[Freeswitch-users] AWS - Audio Delay - rtp timer
jurijs.ivolga at gmail.com
Mon Jan 18 15:27:10 UTC 2021
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="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.
On Mon, Jan 18, 2021 at 5:10 PM Jacob Greene <jacobgreene1991 at gmail.com>
> 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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