<div dir="ltr">Happens on all browsers.<div><br></div><div>Just want to clarify my previous message, though. I had a call bridged A -> B (A is the WebRTC side, B the PSTN). I recorded both legs of the call individually. On the recording for B, B's audio is clear and smooth. On the recording for A, B's audio has dropped packets that correspond with the logs mentioned on FS. Unless I'm misunderstanding something I believe this should eliminate network/WebRTC/clients as being the issue.</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Feb 6, 2018 at 8:45 AM, Geoff Mina <span dir="ltr"><<a href="mailto:gmina@connectfirst.com" target="_blank">gmina@connectfirst.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="auto">What WebRTC client are you using? Does this happen in all browsers or just one?<div><div class="h5"><br><div><br>On Feb 6, 2018, at 5:09 AM, Colin Morelli <<a href="mailto:colin.morelli@gmail.com" target="_blank">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" target="_blank">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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