
<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto">What WebRTC client are you using? Does this happen in all browsers or just one?<br><div><br>On Feb 6, 2018, at 5:09 AM, Colin Morelli <<a href="mailto:colin.morelli@gmail.com">colin.morelli@gmail.com</a>> wrote:<br><br></div><blockquote type="cite"><div>Appreciate the input, Brian. I’ll definitely try to avoid setting the timer option.<br><br>In other news. I deployed The exact same FS instance (same docker container) on baremetal last night and it experiences the same issue. So, virtualization does not appear to be the problem. I just can’t figure out what else would cause this. I’m sure it’s something simple.<br><div class="gmail_quote"><div dir="ltr">On Tue, Feb 6, 2018 at 3:49 AM Brian : <<a href="mailto:brians@iptel.co">brians@iptel.co</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Colin,<br>

<br>

Depending on what you are doing with Freeswitch setting the rtp-timer<br>

to none can produce all sorts of subtle weirdness. I would advise<br>

against it. 2 things that I remember from our tests with this was lots<br>

of blocked / hung calls that would build and need to be HUPed and also<br>

carriers that would send SDP but no RTP when silence was being sent -<br>

the call wouldn't progress through dialplan - it just got blocked<br>

waiting on RTP><br>

<br>

B<br>

<br>

On Tue, Feb 6, 2018 at 12:56 AM, Colin Morelli <<a href="mailto:colin.morelli@gmail.com" target="_blank">colin.morelli@gmail.com</a>> wrote:<br>

> Hey list,<br>

><br>

> I'm running FS on EC2 (I know, I know). Having some issues with random<br>

> packet loss, which I believe almost certainly I have narrowed down to timer<br>

> issues and/or network latency/jitter (seems surprising since I'm using<br>

> c5.xlarge instances).<br>

><br>

> Behavior is that, during a call, brief pauses or notable audio loss will<br>

> occur. This is on high bandwidth links that are otherwise stable. Freeswitch<br>

> logs with max debug spew out "Hot Hit 1" through "Hot Hit 10" and eventually<br>

> "auto-flush catching up 1 packet(s)" in rapid succession (usually going<br>

> through the cycle 4-5 times) before things settle again. Obviously that<br>

> means a minimum of 4-5 audio packets were dropped within the span of a<br>

> second which results in considerable audio artifacting.<br>

><br>

> Changing rtp-timer-name to none, which I understand to perform synchronous<br>

> reads of RTP audio (as opposed to timer-based async reads) makes the audio<br>

> notably smoother. That said, I'm having a hard time uncovering the<br>

> consequences of doing this. Obviously I understand that reads will block the<br>

> RTP thread, but I can't seem to understand the potential ramifications of<br>

> this. Could anyone help clarify?<br>

><br>

> My other question is: assuming "timer while hot" indicates what I believe it<br>

> does (that when the timer hit there was >1 packet in the queue to be read),<br>

> couldn't this issue also just be caused by network jitter, and not<br>

> necessarily just timer inconsistencies?<br>

><br>

> Thanks in advance.<br>

><br>

> Best,<br>

> Colin<br>

><br>

><br>

><br>

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