[asterisk-users] IAX vs SIP trunks between Asterisk boxes

[Gordon Henderson]

On Thu, 4 Jan 2007, Noah Miller wrote:

> Hi Damon -
>
>> Can anyone comment on the overhead added when a SIP call comes into one
>> asterisk box, is routed to another with IAX instead of SIP, and is then 
>> sent
>> to the UA from the second box with SIP?
>> 
>> DTMF passthrough issues?
>
> I've got a client with sip phones on several different servers and
> IAX links between the servers, so I guess that's pretty similar to
> your setup.  I've never bothered to check for overhead since it was
> never an issue (all servers are P4 2.8 ghz or Xeon 2.8 ghz, 1GB ram,
> with never more than 3-4 calls going through any one of the IAX
> links).  I can say that DTMF works fine in this setup.

I'm doing the same on 1GHz processors - CPU usage is virtually nil unless 
there's transcoding going on (about 4% per GSM transcode)

ADSL bandwidth is more of a concern for me in these applications )-:

Gordon