[Icecast] Typically end-to-end 'delay' of live audio

[Tony]

Thanks.  Here's bit more detail.

We have a scientific audio instrument that comes with its own audio
driver.  This driver gives us Opus 'frames'.  From a remote location
(typically less than 200mS round trip) we would like someone using a
browser (without a plugin) to 'hear' this device.  We need the low
end-to-end delay as there is an active Video session (separate application)
connecting the same two locations.  The remote participant has the ability
to mute either feed (video or scientific device).  However, if the device's
feed is too far 'delayed' from the video feed, it becomes rather annoying
for the remote participant even with the audio portion of the video feed
muted.

Today, we have a browser plugin.  And we transmit the device's audio data
to the remote participant via websockets, wherein the browser feeds the
data to the plugin for decoding and playback.

Unfortunately, browser plugin's are dying technology.  In fact, chrome is
phasing them out later this year.  And Firefox claims they will do the same
"soon".  So, we need to find a replacement.


On Thu, Feb 5, 2015 at 11:03 AM, "Thomas B. Rücker" <thomas at ruecker.fi>
wrote:

> On 02/05/2015 02:45 PM, Tony wrote:
> > I have an audio device driver for a live feed that produces Opus
> > frames, if I were to use icecast, what sort of real-time delay can I
> > expect? 3-5s? 5-10s? more?
>
> Likely that or more. This highly depends on the client side buffers.
> Icecast itself doesn't buffer much and that can be disabled by turning
> off the burst-on connect functionality.
> HTTP streaming is not "real-time" for most values of real time, it's
> still more "real time" than HLS thought, to my surprise.
>
>
> > I've tried using the html5 <audio> tag directly to stream my source,
> > but it seems browsers like to queue-up 250K to 500K of audio 'data'
> > before they begin playback.  That introduces a 15-30s delay depending
> > on browser and the audio compression used.  I'd like an html5
> > (plugin-less) solution that introduces no more than a 3-5s delay.
>
> Yes, you can't control client side buffers for HTTP based streaming to
> browsers.
>
> If you aim for something reliably <<5s without having total control of
> the client side, then your best bet is to look at something else. Maybe
> WebRTC?
>
> I'd probably be able to point out something more helpful, but you don't
> state your actual use case.
>
> Generic standard comment:
> Most cases where people think they need an Icecast stream to have a
> "low" delay are cases where some slight change in approaching the
> problem would help. E.g. for collaborative streaming a real time / VoIP
> backend that then streams to Icecast. Or in case of a "internet radio" a
> question queue instead of trying to achieve "chat like response" over
> the air, or rely on listeners actually "dialling in" through e.g. VoIP
> for questions.
>
>
> Cheers
>
> Thomas
>
>
>
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-- 
Tony
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