I think the guys over at the Gizmo Project have got the right idea behind that their mix of open standards and design. It must be remembered that Skype is popular now, because it provides something that is rarely available in the open source world (and I am a big user and fan of open source), which is a consistent user experience. Users don’t care what standard is underneath a piece of software they are using as long as it makes their lives easier and it works ALL the time.

I haven’t tried the packages that you mentioned and I do believe you when you say they “Just Work”, but unless they can, from the users POV, demonstrate a CLEAR advantage to the lay man over Skype, you are not going to see a shift away from the commercial packages. Firefox pulled market share away from IE by providing features and speed that were OBVIOUSLY superior to IE.. but even it still does not hold an enormous market share because most people still don’t see enough of an advantage to stuff around downloading and installing something unfamiliar. This is even more pronounced for Open Office, where people would prefer to pay Microsoft’s high price than switch away for something technically better, but functionally the same (with worse graphics) from the average user’s perspective.

The fact that SIP didn’t have a good a user experience when Skype came on the scene, regardless of what it is now, is why Skype has the market share. So until SIP can demonstrate to the END USER that it is markedly better than Skype it will not draw the public’s attention in any major way. Google’s entrance into the market complete with SIP is probably the best thing that could happen for the protocol as they have enough of a user orientated philosophy to make the innovations needed to create a clear distinction between Skype and the rest.

Do u j\know wat port number’s Google Talk use?

One comment, though: why the totally superior tone?

Skype, Google etc. are made up of ordinary people trying to do creative things. They may not get them totally right or flavour them to everyone’s liking, but there is no good reason to diss other people or businesses (especially if they are successful, becuase then it looks like “sour grapes”?)

Thanks for the great info!

The way that skype manages this (user behind firewall calls another user behind a different firewall) is it uses other users to route those calls. If you are a skype user and you are not behind a firewall, or if you are behind a home firewall that they can reconfigure with uPnP, you might become a supernode. Becoming a supernode means that skype users who are not supernodes (perhaps because they are behind a firewall) will use you as a relay.

So, using the connection protocol, eventually, both machines make inside-out connections to the supernode, and the supernode (which is just a capable ordinary user who might not know that they have been elected) relays the traffic. It looks like a single call will saturate a dial-up.

Thus, without your knowledge or specific permission, skype will automatically use some of your bandwidth to route other people’s calls, people you don’t know.

This is not ordinary practice – it would be theft of service except for the clause in the user agreement.

I’ve never met a skype user who knew that this happened. Most of them just know that firewalls present no impediment to skype – and they do not know why this is.

This is unethical – it is, to me, as if they had a paragraph hidden inside their user agreement that allowed them to stop by your house and use your car while they were in town – and it was hidden in paragraph 63 of the fine print where you expect to find the legal disclaimers of liability, rather than being up front as would, say, the cost. People expect the bandwidth that they pay for to be there to use. No doubt many would volunteer to be supernodes – and many would not to.

It is also true that many people have user agreements that specifically disallow them from running servers on their systems – and what skype does is a violation of this end user agreement.

It would be ethical of them to do this if they made it clear that you were going to be routing stranger’s calls even when you were not making use of the product other than simply being logged in bandwidth – putting this in bold at the beginning of the user agreement – making sure that people knew of this – allowing the function to be turned on or off – and perhaps only allowing people who left the function on to do the behind firewall to behind firewall connection.

It is NOT ordinary practice to use people’s memory or bandwidth to route your unrelated traffic, simply because they use your system.

If skype wanted to do this in the standard manner – they would need some servers – and they would have to route the firewall to firewall calls through their servers. Of course, initially they were a free service, and they did not have the budget to do this. And, if they had banks of paid-for relay servers running as supernodes, firewall owners could block access to these servers and thus block skype.

No one else does this – gets such good firewall to firewall routing – because all the other players in the game are ethical.

To put this another way – supposing you were using a file sharing service – and they not only used your downloaded copies to feed other people who wanted copies of the files you downloaded (while and shortly after you were downloading them), they also downloaded popular files to you – and they used you as a conduit so that they could get the files to places that were intentionally blocking the sites you could access. This might make their file sharing service work real well. It would not be ethical unless they made it clear that they would be using your hard disk and bandwidth to share files you had no interest in.

Furthermore. skype tries to pick ports that are not likely to be blocked – I’ve seen skype running on the port that https normally uses – a network administrator might have a legal requirement to record the destination of all calls made from his internal net – if he blocks all the regular sip ports and requires that those ports run through a proxy, well, skype will just pick some random port that is likely to be trusted and run its traffic on that port. Or so the rumors go. It might be that they use port 80 and 443 more or less exclusively.

The simple thing to do is simply to fire any employee who installs skype on a computer that is company owned or plugged in behind the firewall – if your corporate policy restricts skype. But the reality is that network admins will get told “enforce this” and not be given any authority to have personnel actions taken when people break corporate policy.

This usually is not a problem – firewall vendors can determine how protocols work and provide filters on ports and contents. Skype has taken steps to make this impossible – just so that people can say, “gosh, any firewall I am behind – it does not matter, I can make a connection to anyone.”

There are a couple firewall vendors who claim that they can block skype. They do this by doing packet analyss of all packets.

http://www1.cs.columbia.edu/~library/TR-repository/reports/reports-2004/cucs-039-04.pdf
is a paper that describes what it can of the skype protocol. They don’t attempt to dig in to the protocol or decode the traffic (which would be possible with reverse compilation – they could trap the packets at the exit of the AES routines, for example.

The nodes that are used to relay others’ calls are called “supernodes” and you can’t stop yourself from being elected as a supernode. From this protocol analysis, it looks as if there might be a simple way to do that – just bind to ports 80 and 443 – since it looks as if the initial connections are made via UDP and then 80 and then 443 to a supernode when someone connects to the net – so if you are already bound to those ports before you start skype, you can block the people behind firewalls from connecting to you.

- Voip island -
If you subscribe with any provider should be able to call any user
at any provider. ( The way actually email is working )

- Most SIP phones don’t support text -
Try do dial in a phone bugs-bunny@whatever.you.like.com

- Easy Dial -
If the above its a nighmare give us a working method “now”
to convert text to number , realm to as number and so on.

- NAT and Firewall issue -
People dont want to bother with tecnical expertize.

- Convergence of Video, Audio, IM, Identity and Presence -
People want to comunicate in any ways.

I think that voip with SIP is only very slowly gaining ground with regular users. Things like WiFi phones, analog telephone adapters (ATAs), routers with voip functionality and PDA soft phones are making it more accessible.

Only by educating people about the importance of open formats like SIP can the voip industry flourish and users see a lot of advancements. So once again thank you for your great post.

In some way Skype is making people aware of voip and hopefully makes them interested in open SIP voip.

Make sure you upgrade the firmware to the latest version. It has some features that make this nicer.

The SPA 3000 is an intelligent an very configurable telephony router.
It has three connector. 1) Analog phone 2) Ethernet 3) Analog land line.
The limits are your imagination, and you can make it work with asterisk for extra possibilities.

My 2 cents.

It will certainly not stay free forever, but it sure is nice for now.

SkypeIn numbers are currently available in the US, UK, France, Hong Kong, Denmark, Finland, Sweden and Poland.

Boo. The countries I would have been interested in are Australia, Papua New Guinea and UAE.

WHAT CAN I DO? to you need my money, Im very sorprised, this is the 5 letter I send you.NO ANSWERS>>>>>>>
bye.

Federico Kristal

It’s true I missed SkypeIn to start with, but I did not miss SkypeOut. It’s also true that the info on Skype’s site for both is rather sketchy, and for SkypeIn is so sketchy that it is almost useless.

The rest of your comments reflect numerous misconceptions. If you think SIP is that hard, why not just download the Communicator from FWD and watch it Just Work?

You are comparing apples to oranges. You are comparing the Skype client, which comes pre-configured out of the box to work with a particular service, to a generic SIP client that does not come so configured. *Of course* you will have to configure a generic SIP client. Or, you can get one that’s already configured (such as the FWD one) and it will be no more difficult than Skype.

As for dialing, you press the numbers and the call connects. Just like any other phone. Can’t get any simpler.

You are also misrepresenting how SIP pierces firewalls. It uses the central server only for *connecting* the call. After that, the traffic goes point-to-point. There is no fuss with sending data streams through a centralized server.

I can see your point for text messaging, although personally I have *never* felt the need for that. SIP already is used for video. A little research revealed that SIP supports RFC2793 for text messaging, and some programs, and even Polycom hardphones, support it today. Still, I’d rather use my preferred IRC client for my text messaging, and my hardphone for voice, and if I have them both open simultaneously, fine. I don’t want to be forced to use the same client for both.

If Skype is just “IM chat with good voice”, then why are so many people still so excited about it? This is not unique in the marketplace by any means. Apple’s iChat does it over AIM, Microsoft’s programs have been doing it for ages, and GnomeMeeting does it on Linux (with H.323). So you’re left with my original assertion that Skype is boring because it brings a whole bunch of restrictions and nothing novel to the modern table. SIP is not the only alternative to Skype (I even mentioned IAX in my article).

SIP is not the protocol to end all protocols, of course. It has its weaknesses, and I personally prefer IAX. But let’s have a fair, apples-to-apples discussion, shall we?

A lot of people are hyping up Skype as a POTS replacement. Probably far more than are commenting about being able to do text chat with it. Even Skype’s own homepage discusses only the voice features. It seems that voice is what people are excited about, so voice is what I’ve discussed.

Make no mistake, text chat is nothing exciting either. After all, 4.2BSD, released in 1983, contained talk(1).

Directory is not integrated into the client in SIP. I’m on FWD somewhere but a quick google didn’t turn up the directory. So on to the website. No mention of the directory in the FAQ. At that point I gave up. So then I went to check out configuration; Public IP Address (FAQ), STUN (FAQ), Outbound Proxy (FAQ) Ugh! Never mind all that geeky goodness, tell me how to call someone. My pointy haired boss would take half a look at that, give up immediately and call me. By contrast he installed Skype and started using it with zero help. Have you looked at the option pages on Xten recently? Would you describe them as dumb consumer friendly?

NAT and firewall. One solution to this is to use a proxy reflector like FWD which you mention. This approach doesn’t scale unless you have the resources of MS or Yahoo!. And it often involves round trips right across the net so two UK clients have to go via Seattle. My experience with this has not been good. By contrast, Skype does not use centralised proxys but distributed call setup supernodes. And once the call is setup, the link is P2P and no longer involves the supernode/proxy. So not in a manner similar to Skype at all, at all.

“why would I need text if I’m talking to someone” To pass them a URL, among other things. IM and Voice go well together. They are both real time communication. Presence is more than just free/busy. And it’s important. This is another reason why the convergence of Video, Audio, IM, Identity and Presence is important. It seems to me that the SIP clients seen so far (with the possible exception of Gizmo and FWD.Communicator) only really deal with one of these.

This is beginning to feel like the usual back and forth of zealots. And I said earlier, perhaps it’s because we’re talking apples and oranges. Skype isn’t VoIP, it’s IM chat with good voice. SIP isn’t real time communications or IM, it’s POTS telephone replacement. And that last sentence is a sound bite and only partly true.

As for Google Talk, I’m waiting for v0.3 because v0.1 is little more than proof of concept. A loose cartel of Google, AOL, Apple, Jabber, SIP could change the game completely. But not yet. They’ve got to prove they can ship early and often and fill in the blanks first.

At the same time, it seems to me that you haven’t researched SIP all that much.

The fact that SIP phones don’t have a single directory is, IMHO, a *feature*. I can use FWD with my phone, or I can choose not to and use someone else, as I wish. That’s a big plus to me. I have choice, and I can use whomever I want. Most of these places, though, do have directories.

Conferencing, etc. are commonplace features in SIPland and work fine. As I’ve already said, NAT and firewall issues are rare with SIP these days, especially if you are using a central server like FWD (in a manner similar to Skype).

It is true that most SIP phones don’t support text. But OTOH, why would I need text if I’m talking to someone already? I really don’t see the point for this integration. If you have audio (and even video), why bother with text as well?

As for free/busy information, that’s up to the SIP provider(s) you choose. FWD does publish that info.

Let’s say that you just want to do pure-IP conversation. That is, both you and the people you speak with have Internet phones. The easiest way there is to go over to FWD and download their communicator. It’s pretty much a click-and-go proposition.

Perhaps you want to be making calls over the Internet to regular (PSTN) phones. The first thing to do is find a provider. I like Voxee, and you can get started with them for as little as $5. They’re more oriented towards technical folks though, so if you’re not technically inclined, you might try someone like Broadvoice instead.

Next, you need a SIP phone. Good starter phones would be the Sipura SPA-841 for a hardphone (about $90 from Voxilla). For a softphone, I recommend the free SJPhone.

My question is: what should I use?

I have been w/o a landline for two years but now have the need for one. I don’t have any phone infrastructure to worry about abandoning, so anything is game. I have broadband and a wireless network.

What do you recommend?

But there’s also an element of comparing apples with oranges here. All the major IM packages now support voice. Does that make them VOIP programs? If you compare Skype with MSN messenger or Yahoo! Messenger it comes out pretty well. It’s got some features that beat them hands down. So is Skype a good IM program with great voice, or a VOIP program with good IM?

Don’t get me wrong, I wish Skype was more open and used SIP or at least had a SIP gateway. And I wish the SIP community would come up with clients that are as good. But right now it’s got momentum and it’s a damn good product.

As for Google Talk. It’s clearly unfinished and effectively an alpha, not a beta. *If* they ship early and ship often. *If* they play nicely with the rest of the jabber community and join the network. *If* they support SIP and interoperate with other SIP phone network. *If* they beef up the IM chat client. *If* they link up with POTS IN and POTS out services. Then maybe it will become interesting. But right now it’s a toy.

I’m pretty interested in this SIP thing and may set it up at my new place, especially because it has a wide range of phones you can use. Big disadvantage with Skype is the limited amount of hardware out there.

One is the IAX protocol. This protocol was originally designed by the Asterisk team, but is really starting to catch on. It’s not nearly as popular as SIP for hard or soft phones, but a number of providers (including FWD and Voxee) are supporting it these days. IAX was designed from the ground up to be NAT-friendly.

The other is that if you are using SIP to dial-by-IP, then yes, NAT will still pose a problem. I don’t know of anybody actually doing that anymore, though. It’s so much more convenient to use an agent such as FWD. (Of course, these agents are just used to establish the connection; it’s still a direct peer-to-peer link once done.)

Anyway, I have never had trouble with SIP and NATs. Most SIP phones/equipment these days should Just Work. I once opened up Asterisk on my home firewall, even carried my SIP phone from my own house to my parents’ place, connected it to their ethernet (behind a NAT), and it hooked right up to my home network. Calling my home phone made it, as well as the phone at my parents’ place, ring, and it all worked just fine.

I know that in the earlier days, this was a problem, but really, I just don’t see it anymore.