timers also.

Thanks, that is what I was looking for, will check that when time.