The VoIP virus has bitten me :)

In the last few days I have played quite a lot with VoIP. It is funny because I was never interested on the subject while everybody else was (around 2 years ago) and now that everybody lost the interest on that, I caught the "disease".

Anyway, the VoIP technology is far from settled. As I could see, there are three main groups of competitors out there:

* Skype, that is proprietary, closed and secret but offers a complete, free, functional and easy-to-use solution;

* "Closed" SIP networks that follow a traditional telecom model: SIP-to-SIP calls are only allowed within the same provider, calls to outsiders must go through PSTN even if the other end has a SIP address. In Brazil, GVT Vono is in this group (very good service indeed), it seems that Vonage and most other businesses follow the same model;

* "Open" SIP networks, that allow free SIP-to-SIP calls and only bill calls that go thru PSTN.

I began to use Skype to talk with my manager and also to allow my wife to make cheap calls to her family that is 3000km far from here. The importance of the service made me to think about buying a cordless phone with Skype software (Netgear SPH200D) but the price tag is a bit high, and it is not easy to find this one in Brazil.

Also, as a free software developer and supporter, the Skype model annoys me a bit, in particular running a closed and secret software in my machine. I'd feel a lot better if it ran in an appliance, so if it were attacked (unlikely to happen anyway), at least would not be my PC.

In the other hand, SIP ATA hardware can be found in every corner here, due to the fact that most VoIP providers do use SIP protocol, and in particular due to Vono's success.

Unfortunately the Vono service is of no use to me, since it bills long-distance calls as if it were a normal PSTN call (my wife's parents have no computer) and does not allow SIP-to-SIP calls. It uses SIP protocol but does not buy SIP philosophy.

The SIP protocol has a very bad interaction with NAT, but everybody has NAT nowadays... In theory, anybody can put a SIP phone available in the Internet, a DNS name (even DynDNS) is all people need to find you. This is true when the VoIP software or appliance has a public IP address. If it does not, things get a lot more complicated. IAX and Skype protocols are much more well adapted to NATs.

To remedy this, enter the NAT-enhanced SIP proxy. It is a SIP that accepts user registration, makes and receives calls in behalf of them, and more important, rewrites SDP media descriptions so the RTP data flow passes through an intermediate server instead of peer-to-peer (which is utterly difficult with NATs in the way).

That is essentially the service all SIP VoIP providers do for you: a data relay to circumvent NAT. Providing a registar service without media relay would be dirt-cheap; the media relay is the "expensive" part since the network connection must have network bandwidth to retransmit all conversations (around 32kbps per client, given an average codec like G.729).

Most SIP software and ATAs have STUN support these days, there is automatic detection of the (few) lucky terminals that have public IPs, who can avoid the relay and do true P2P conversation, with the additional advantage of reducing latency.

Of course I would like to have an "open" SIP service (i.e. being able to call any other SIP number), without having to change my small network (i.e. keeping NAT). Apart from Vono, I tried two services: FreeWorldDialup and SIPPhone/Gizmo. I have accounts in the two right now.

SIPPhone is certainly the best. The configuration directions were much clearer, it was just a matter of reading the e-mail, configuring X-Lite and everything worked well. The thing is even easier if you download Gizmo, a closed-source client for SIPPhone. It is pretty much like Skype: just asks account name and password, and everything works.

Even the Gizmo call-to-PSTN rates are similar to Skype, so I would say that Gizmo is trying to be a kind of "Open Skype", that is, offering a verticalized solution with PSTN gateway but keeps open doors for non-Gizmo SIP softphones, as well as SIP hardware.

This is good news since, as I said, SIP ATAs are cheap and can be found in every store. Also, it allows SIP-to-SIP calls, so it is a true part of SIP ecosystem and gives Gizmo a escalability advantage over Skype. My doubt is whether Gizmo will make enough money to keep operations in the long term (well, Skype is well off, so it may happen for Gizmo too.

FreeWorldDialup (FWD) provided directions but they failed to work in X-Lite. Either I got an 600 error, or the calls did not receive audio, or the audio would not be sent, or an authorization failure would happen. After some digging in FWD forums, I found that FWD SIP servers have issues with X-Lite, using an alternative server made the thing work. Both STUN server and SIP proxy must be configured, the instructions did not state that clearly.

So, in conclusion, FWD works, but it is not user-friendly. A simple change in SIP proxy server (to remove the 600 error issue) and instrutions made more clear would improve a lot the friendliness of the service.

FWD has an interesting "call out" scheme, quite radical but very innovative. You must connect an Asterisk server to your PSTN line, so you become a VoIP-PSTN gateway and people can make PSTN calls through your telephone. In exchange, you get the right to do the same via the telephones of the other participants.

The rationale is to seize the free minutes every telephone has, so the increase in PSTN bill would hardly be noticed. The Asterisk server would be programmed to limit the number and distance of PSTN calls, to avoid a runaway bill. The more calls you allow, the more PSTN calls you would be allowed to do.

Honestly I don't believe that this scheme will work, but it is nevertheless inovative. IMHO the main problem is to convince people to allow others to use their telephone, and on top of that you would need to keep a 24x7 Asterisk server running. If FWD could provide a modified firmware to some popular ATA, things would be a bit easier.

So, now I have a Gizmo SIP account, and will probably buy an ATA to use it without a computer, so I no longer need SkypeOut.

I still have the wet dream of having my own SIP domain account. Since I have my domain server with public IP, I tried to find some SIP proxy software to install on it. The nearest candidate was Siproxd, but it is aimed to be installed in the NAT firewall, not in an external machine. I tried to make it work that way, but I quit in the SDP/RTP translation part; I would need to add STUN support and it is not a task for just a couple evenings. (If someone knows a SIP proxy server with SDP/RTP relay that works well as a public service, let me know!)