[Asterisk-Users] Asterisk SIP Packet Time (20ms)

[Chris Albertson]

The reason you use UDP over TCP for realtime meadia is that
TCP's ability to reliably deliver every packet in order actually
sounds worse.  Reason being is that with a UDP system a dropped
packet sounds like just a dropout but if you used TCP the audio
stream would be held up and delayed in a queue while that lost
packet was being retransmitted.  In stead of a dropout the audio
would sound as if someone kept hitts a "pause" button on a tape
recorder.  A dropout sounds better then a delay of potentialy 
several seconds

Almost all realtime meadia systems (telephony, video, possition
reporting and so on) maintain some kind of a buffer on the recieving
end.  But you trad the buffer lenght for delay.  Using UDP allows
the application to do the buffering where as TCP putting this buffing
functin in the operaing systems network code.



--- Andres <andres at telesip.net> wrote:
> On Tuesday 23 December 2003 11:40, Rich Adamson wrote:
> > There's no reassembly with udp, and there is no sense of packets
> arriving
> > in the same order as what was sent. Udp is a best-effort
> low-overhead way
> Right, UDP itself does not care about order, but at the application
> layer you 
> can keep track of it.  You can design your application to buffer X
> packets 
> and then reorder them according to sequence numbers.
> 
> > of transmitting data (with UDP often times referred to as the
> Unreliable
> > Data Protocol). Changing to TCP would allow reassembly, however the
> > overhead would be substantial.
> >
> > ------------------------
> >
> > > The problem occurs when the software is expecting the packet in a
> certain
> > > timeframe so that it can reassemble it in a timely manner.  It's
> not a
> > > big deal with a web page or something along that lines.  But when
> a voice
> > > application cannot get reassembled in a timely manner, you'll
> surely
> > > notice it!
> > >
> > > -----Original Message-----
> > > From: Joel Maslak
> > > To: asterisk-users at lists.digium.com
> > > Sent: 12/23/2003 10:41 AM
> > > Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
> > >
> > > On Tue, 23 Dec 2003, Rich Adamson wrote:
> > > > If a collision or dropped packet occurs (in a voip udp
> environment)
> > >
> > > there
> > >
> > > > is no way to retransmit the missing/damaged packet. Missing one
> packet
> > >
> > > isn't
> > >
> > > > a big deal, but if you have collisions and/or dropped packets,
> there
> > >
> > > is a
> > >
> > > > very high probability that lots of packets will be dropped. If
> too
> > >
> > > many
> > >
> > > > are dropped, you'll hear the result in the undecoded voice as
> choppy
> > > > voice.
> > >
> > > Actually, collisions occur at Layer 2, not Layer 3, and the layer
> 2
> > > hardware automatically resends packets involved in a collision -
> layer 3
> > > is never aware of it happening (although it may cause additional
> delay).
> > > Eventually the ethernet card will give up if too many collisions
> occur
> > > during retries, but this is very rare in practice unless the
> network is
> > > *VERY* loaded.
> > >
> > > > Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex
> 10 meg
> > > > ethernet would handle roughly 20-25 rtp sessions before bumping
> into
> > >
> > > the
> > >
> > > > problem (your milage may vary). The majority of the folks on
> this list
> > > > seem to be running home/soho systems and would likely never run
> into
> > >
> > > the
> > >
> > > > issue. But the heavier users will.
> > >
> > > For a duplex mismatch, my experience is that if one end on a 100
> Mb/sec
> > > link is half and the other is full, bandwidth is limited to about
> 8
> > > Mb/sec
> > > max.  This is based on some tests I've accidentally conducted. 
> If you
> > > try
> > > to send 9 Mb/sec over that link, yes, some packets will get
> dropped as
> > > they simply won't fit.  (But I do agree that for a half-half
> link, you
> > > can
> > > get about 20 Mb/sec)
> > >
> > > --
> > > Joel
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ---------------End of Original Message-----------------
> >
> >
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=====
Chris Albertson
  Home:   310-376-1029  chrisalbertson90278 at yahoo.com
  Cell:   310-990-7550
  Office: 310-336-5189  Christopher.J.Albertson at aero.org
  KG6OMK

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