Hi, I got similar problems. For me solution was to enable jitter buffer, something like this: .... <action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="set" data="jitterbuffer_msec=60:200:20"/> <action application="export" data="nolocal:jitterbuffer_msec=60:200:20"/> <action application="export" data="nolocal:rtp_jitter_buffer_during_bridge=true"/> <action application="bridge" data="some_bridge_data" /> .... In this case I didn't get delay when I set "<param name="rtp-timer-name" value="soft"/>" in sip profile. Jurijs On Mon, Jan 18, 2021 at 5:10 PM Jacob Greene <jacobgreene1991@gmail.com> wrote:
Hello everyone,
Recently, I am running into an issue with some severe audio delay with some freeswitch boxes running on 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). Making this change immediately fixes the issue.
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 disabling the rtp-timer.
Has anyone had a similar problem? Or does anyone have some more information/resources regarding 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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