AWS audio delay and rtp timer.

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atmosphere617

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May 19, 2018
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Recently, I am running into an issue with some severe audio delay with some of my deployments in AWS. It gets to the point where audio is delayed up to 10 seconds in both directions. The calls start off fine and slowly get worse.

The only way I've been able to fix this is by disabling the rtp timer on the sofia profile(rtp-timer-name=none). No obvious system bottlenecks, 1/5/15 min load averages all less 1 on dual core CPUs, tons of free memory, no I/O issues on the nic, etc.

I suspect this might have something to do with some weird CPU sharing AWS is doing on their hypervisor, but I'm not sure.

I can't find a ton of info on exactly what the rtp timer does, other than disabling it "disables asynchronous rtp" and "makes freeswitch handle media the same way as asterisks". I'm not really sure what this means. I've read it's less effecient and come across a couple very old post on the mailing list of Anthony steering people away from using it.

Has anyone had a similar problem? Or does anyone have some more information on exactly what the function of the rtp timer is? I'm trying to get in front of this. What are the drawbacks of not using an rtp timer?

Thanks for reading!
 
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lodperera

New Member
Oct 24, 2019
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Any update on this mate. good find.

Are you having issue after rtp-timer-name=none ?

I have same issue where the first few seconds of the call is delayed or not heard.
 
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lodperera

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Oct 24, 2019
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gflow said:
How are you testing this?
Click to expand...
What do you mean mate?

I havent applied this setting yet.

My issue is intermittent.
Customer provides call samples. That's how i know.

First few seconds silent and rtp start after few seconds. But not delayed. Missing audio like a buffer flush.
 
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gflow

Active Member
Aug 25, 2019
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I mean how are you testing for the missing audio but it sounds like you have the actual packet captures.

I've had clients tell me audio is breaking up or missing the first few seconds but they are testing by calling their mobile in one ear and the deskphone in the other ear and because of noise & echo cancellation on handsets the audio will sound bad having the phones so close.

Another thought as well is if the client uses a headset some of the better headsets like the Jabra Evolve2 series have auto pause, so when you answer a call it'll be muted for a little bit and then unmute. I use that headset myself and when I first got it (or any client first gets it) they will say "hello" but the person on the other end won't hear that first hello because of this headset feature. They either get used to or or switch to a more basic headset.

BTW - I've been running a system on AWS for years and never had this reported or experienced myself.
 
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fusionpbxnoaudio

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Jul 15, 2021
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lodperera said:
Any update on this mate. good find.

Are you having issue after rtp-timer-name=none ?

I have same issue where the first few seconds of the call is delayed or not heard.
Click to expand...
Audio not heard on the first few seconds - what is the client that is being used?
 
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