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

[Geoff Mina]

What WebRTC client are you using? Does this happen in all browsers or just one?

> On Feb 6, 2018, at 5:09 AM, Colin Morelli <colin.morelli at gmail.com> wrote:
> 
> Appreciate the input, Brian. I’ll definitely try to avoid setting the timer option.
> 
> 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.
>> On Tue, Feb 6, 2018 at 3:49 AM Brian : <brians at iptel.co> wrote:
>> 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 at 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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> _________________________________________________________________________
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> 
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