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

[Rich Adamson]

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
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.

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> 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
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