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On Wed, 7 Oct 2009, Grant Sewell wrote:
Hi all, I've been asked by the-people-who-wear-ties to look into out phone costs, specifically to compare costs of international calling with our current provider (UKTelCo) with Skype. I'm broadening my defined scope of comparison and would like to look into other forms of VoIP as well. It would seem that Asterisk et al are the way to go. And that's where my knowledge of VoIP seems to stop. We have 2xADSL lines, neither are being fully utilised, so we could have our VoIP going through one of those. Do we need to sign up with a SIP provider or can we be one ourselves? If the former, any recommendations? If the latter, how to do it? Etc, etc. Any "idiots guides" to VoIP/SIP? The vast majority of things I've come across seem to jump straight in with an assumption that you already know how VoIP/SIP systems work... I don't.
OK... So bearing in-mind that I do this for a living... I'll try to not be too biased :-)
Firstly - what is the aim? Just to have 1 or 2 phones, or to convert the whole company - and if so - how many phones.
(I'll use phones is a loose sense - either desk phones or soft-phones although I don't like soft phones in a business setting purely due to the hassle of being wired to your desk and getting annoying pop-ups on your screen when someone calls!)
Skype is obviously free - but only of the other party is also on Skype. I'm personally biased against it because (a) it competes unfairly with me and (b) lack of support in a business manner. Saying that - it does work well a lot of the time and I know a lot of people who'll use it, and only fall-back to using the PSTN (or other VoIP) when it stops working!
There are Skype <-> SIP gateways now and there is a product (because you have to pay for it) which integrates Asterisk to Skype. I have not tried it yet. You pay per channel (ie. concurrent call)
You can do direct SIP to SIP calls without using the PSTN - but like Skype, the other end needs something compatible. (And NAT is going to stop it working anyway)
Asterisk is a software PBX which runs under Linux (and others, but mostly Linux) It can handle PSTN connections as well as VoIP connections - usually needs a PCI card to talk to PSTN - analogue or ISDN, but there are external adapters. It acts as a SIP end-point - so local phones can talk to the asterisk box and it can do the clever stuff like work out which phone to make ring when you dial 212, and then forward calls out via an external connection (PSTN or SIP) when you dial 0... and take incoming calls and so on. With SIP phones and a PSTN connection you're effetively your own SIP provider, but ... as it sounds like you want to reduce costs on your current PSTN connection this doesn't sound like a way forward.
SIP providers - Other than myself, there are many. Sipgate do individual lines and numbers - not really aimed at a PBX endpoint, but I know people who use it this way. Gradwell offer both a virtual and "trunk" service, but charge per SIP account as do the likes of Telappliant, etc. (charging per SIP account is common - because they're usually licensed from the hardware venduhs like Cisco and so on. I'm unusual in that I don't) If you want free, look at the Betamax resellers, http://backsla.sh/betamax but you get no support. There are a few others too - just google or check the ITSPA website - http://www.itspa.org.uk/
Running with 2 ADSL lines is ideal - but ideally you'll need a PBX (asterisk) to handle the routing for you - unless you've already got a clever load balancer thingy fronting the 2 lines. I have a few clients setup like this - the easiest way is to make the primary line do DHCP, etc. for the network and the 2nd ADSL router just be another device on the LAN, (not doing DHCP) and make the default route of the Asterisk PBX point to the 2nd router while everything else points to the primary one.
Number of concurrent calls: A simple rule of thumb is to allocate 80Kb/sec *each way* per concurrent call. So if you anticipate 4-5 concurrent calls, then you'll need at least 400Kb/sec outgoing. Using compression (e.g. G729) gives you many more concurrent calls. (Usually 3 times more)
And even with a dedicated ADSL line, it still needs to go via a decent ISP.
A common scenario I do is install one of my PBXes and start by using existing PSTN for incoming and outgoing calls, then PSTN in, VoIP out, then totally VoIP in & out - migrate slowly over weeks or more, as required. Numbers can sometimes be ported - but right now I can only port BT and Telewest numbers. (And that means ones that have been ported out of BT are not available for me to port - maybe one day) What I can do is present existing numbers out via VoIP though - so say you have 4 PSTN lines and the majority of your calls were outgoing - you may be able to cancel 2 of them as all outgoing would be via VoIP, leaving 2 for incoming, but still presenting the existing incoming number on outgoing calls... and so on...
An alternative might be to VoIP enable your existing PBX (assuming you have one) if it's got a few ordinary analogue ports, it may be possible to front it with an analogue adapter of some sorts, so there's no wiring or phone changes, etc. Might be a good way to start, depending if you needs the features that a newer PBX maybe able to offer. ISDN ones can be done too in this way, but it's a bit harder (and more expensive)
If you want more details, and some real prices, etc. drop me an email. Gordon -- The Mailing List for the Devon & Cornwall LUG http://mailman.dclug.org.uk/listinfo/list FAQ: http://www.dcglug.org.uk/linux_adm/list-faq.html