[Freetel-codec2] codec2 for VOIP

[David R. <da...@ro...>]

Hello List,

There may be some important applications for Codec2 over VOIP.  

The issue with low bit rate codecs over VOIP is the IP/RTP protocol
overhead.  For example this overhead is 16 kbit/s for 20ms frames, and
even worse for g729 which has 10ms frames.  This means a low rate speech
codec has only a small impact on the overall bit rate.

However in many (perhaps most) VOIP calls, multiple VOIP channels are
flowing through one pipe for at least part of their journey.  This pipe
might be relatively narrow. For example a small office IP-PBX connected
to an ITSP via a commodity DSL line.  Or long distance links between
nodes in an ITSP network with 100's of calls.

QoS (sharing data and voice) also gets a lot easier when the codec data
rate is reduced.

There are people with good businesses based on reducing VOIP bandwidth
by RTP aggregation.  Vibe http://www.voip-x.co.uk/vibe/helps.shtml have
a popular (closed) solution that creates an interface on your Linux box
(like a VPN network interface).  The interface allows you to send n g729
RTP streams to a similar interface at roughly n x 8 kbit/s, e.g. each
additional after the first one call requires only 8 kbit/s.  It's like a
closed source, SIP based version of IAX2.  I guess they have added some
value, support etc, that makes it a good solution for their customers.

So reducing the bandwidth of multiple VOIP calls is a more important
problem that I had first assumed.  Making additional calls n x 2 kbit/s
or even 4 kbit/s would be a significant improvement.

There are also interesting possibilities for the developing world. I
have friends in developing world countries with 64 or 128 kbit/s DSL
lines.  Codec 2 could give you 32 calls in one 64 kbit/s a-law channel.
This could be make or break for a developing world ITSP start-up
business model.

Cheers,

David