Neil D. McAliece wrote:

>I've read (& I guess it makes sense) that the low bandwidth codecs are worse 
>when there is any packet loss. Each packet lost is going to represent a higher 
>percentage of audio that is screwed up compared to a higher bandwidth codec. 
>
>I can't remember the last time I saw any packet loss on our ADSL connections.
>
>We are using g729 for everything including ATAs and IP deskphones and PABXs. 
>Quality seems fine.
>
>We have 3 sites linked by VPN with ADSL connections dedicated to VoIP (with 
>Internode). 2 of those sites have Asterisk PABXs with a IAX trunk running 
>between them. The main Asterisk box has an IAX trunk out to a provider (Nehos, 
>was called I-fone until recently). Last time I spoke to Norman at Nehos, they 
>were also using Internode for bandwidth which couldn't hurt. They are chasing 
>the business market and seem to be very serious about call quality problems. 
>Their call provision is built around Asterisk, so they are very knowledgeable 
>re Asterisk issues.
>
>We are using the commercial g729 licences on the Asterisk machines. 
>
>IAX is much more efficient than SIP when trunking is enabled. Somewhere around 
>40 simultaneous calls with g729 on 512/512 kbit link.
>
>Quality seems fine so far. No one noticed much difference when I switched to 
>ulaw/alaw higher bandwidth codec. There are so many variables with VoIP 
>though. The cheap deskphones don't sound as good as our single SNOM 360 no 
>matter what codec I use. 
>
>We've got some sipura SPA-841s and a Budgetone 102 which sound average 
>(completely useless when you use the speaker phone on them).
>
>Until we can go with decent IP deskphones I'll finish off adapting our old 
>legacy PABXs (which mostly have decent speaker phones attached to them already)
>  
>
Thanks Neil, that's great.

Did you build your own asterisk box or did ifone supply?

David



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