[Asterisk-Users] It's possible to do a codecs translation during a call in Asterisk?

[Matt Riddell]

Raúl Gómez Cabrera wrote:
> Hi everyone,
> 
> We are using the IAXy boxes and Asterisk over the internet and I was
> wondering if Asterisk can do a codec translation during a call in order
> to lower the bandwidth that the comunications consumes?
> 
> I mean, the IAXy boxes only support the ADPCM and uLAW codecs, but for a
> certain number of calls our bandwidth runs out, then I think if Asterisk
> can convert the signal that comes in ADPCM format (the lighter codec out
> of these two) to another codec that use less bandwidth to interconect to
> another Asterisk Server via IAX2 protocol.
> 
> We are trying not to upgrade our 30 internet connections because the
> cost of it will be multiplyed by 30 (obiously).

First a couple of keys, so we know we're talking about the same things.

Your setup (as I understand it) is:

IAXy -> Asterisk A --IAX--> Asterisk B

The easiest way would be:

Asterisk A should have accounts in iax.conf for the IAXy's and the IAX 
link to Asterisk B.

In the section for the link to Asterisk B, put:

disallow=all
allow=gsm

and do the same in Asterisk B's iax.conf file for the Asterisk A entry.

That way you will end up with:

IAXY --adpcm IAX--> Asterisk A --GSM IAX--> Asterisk B

Asterisk A will convert from adpcm to GSM and the link between will use 
this.  That way you have highest quality on your LAN (where bandwidth is 
unimportant) and then a compressed codec for traversal of the WAN (where 
bandwidth is obviously more important).

Make sense?  :-)

-- 
Cheers,

Matt Riddell
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