VoIP sickness progressing, now I have my own registar :)

I was still at unrest because I couldn't have my own SIP registar. Yesterday I Googled a lot again, and again found the OpenSER software.

I had avoided OpenSER before since it has a reputation of being extremely difficult to configure, but this time I found a 'hot' information that OpenSER does NAT translation in belalf of users, and also found an "OpenSER" wizard to aid first configuration.

The main complaint is that parts of OpenSER configuration are scripts, not switches. Personally I found myself at ease with this.

Well, I used the wizard, configured the items that wizard cannot do for you e.g. creating the MySQL database and SIP users, and put it to run. Magic: it translates all SDP sessions transparently, you don't need to configure anything at client side (not even STUN server).

To say the truth, OpenSER does not do RTP relaying by itself, but it possesses all the needed "intelligence" for it, and can drive a relay software like RTPProxy. It is just a matter of installing rtpproxy and pointing OpenSER to rtpproxy control socket, the rest comes by itself.

Probably Asterisk can do the same as OpenSER does (and much more) but for an evening experience, it helped me to do what I wanted.

So now I have my own SIP domain and phone. I'm not sure if I will use it in place of Gizmo; my Gizmo account has dial-out and other nice capabilities. But I feel good :)