Re: streaming protocol, satellite links

Sean Adams <[email protected]> Fri, 5 Oct 2001 23:44:09 +0000 (GMT)
Newsgroups gmane.music.equipment.slimp3.dev
Message-ID <[email protected]>
Mike,

No, we can - in fact, for shoutcast proxying we're effectively doing this
already. We don't keep the buffer in the application though; the TCP stack
takes care of that for us.

I was referring to the case where there is a long link between the player
and the server (not talking about proxies here - I missed the point of
your original question).

If the SliMP3 protocol is only between the player and the SliMP3 server,
then the shoutcast server can be far away - that's no problem. It's just
the SliMP3 protocol that's currently latency-constrained.

Sean

On Sat, 6 Oct 2001, Mike O'Connor wrote:

> Hi Sean
>
> So what your saying is that you can not buffer the sound at the server ?
> This would make it so you did not have to change anything on the slimp3 side
> of the system
>
>
> Mike
>
> > Use a new protocol (best, I think)
> >
> > I think the best protocol is one where the client acknowledges every
> > packet individually, and it's the server's responsibility to keep track of
> > the client's buffer usage, which packets need to be retransmitted,
> > timeouts, etc.  This would simplify the client side quite a bit, at the
> > expense of having to implement a fairly complex TCP-like protocol on the
> > server side. Ultimately this looks like the best solution because it gives
> > full control of the protocol to the application. We'll have to deal with
> > the same issues as other UDP streaming protocols - in particular,
> > detecting link capacity, being fair to TCP connections sharing the same
> > path, and refusing to stream or reducing the data rate (reencoding the
> > MPEG!!) if we're causing too much packet loss.
> >
> >
> > Sean
> >
> >
> >
> >
> > To unsubscribe from this group, send an email to:
> > [email protected]
> >
> >
> >
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> >
>
>
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