IP telephony

Dirk Koopman djk at tobit.co.uk
Wed Oct 18 17:23:03 BST 2006

On Wed, 2006-10-18 at 15:57 +0100, Toby Corkindale wrote:
> Hey,
> Is there any service around that will route real, live telephone numbers
> through to a server's IP, via internet, using some kind of open-standard
> protocol? (Eg. SIP)

Are we talking existing telephone numbers or a new block?

> Not just for a single phone number, but a whole block.
> ie. So that one can run an office PBX without needing to buy an ISDN or SS7
> card, and all the faffing about with that kind of thing.

And are you aware of the issues? 

* You need a line that does not have any other services (eg web, mail,
ftp) etc. Otherwise someone sending you a large email (for instance)
will completely wipe out any other incoming RTP (whilst letting them
still here you - most frustrating). 

* It also helps if people in your organisation don't use that line for
downloading huge volumes of stuff (otherwise the opposite problem occurs
- it will clobber outgoing RTP). Having the correct router would allow
you to use QoS to fix this.

* You need an ISP that understands the issues and is happy to let you do
it in the first place. More and more are doing their own
"triple"/"quadruple" play thing and are therefore reluctant / unable /
don't understand how to do useful things like QoS at their end (which
would mitigate the first issue). 

* Using asterisk works well, except that it needs to be configured
correctly, by someone clueful. A wrong entry may stop the whole thing
working and the config files (particularly for a large organisation) are
long and easy to get wrong. There are solutions to this of varying
efficacy, but I still do it the "hard" way. 

* If you go asterisk and RTP then go to someone like Voiptalk.org /
Magrathea (same people really) and use IAX. It's more efficient. You
only really need one number, because you can multiplex several
conversations down the same channel.

Finally: don't dismiss using one of Digium's E1 cards and getting a real
telephone line in. That asterisk config file is actually one of the
shortest and easiest to deal with. It also gives you guaranteed
bandwidth. Dealing with all the "issues" of WAN based VoIP will cost you
quite a bit to do properly and get the quality that I suspect you will
need. A real E1 line and a line card will be better quality, less hassle
and cheaper (at least for incoming calls). You can still use VoIP
to/from the desktops and you can mix and match the real telephone lines
with outgoing VoIP calls. 


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