Hi Colin, Depending on what you are doing with Freeswitch setting the rtp-timer to none can produce all sorts of subtle weirdness. I would advise against it. 2 things that I remember from our tests with this was lots of blocked / hung calls that would build and need to be HUPed and also carriers that would send SDP but no RTP when silence was being sent - the call wouldn't progress through dialplan - it just got blocked waiting on RTP> B On Tue, Feb 6, 2018 at 12:56 AM, Colin Morelli <colin.morelli@gmail.com> wrote:
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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