I am assembling my own SIP VoIP infrastructure, and have the endpoints ready: a fancy Cisco and digital PRI at the office and a Sipura 2100 at home. The only remaining part of the puzzle is the SIP registration server. I am trying to determine what servers are available and which is the most convenient. A quick research turned up these potential candidates:
Asterisk * SIP Express Router: An Open Source SIP proxy/router * OpenSER: GPL SIP server
Question Number 1: Can Asterisk be a SIP registrar for non-Asterisk calls? Should I use a dedicated server instead?
Yes. And it can do a lot more: it can register on other servers, acting as client; route calls, handle voicemail, conferencing etc. like a PBX; translate between different protocols (SIP+RTP IAX) and codecs (G.711, GSM, iLBC etc.)...
It depends on the amount of traffic you want to handle: some large sites such as FWD use a combination of SER and Asterisk. But if it is a small infrastructure, even a single registrar might represent overkill: you could e.g. use the free service of Like2Fone
) or FWD. However, this would assume that you may work around NAT issues, which can represent a thorny issue depending on your NAT router. Hint: if you have more than one phone on the same NATted LAN, make them listen on different UDP ports, and on each enable both STUN and symmetric RTP.
Alternatively, if you can, you should run Asterisk on top of the NAT router, which is possible if the latter is a Linux machine or a Linksys WRT54GS reflashed with OpenWRT. If Asterisk binds to the address
0.0.0.0, it will listen to both internal and external interfaces without any natting: so the phones on the LAN will register on it, and in turn it will register on any number of external registrars.
It is also possible to run Asterisk behind a NAT, but as Asterisk doesn't support STUN, you'll need to configure its SIP service telling it the external IP address of your router and the local network IP/netmask. This may be problematic if your Internet connection has dynamic IP address! In that case, you may be avoid a lot of headaches giving up SIP for the connections between Asterisk and the rest of the world, and use IAX. This will still allow you to connect to a number of free providers (e.g. FWD) and commercial PSTN termination providers (Voipjet, Voxee and others). Your phones may still use SIP to talk with their local Asterisk(s).