[95211] in cryptography@c2.net mail archive
Re: How the Greek cellphone network was tapped.
daemon@ATHENA.MIT.EDU (Bill Stewart)
Thu Jul 19 09:37:25 2007
Date: Wed, 18 Jul 2007 15:43:01 -0700
To: Eric Cronin <ecronin@gizmolabs.org>
From: Bill Stewart <bill.stewart@pobox.com>
Cc: cryptography@metzdowd.com
In-Reply-To: <BD585F25-4BE9-4147-83CE-1E6092AE1233@gizmolabs.org>
At 07:37 AM 7/12/2007, Eric Cronin wrote:
> With current CPUs and audio codecs you can get
> decent voice quality over 9600bps.
Yes and no. There are lots of 8kbps codecs, and some 6.5 and 5.3kbps codecs,
all off which give acceptable voice quality if transmission's ok.
(And you can reduce average transmission rates by 40-50% with silence
suppression.)
However, that's the raw codec rate - if you're taking the VOIP packets,
wrapping them in RTP, UDP, and IP headers, and then transmitting them on
a layer 2 protocol with as little overhead as PPP or Frame,
the 8kbps becomes more like 26 kbps (Ethernet and ATM are worse,
and DSL is ATM underneath - I'm not sure what the cellular carriers do for
framing.)
The problem is that the Voice-stream data packets are extremely small -
the same headers don't add much overhead percentage when you're using
1500-byte data packets.
In some environments you can do header compression to save about half the
bandwidth,
but in general you can't. The Asterisk IP PBX has a trunking protocol that
lets
you use one set of RTP/UDP/IP headers to carry multiple streams of voice
packets,
so you can connect two locations together for close to the raw protocol speeds,
but that's not likely to apply to a mobile phone situation.
The other way to avoid the VOIP overhead is to use one of the old
voice-over-data designs that uses point-to-point async or sync connections
without an IP layer (e.g. raw modems.) That lets you send voice for
much closer to the 9600 bps (depending on sync protocol, async stop-bits, etc.)
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