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

[Joel Maslak]

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