[Asterisk-Dev] Petition for IAX firmware

[Jerris, Michael MI]

WRT is a perfect platform for this sort of device.  Running chan_sip and
chan_iax2 along with core asterisk and a few other modules would be able
to do this quite nicely.  The pluggable pbx could be used quite nicely
in this case to strip all dialplan processing totally off the platform
and replace it with a hardcoded dial across a trunk.  Given no
transcoding or other manipulation of the audio payload, I think we may
all be surprised at how well a device like this could work.  

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Greg Hill
Sent: Wednesday, April 06, 2005 12:45 AM
To: Asterisk Developers Mailing List
Subject: Re: [Asterisk-Dev] Petition for IAX firmware

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

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