Hello Steve,

Over IP you are more likely to get packet errors, i.e. one complete
packet will be missing, or arrive too late to be useful.

Is Codec 2 actually of any use to anyone over IP?  The RTP/UDP/IP
overhead is about 8 kbit/s...... so unless you are trunking multiple
channels in one packet you won't save a huge amount of total bandwidth
over say g729.

- David

On Wed, 2012-02-15 at 07:54 -0700, Steve Strobel wrote:
> On Tue, Feb 14, 2012 at 8:49 PM, David Rowe <da...@rowetel.com> wrote:
> >
> > 4/ Write an interleaver to re-arrange bits in a frame using a random
> > pattern known to the encoder and decoder.  Useful so that a few
> > consecutive bit errors are spread randomly over a frame.
> 
> It seems to me that interleaving bits like that would be especially
> effective in combination with using gray code to minimize the
> magnitude of the error introduced into any field by a single bit
> error.  Maybe that is SOP for this kind of thing, but the light just
> turned on for me :)
> 
> If interleaving the bits adds significant overhead, it might be worth
> making it optional or even doing it in a separate layer (possibly
> along with FEC) that could be easily disabled.  I don't think
> interleaving bits (or FEC) will be of any advantage when the codec2
> audio is being transported over IP, as packets with errors are
> generally dropped before they ever reach the VoIP application.
> 
> On that subject, does anyone know if packets with errors can be
> captured rather than dropped by putting a network interface into
> promiscuous mode or something?  If so, it might be possible for errors
> that are introduced in the last hop of a IP connection (WiFi, 3G) to
> be corrected.  There probably isn't anything we can do to get routers
> to pass corrupted packets, but that might give us a shot at correcting
> some errors.
> 
> Steve
> 
> 



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