[Asterisk-Dev] Petition for IAX firmware

[Greg Hill]

On Wed, 6 Apr 2005, Paul wrote:

> Steve Kann wrote:
>
>> 
>> I have two points to make:
>> 
>> 1) If I were a vendor, and there was documentation on the IAX2 protocol 
>> (not even an RFC, but at least some kind of semi-official documentation), 
>> I'd be a lot more likely to implement it.
>> 
>> 2) The idea of an IAX2 trunk-aggregator is interesting.  This is probably 
>> something that doesn't need a whole x86 linux box -- taking multiple IAX2 
>> streams and putting them into trunks is trivial in terms of computational 
>> requirements..  A small microcontroller or an ARM chip is more than enough 
>> for this, and would be a neat idea of virtual PBX deployments..
>> 
>> -SteveK
>
(snip)

> But I still don't know the answer to my question about SIP vs. IAX2. 
> Suppose the remote site has a mix of SIP and IAX2 devices. Does the 
> presence of SIP devices increase the computational requirements much? My 
> thinking is that any SIP ata's or phones at the remote site are going to 
> be extensions of the master * pbx. Hopefully that makes it easier on the 
> trunk-aggregator cpu. Also I expect that in most situations where this 
> was deployed a codec other than g.711 would be used since there is a 
> motivation to conserve bandwidth. So we have SIP traffic from a provider 
> to the master server, IAX2 trunking to the remote slave server and back 
> to SIP over the LAN to somebody's deskset. Will that conversion back to 
> SIP to reach the deskset degrade the call quality?

SIP and IAX are control protocols. They don't deal with audio coding. 
Converting a call between SIP and IAX protocols should require minimal 
(no?) re-writing of the audio packets if the same codec is used on each 
side. So it seems (to me) that a translation between SIP and IAX wouldn't 
be a computationally expensive task.

Greg