[SWLUG] SIP questions

[Steve Hill]

On Thu, 20 Aug 2009, Dick Bain wrote:

> I sort of heard some discussion of SIP at the lug meeting but old age
> makes ones hearing feeble, could some of those questions be answered
> on this list?

There weren't many specific questions that I can remember, just very 
general stuff.  I'll try to summarise off the top of my head:

SIP is one of a small number of standard IP telephony protocols (by far 
the most commonly used - H.323 has more or less disappeared these days in 
favour of SIP).  It is used by pretty much everything that isn't Skype - 
Skype use their own proprietary protocol which they guard very closely and 
ensure it doesn't interoperate with anything.

The question was asked: how do you make calls to and from the PSTN (public 
phone network) using SIP.  The answer is that you pick a SIP->PSTN 
gateway, of which there are many, and use it to dial out to a real phone 
number.  Similarly, there are many PSTN->SIP gateways which will assign 
you a telephone number and forward calls to a SIP address.

voipuser.org is a good starting point for trying this stuff out - they 
provide 0845 and 0870 numbers that they will forward to a SIP address. 
They also allow short calls to the PSTN for free (calls to the PSTN are 
limited to a few minutes and intended for testing purposes).

sipgate.co.uk provides geographic numbers for free, which they will 
forward to a SIP address, and do pay-as-you-go calls to the PSTN (similar 
to a pay as you go cellphone, you credit your account with some money and 
then the calls you make run down that balance).

There are many more SIP<->PSTN gateways, so it is very much a case of 
shopping around and seeing who does the best deal for you.

> First of all I didn't catch the name of the guy who came with Just,
> Alan and Telsa; he seemed to have some things to say on the subject??

It was me. :)

I make quite a lot of use of SIP - my business partner is in Sommerset, so 
we make SIP<->SIP calls to keep in touch.  We also run a CallWeaver server 
which handles calls from our customers - they dial a normal phone number 
and it rings through to both of us over SIP.

I also have a Cisco SPA3102 VoIP gateway which my normal phone line and 
analogue phones connect to.  That essentially consists of 2 separate SIP 
useragents, one for the phone line and one for the analogue phones. 
Normally that connects to my CallWeaver server, so calls coming from both 
my phone line and the internet are all handled in the same way (and can 
answer both SIP and calls from my BT line on my analogue handsets, my SIP 
handset or a softphone running on any of my computers (I use Ekiga).  If 
the CallWeaver server becomes unreachable or there is a powercut, the 
SPA3102 automatically connects my analogue handsets directly to the phone 
line, so they continue to work.

-- 

  - Steve
    xmpp:steve at nexusuk.org   sip:steve at nexusuk.org   http://www.nexusuk.org/

      Servatis a periculum, servatis a maleficum - Whisper, Evanescence