[arm-allstar] Asterisk internal latency

[Patrick Perdue]

Very cool, can't wait to see what that's all about.

For my closed-loop radio-less node with headphones (I'm too deaf to 
properly use speakers anymore), I have a script that just unmutes the 
microphone playback when I key up so I can monitor my local audio, then 
it mutes again when I drop the key. OF course, this is real time, since 
the audio from the microphone is not being processed by Asterisk in any 
way, but doesn't allow for anything like squelch tail compensation. I 
don't need that for my node, but that bit of delay would be good on a 
repeater.


On 4/28/2022 10:47 PM, "David McGough via ARM-allstar" wrote:
> Steve,
>
> I run Asterisk 18.x on all my phone PBX systems.  To my ear, the realistic
> latency through Asterisk 18 is about the same as the ancient 1.4.x
> version, when using IAX2, SIP and transcoding....All these components
> being required elements.  This being said, I've not actually measured the
> Asterisk 18 latency using scientific tools (e.g.: an oscilloscope).
>
>
> 73, David K4FXC
>
>
>
> On Thu, 28 Apr 2022, "Steve L via ARM-allstar" wrote:
>
>> I'd venture to say a 1.40 thing.  There were very noticeable timing
>> improvements in subsequent versions
>>
>> https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces
>>
>> I remember when I was using Asterisk as a PBX to hairpin my pots line, and
>> the echo induced by timing latencies was much improved when I upgraded to
>> 1.6
>>
>> On Thu, Apr 28, 2022, 7:52 PM "Patrick Perdue via ARM-allstar" <
>> arm-allstar at hamvoip.org> wrote:
>>
>>> If using Asterisk in full duplex mode, I.E. duplex=2, there is obviously
>>> an inherent delay. What I want to know, however, is why that delay is as
>>> high as it is. It seems to have nothing, or very little, to do with the
>>> OSS sound driver, as this delay is about the same with a local VoIP
>>> device as any of the attached radios using the SimpleUSB channel driver.
>>> It just strikes me as odd that I can use a program like Jamulus or
>>> Sonobus, and get a high quality, uncompressed stereo PCM or OPUS stream
>>> to and from myself across the Atlantic via UDP in quite a lot less time
>>> than I can hear myself coming from my own repeater with all the delays
>>> from the SimpleUSB channel driver at zero, when network latency isn't
>>> even a factor. Yes, I realize that I'm talking about pushing more data
>>> across in these cases, and these programs are specifically designed for
>>> near zero latency due to their very nature, but it's just interesting to
>>> me.
>>>
>>> Is this just the inherent behavior of the old Asterisk 1.4.x core, or is
>>> there something else going on here?
>>>
>>> A curious HamVoIP user:
>>>
>>> N2DYI
>>>
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