T1 Voice Card and Asterisk

Our company wants to offer voip service to residential and business customers. Initially, we have 500 to 600 aspiring customers and most probably will grow to thousands in the future. I'm planning to use Asterisk.

my question here is this:

How many lines a T1 Voice card has? Would that enough for that number of customers? I'm not sure how many concurrent calls may occur. Do you have any recommended T1 card from Digium maybe? Please advise if you have this experience.

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I'm planning to use Asterisk.

A strange decision? Are you absolutely sure you want to do that? Do you realise that you will be proxy-ing every single call, and are you fully aware of the bandwidth implications of that?

Most VoIP companies use peer to peer systems like SIP for this reason. To be honest, you really don't want to be doing this with Asterisk. Asterisk is a fantastic bit of kit, but it's just a PBX.

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How many lines a T1 Voice card has?

It's the matching end for a T1 line. That's usually 24 simultaneous calls.

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Would that enough for that number of customers? I'm not sure how many concurrent calls may occur.

That's a question (along with the asterisk point) which, being blunt, indicates you need to do a little more research:-

http://www.erlang.com/calculator/

Most of these work on a basis of scale - at your level, it's likely to be highly inaccurate. In the absence of further information it's not really possible to give you any kind of accurate numbers.

Dean

Let's say, i have 2 x T1's. One for voice and one for data. I will check the erlang calculator and see how it works on number of calls.

You're saying that this will not work... Do you have recommendations?

Yes, you should be using a SIP server like SER or openSER.

Asterisk will give you major scalability headaches.

I'm not knocking Asterisk btw - its a fantastic product, but it's a PBX, it's not designed for running a VoIP ISP...

Have a look at:-

www.openser.org/

That's what we use here. We have 3,800 users currently logged into it (as I look at it right now) and making calls etc. Bandwidth usage is practically zero and CPU is running 99% idle.

You will also need a STUN/TURN server for users behind NAT devices. Where you have two users behind NAT devices you will need to proxy audio, so you'll need some form of proxy in place and some means of deciding when to use it.

See Jasomi:-

www.jasomi.com

For hardware that will do that for you. You're probably looking at a peerpoint SBC:-

http://www.ditechnetworks.com/platforms/sbc.html

That's about $10k, although you can always build your own (we're currently doing that, although it's a long term project for VoIP User).

As the ISP you're in control of far-end connections, so you might be able to get by without any kind of proxy device, especially if you're not intending to peer with other VoIP ISP's (openSER's NatHelper module will probably do the trick).

Dean

I'm looking for a consulting company for this openser or SER. Do you have any idea?

There's quite a few here. You may have already been approached by PM. If not, Daniel I believe is still up for hire, and he's the guy that wrote it so you can't really do much better. "team @openser.org" will get him by email.

If he can't help I'm sure he can point you in the right direction.

What you'd be looking to do is have openSER/SER at the front end handling registration duties (and peer to peer call hand off if your users are calling each other) with your asterisk setup at the back end operating as a PSTN gateway. With openSER taking the brunt of the registration work you'll be in good stead up to a 5 figure number of users until you need to think about load balancing.

When a user wants to call the PSTN, openSER would hand off to your asterisk box which would be limited by the 24 line capacity of your card anway. If you need more than 24 simultaneous calls, you'd simply bolt in a second asterisk gateway, or start outsourcing some of your PSTN routing to other providers who you could hand off to directly.

That way you have persistent state duties (like registrations) handled in an efficient way on a single server and maximum flexibility for expansion.

Dean

Hi

I have to agree 100% with Dean, Asterisk is a PBX not a CO switch. It can make a very good gateway though. But what you have to think abut is, One of the reasons your users will want want too use VOIP is to save money, By providing a gateway of your own for 600 users you will not get near the volume of calls to get a decent rate that will beat the wholesalers.
As to number of cards, It depends on the Level of service you want to provide, a guide is between 5-10 users per trunk so for 600 users you will need between 60-120 channels. Are these to be the incoming DDI (DID) channels as well?

Have you thought about using a white label service till you get the critical mass of users, Then migrating to your own platform?

Ian

Yes, incoming/outgoing. I got a vendor works with OpenSER/SER/ASterisk etc ... I think, they're experience.

i would say, we'll start at 8 x T1's. What do u think?

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i would say, we'll start at 8 x T1's. What do u think?

Impossible to say as it depends on call volumes, and it's impossible to predict call patterns with a low number of subscribers.

Best thing to do would be to try it, although with an initial roll-out of 600, gut instinct says 8 T1's is overkill. That's 1/3 of your subscriber base on the phone at any given moment. Are you expecting that level of call volume?

Dean

We will try and if it is saturated, then add more card. I believe, it will not start 600 customers at the first day.