[Freeswitch-users] rtp-timer-name / timer issues

[Colin Morelli]

Hey list,

I'm running FS on EC2 (I know, I know). Having some issues with random
packet loss, which I believe almost certainly I have narrowed down to timer
issues and/or network latency/jitter (seems surprising since I'm using
c5.xlarge instances).

Behavior is that, during a call, brief pauses or notable audio loss will
occur. This is on high bandwidth links that are otherwise stable.
Freeswitch logs with max debug spew out "Hot Hit 1" through "Hot Hit 10"
and eventually "auto-flush catching up 1 packet(s)" in rapid succession
(usually going through the cycle 4-5 times) before things settle again.
Obviously that means a minimum of 4-5 audio packets were dropped within the
span of a second which results in considerable audio artifacting.

Changing rtp-timer-name to none, which I understand to perform synchronous
reads of RTP audio (as opposed to timer-based async reads) makes the audio
notably smoother. That said, I'm having a hard time uncovering the
consequences of doing this. Obviously I understand that reads will block
the RTP thread, but I can't seem to understand the potential ramifications
of this. Could anyone help clarify?

My other question is: assuming "timer while hot" indicates what I believe
it does (that when the timer hit there was >1 packet in the queue to be
read), couldn't this issue also just be caused by network jitter, and not
necessarily just timer inconsistencies?

Thanks in advance.

Best,
Colin
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