Vonage have an optional service that allows you to connect to your Vonage account using a SIP phone (X-Lite). This costs $15 per month extra and is only available *in addition* to the basic service that comes through their set top box. However, you can configure Asterisk to be a SIP client on that Vonage option.
Another way to use Asterisk with Vonage is to connect an analog FXO port on your Asterisk server to the phone jack on your Vonage set top box. Not exactly very elegant but it works.
Actually, a Gartner study said that the last PSTN phone call on the planet will be made some time in the year 2020.
Considering that phone companies themselves are getting into VOIP and it is far more cost effective to deliver voice services through a general purpose packet network than through a special purpose circuit network, you can bet that VOIP will eventually take over all telephony.
Gartner may be off by a few years, but they are unlikely wrong in their assessment that circuit switched telephone networks will completely disappear because they will.
I know that Yahoo were the first to start VOIP in Japan and they have established market leadership dwarfing all other VOIP services there, it's called Yahoo BB Phone.
Depending on whom you believe it's got either over 1 million or over 3 million paying customers [recent articles returned by google bring up both these numbers, so I don't know which one is correct].
The way this works is you get a set top box from Yahoo which plugs into your DSL and your phone line. Your telephone also plugs in to the box.
Then if somebody calls you on your landline the call will go to your phone as usual. But if somebody who is also a Yahoo customer calls you using your existing phone number, then the call goes VOIP over Yahoo's network and it is free of charge.
If you call somebody who is not a Yahoo customer, you have the choice to use your landline or going through Yahoo's VOIP service at lower tariffs.
Very much like Vonage but it seems to be much better integrated with your existing landline and phone number.
Maybe Yahoo should consider a similar service in the US and other countries. Who knows, maybe they're already working on it.
that's not entirely correct and probably a misconception on your part.
I think you will find that any provider in Italy from whom you are getting a phone number for inbound calls will charge you some form of line rental.
Those who don't charge a recurring fee will generally not provide you with a phone number for inbound calls, they only provide outbound service. An I right or not?!
The only two exceptions are
- personal or premium rate numbers, where the caller pays a higher rate
- prepaid mobile plans, where you pay more on your outgoing calls than you would if you had a monthly plan
In the UK, prepaid mobile plans now make you pay the monthly line rental indirectly by charging more for the first so many minutes per day or per month which in the end adds up to the same amount you would pay for the line rental on a monthly plan.
It used to be very expensive to make calls on prepaid mobiles in the UK, like 50 pence per minute, compared to 10-15 cents per minute on monthly plans.
Then Virgin came along and introduced a new scheme by which you would be charged a premium rate on the first 15 minutes per month and after that you'd be charged a rate competitive with monthly plans. When you added up the premium you paid on the first 15 minutes, you got to 17.50 pounds which was the cost of the lowest monthly plan at that time. This system was so successful that all the other providers have adopted similar schemes.
As a result, in the UK, today, you pay your monthly line rental even on prepaid plans. The only difference is that you pay it indirectly. And of course, if you don't use your prepaid phone for the entire month, then you don't pay the line rental equivalent either.
The fact however remains that somebody has to pay somehow for the phone number. So, it's no difference in principle when you pay for the inbound number with Skype.
They could of course use premium rate numbers and let callers pay, but most people don't like to call those "expensive numbers" and so you wouldn't get many calls then. Kind of defeats the purpose of having a number for incoming calls in the first place.
... but it can't hurt to point out the dangers when going with something like Skype and thereby giving them support.
What I fear is that vendors of telephone equipment will find themselves in a situation where they feel they have to make a deal with Skype to release Skype speaking versions of their exsiting designs in order not to miss out on the huge market share Skype seem to be building up on their closed network in which they have a captive audience of users.
Not only will Skype extract a tax from those vendors for allowing their devices to speak Skype, but the fact that the vast numbers of users on Skype's network who don't realise they still have a choice, will probably buy those devices and thereby creating a foothold for Skype to hold manufacturers ransom, indulging in such practises as we now hear Intel has been involved in, namely exorting pressure on those vendors not to support anything other than Skype.
As, I said, I hope you are right and they will hit a wall somewhere on the way, but you can't really deny that there are significant dangers.
Your analogy doesn't fit for a variety of reasons...
First of all, AOL do not exercise a monopoly over access to their network as Skype do. For AIM there are several third party clients, both proprietary and open source. Yes, the only client software that is allowed to participate in the Skype network is the Skype client software.
You cannot just hire somebody to write an interface to the Skype network for your company's telephone system in order to let employees use their office phones to make and receive calls through the Skype network. Skype don't allow you to do this. You have to use their software or devices they control from vendors that they cherry pick and tax.
A very significant difference is also that IM is exclusively used through software clients on common platforms such as PCs or PDAs. There is no market for dedicated IM only devices. This is significant because the software clients for accessing IM are all free downloads and there is no investment into a device that will later prevent users from changing service providers.
This is all the more significant when you consider that IM is not anywhere near as important to our daily lives and to doing business as telephony is. Even if AOL did have a monopoly on Internet Messenger client software and services, it would not anywhere near matter as much as ceding control over telephony to a global monopolist.
What we are heading for with Skype is control over access to telephony by a single global monopolist who tells manufacturers of telephone equipment what they are allowed to do, which providers are allowed to be accessed and how much royalties they have to pay to make their phones compatible with the monopolist's ubiquitous network.
Typically I don't respond to people who have no other means to support their opinion than by using vulgar language as you did, but since I feel other readers will benefit from my dealing with some aspects of your post, I will make an exception.
Now, first of all, let's look at the name and meaning of Skype.
There are at least two Skypes, just like there are two Fireflies.
One meaning of Skype is the client software that you install on your PC (or Mac).
Likewise, there is a client software called Firefly.
Another meaning of Skype is the Skype network that the Skype client software uses to talk to other Skype clients.
Likewise, Firefly, the client software can use Firefly, the network to connect to other clients.
There is thus little point in complaining about the Firefly network having the same name as the Firefly client, because this is exactly the same way it is with Skype.
You say that downloading Skype (meaning downloading the Skype client software) and using it was easy and free. Well, same applies to Firefly. Downloading the Firefly client software and using it is easy and it's free, too.
However, there is an important difference between the Firefly client software and the Skype client software. The Firefly software gives you a choice which network you want to connect to. One of those networks is the Firefly network, but you have a choice not to use it. With the Skype client software on the other hand you get no choice. The Skype software is deliberately made so you can only use it with the Skype network.
Skype, the company, is trying to coerce us all to speak Klingon and nothing but Klingon, and only through one of their own Klingon interpreters, so that we won't be able to communicate on our own without their control. On the other hand, the people who made Firefly have recognised that there are common languages most people already speak: English and Spanish and they don't get to control who we can talk to.
The people who made X-Lite also happen to let us talk English to others (but not Spanish) using their client software. As a result, we can use the Firefly software to talk to somebody who uses the X-lite software and we can do so peer to peer without any involvement of any third party or we can do so using any service provider we choose to.
The same applies to just about everybody else who makes software or devices for internet telephony.
There is nothing wrong with a proprietary protocol per se in order to solve a particular problem. What is bad, however is forcing people into a lock in where they have no choice but to use just one company's lilcensed products and services.
Take Cisco as an example. They do have their own proprietary protocol. It's called Skinny Client Control Protocol and the acronym is SCCP but it's more commonly called "Skinny". As the full name suggests, this is a special lightweight protocol for connecting telephone clients to servers.
Only Cisco have Skinny support on their IP phones, but you can change the setup and use SIP instead. So you are not stuck, you still have a choice. This is OK.
Another example of a new protocol that did not follow established standards is the Inter Asterisk eXchange protocol, or IAX in short. Again, this is a special purpose protocol, designed to hook up two telephone systems over the internet. In this case the designer of the protocol decided not to keep the protocol specs for himself, like Cisco did with Skinny. Instead he published the specs and released an open source library for the protocol under the GPL so that everybody who wanted to could take the protocol and use it. Again, this is OK.
This is a great example of a protocol that started life as a proprietary novelty given into the public domain and getting widely adopted. IAX has become very popular, not only because it is open and there's an open source library, but also because it is very compact, very efficient, extremely resilient against bad network condition
> I'm saying that Skype joining the interop means Skype opening their APIs, and their protocols
It doesn' mean that at all.
Skype have not opened their protocol to anybody, nor are they likely to do so in the future, unless forced as a result of some anti-trust action.
What they have done fits perfectly well into their strategy of keeping their protocol closed. What they are doing is control access to the gateway into the POTS network. They can only control that if they keep their protocol closed. This is why they do it.
They lure people with free p2p calls, but in return those people give up their choice of service providers. The only service provider for going in and out to and from Skype's p2p network is Skype and will remain Skype. People will not be able to choose another provider. There is no competition.
Skype's strategy will be to get more and more phone equipment manufacturers to make Skype devices (of course they have to pay Skype a tax for that) and thereby locking in end users even more because once you have spend 50, 100 or 200 bucks on that Skype device, you are unlikely to try any other provider since the device will only work with Skype and you would have to buy yet another device to change providers.
Skype offering POTS gateway service is the very reason why they keep their protocol closed. It is not a sign of opening up.
There should be a nice page explaining why Skype is bad (proprietary), and the advantage of using open, inter-operable solutions (SIP).
There is quite a lot of information on open VOIP standards. The benefits and possibilities of open standards are so manyfold, it wouldn't fit on a single page. However, the VOIP Wiki is a good source of information...
http://www.voip-info.org
BTW, there are several standard protocols, not just SIP. There is H.323 which is widely used as a protocol between carriers. There is MGCP, which is also used for interconnects between carriers but also for connecting customer premises equipment.
SIP was meant to consolidate but it has become somewhat bloated and it is mostly used for connecting customer premises equipment. One of the cool things about SIP is that it uses plain text, like HTTP and thus it is much easier to read even without special debugging tools.
All of these VOIP protocols are troublesome in NAT and firewall situations. For carrier interconnects this doesn't matter much because carriers will often set up tunnels between them anyway. However, with SIP being predominantly used for delivery to the end user, this hurts. There are several work arounds for NAT traversal and all of them have trade offs, mostly security trade offs.
Skype and Apple have taken SIP and customised them so that those work arounds are permanently bolted on. That's why Skype can only talk to Skype and iChat can only talk to iChat. They are both based on SIP but incompatible. But worse than being incompatible is the fact that both Skype and Apple keep their communications protocols secret. It's one thing to close APIs, it is quite another to do so with comms protocols.
It would have been better to solve the NAT problem by creating a new open protocol, designed with NAT traversal right from the start, avoiding the need for work arounds.
Well, I said "would have been" but actually somebody did just that and the protocol is called IAX, designed by Mark Spencer of Digium, who also released an IAX library as open source. IAX was meant to be a protocol to connect PBXes over the internet, but it has become popular as an alternative to SIP. Various softphones support IAX and at least one chip manufacturer has implemented IAX into their chipsets and many IP phone manufacturers have released hardware using those chips supporting IAX. Many VOIP providers also support IAX now.
The benefit is the same as with all other open standards technologies. You can choose amongst devices from different vendors who do not have to beg to be allowed to use the standards, nor have to pay a tax to use the standards. Likewise, you can choose amongst services from different service providers. Your investment into any particular device is protected. You can take your device and change your service provider.
With Skype, all this is in jeopardy. If they get a certain critical mass and by virtue of market share coerce vendors to make Skype devices paying the Skype tax, eventually to only make Skype devices, then there is the danger of another Microsoft, this time in the telephone space. If you thought the AT&T monopoly was bad, you ain't seen nothing yet.
I am talking about the fact that they are using a closed communications protocol in order to lock customers into Skype only devices so they cannot easily change providers.
This is like the Palladium chip that would make sure that only the big guys who licensed the technology would be able to sell meaningful software.
What would you say if TCP/IP was owned by some corporation and everybody who wanted to use it needed to buy hardware from OEMs who paid the TCP/IP tax, anybody else was locked out?
I repeat, communications protocols must be open, not closed. It's the very nature of communications that it needs open protocols. APIs are a different matter. This one is not about APIs.
Then you can run X-Lite on OSX under MoL to talk SIP to Asterisk (on the same Mac) which then talks IAX to the rest of the world.
In your case, you could run Asterisk either on Linux or on OSX, doesn't really matter. If you run it on OSX, there are native setup assistants, since you said you don't like editing config files.
You are mistaken. The fact that Skype now offers you POTS phone numbers for rental which they feed into their proprietary network does not mean that they are any more interoperable than before.
It sure matters that the protocols are different as long as Skype control those protocols which they do. This means that for any interconnect with POTS, whether inbound or outbound, there is only one place you can go and that is Skype. A classic lock in.
Also, we are not talking APIs here, we are talking communications protocols. To use a networking analogy, what Skype are using is something like the equivalent of Token Ring while anybody else has long settled for Ethernet.
It's one thing to come up with a proprietary API, but it is quite another to come up with a closed comms protocol. Closed comms protocols are just evil. Period.
I said use the Firefly SOFTPHONE, not their service.
Firefly (the softphone) supports both SIP and IAX, which give you access to hundreds of VOIP providers and devices. It doesn't get more multi-platform than that. Using the Skype client you have only one provider you can use, Skype.
IAX is preferable in a NAT situation because it doesn't compromise the security of your NAT firewall. SIP and its proprietary derivatives (ie Skype and Apple's iChat) do NAT traversal by punching holes into your firewall and other such work arounds that sacrifice security. IAX doesn't, it only uses a single UDP port. You call out on that port and you receive calls back in on that port. Signalling and video also go in and out on the same port.
ILBC is a free (as in free beer) codec at present, but that's only free as in "free lunch", just like Skype is free as in "free lunch". Anybody with a little reasoning knows that there ain't no such thing as a free lunch aka TANSTAAFL. Once a market dominance is established, those lunches will become expensive and at that point you won't have anywhwere else left to go.
Besides, ILBC is available in any OSS solution, too. The difference is that you are not stuck with just one single codec. IP phones are designed in such a way that they negotiate codecs when the connection between them is established. If you have a phone that can only speak one single codec, then you are limited to talking to devices which have ILBC, so if somebody has a phone that doesn't speak ILBC, you won't be able to talk to them. More importantly, if you are talking over a connection with characteristics that make ILBC less favourable, - for example, ILBC doesn't cope so well with packetloss - then you would get better quality if the phones can negotiate a more suitable codec. With Skype, they can't.
As far as FWD goes, you do get POTS service through LibreTel directly via your FWD account. Moreover, since you are using a standard IP phone with FWD, you can use any of several hundred VOIP providers who support SIP or IAX. Most IP phones can be configured to use multiple providers without reconfiguration. For example, with Firefly (the softphone) you could tell the phone to use FWD for "internal" calls and NuFone, or Voicepulse or VoipJet for "external" calls. The fact that you are using a device which supports open standards means that you are free to take your business anywhere you like. No lock in.
The FWD interconnects are listed at
http://www.freeworlddialup.com/advanced/peering_ nu mbers
FWD also have a new service called FWDout. This allows you to make free POTS calls in return for letting other FWDout users make local calls on your phone line (you determine the limits).
Finally, If there is any "cobbled together frankensolution" as you have put it, then it is Skype, both in terms of the hacked together technology they use and in terms of the final outcome.
If you want to use the NAT friendly IAX protocol, you can use a cross platform softphone called iaxComm, it's somewhere on sourceforge. Granted, it doesn't look as stylish as Skype or Firefly but it is as easy to use and it works just fine.
If you want a more fancy softphone, but still don't want to struggle with NAT traversal, you can download Asterisk for OSX and run X-Lite together with Asterisk on your Mac.
X-Lite will talk SIP to Asterisk and Asterisk will talk IAX to the internet.
This may sound complicated but is actually straight forward. There is a wiki that shows you how to do this at
http://www.voip-info.org/tiki-index.php?page=Loc al host%20gateway
I didn't think that Skype runs on LinuxPPC, but if you say so...
In any event, there are quite a number of softphones for Linux, and there are quite a few for OSX, too.
For easiest NAT traversal, you can use a phone that supports IAX. Such phones are available for Linux, OSX and Windows. Many VOIP providers now support IAX, including the free FWD network.
The most user friendly IAX softphone is probably Firefly (currently only for Windows, Linux version is under development). It's just as easy as the Skype client and it supports both SIP and IAX.
On your Linux Mac, you'd have to use iaxComm, which is open source and easy to use but doesn't look as nice as Skype or Firefly.
In any event, you'll get something that's just as easy to install and use as Skype but based on open standards, which means you can also interconnect with devices and services that don't play with Skype.
FWD aka Free World Dialup is a free VOIP network with access numbers for inbound access in various countries. Goto http://www.freeworlddialup.com for more details.
Goto http://www.voip-info.org and search the site for VOIP providers. There are hundreds of them. All offer termination (what Skype call SkypeOut) and many offer origination (what Skype call SkypeIn).
Agreed, Skype aim to lure us into lock-in with their "free lunch". Their closed and proprietary ways are reason for concern.
Yet there is nothing Skype does that OSS doesn't match or beat.
Ease of use? Take a look at the Firefly softphone which supports both SIP and IAX2. It's just as easy as Skype.
P2P? Take a look at E164.org or DUNDi. Those don't require an organisation that finally calls the shots and can hold us all for ransom one day.
NAT Traversal? Skype's protocol has to use the very same bad hacks SIP needs to do NAT traversal. Not really a big surprise because it's a derivative of SIP. Take a look at IAX. This is a protocol that is NAT friendly by design. It doesn't need work arounds to travers NATs or firwalls and it's open source.
Codec? Skype use the ILBC codec which is widely available in other proprietary and OSS solutions. Then again, take a look at Speex. It's at least as good as ILBC if not better and it's open source.
Then look at interconnects and you'll find that Skype is not such a nice netizen as they try to have us believe. For example, can you call a Vonage subscriber from Skype? Can you call a Skype user from Vonage? Replace Vonage with a variety of other VOIP networks and ask the same questions again. You will find that there are no interconnects. Skype want it all.
Take a look at FWD. You can call a whole bunch of subscribers of other VOIP networks and vice versa. Many VOIP networks, commercial or not, have interconnects with each other. Skype don't.
Besides, their gateway service to POTS is pretty bad. Most other VOIP services deliver much better quality.
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Vonage have an optional service that allows you to connect to your Vonage account using a SIP phone (X-Lite). This costs $15 per month extra and is only available *in addition* to the basic service that comes through their set top box. However, you can configure Asterisk to be a SIP client on that Vonage option.
Another way to use Asterisk with Vonage is to connect an analog FXO port on your Asterisk server to the phone jack on your Vonage set top box. Not exactly very elegant but it works.
Actually, a Gartner study said that the last PSTN phone call on the planet will be made some time in the year 2020.
Considering that phone companies themselves are getting into VOIP and it is far more cost effective to deliver voice services through a general purpose packet network than through a special purpose circuit network, you can bet that VOIP will eventually take over all telephony.
Gartner may be off by a few years, but they are unlikely wrong in their assessment that circuit switched telephone networks will completely disappear because they will.
VOIP might be the keyword.
I know that Yahoo were the first to start VOIP in Japan and they have established market leadership dwarfing all other VOIP services there, it's called Yahoo BB Phone.
Depending on whom you believe it's got either over 1 million or over 3 million paying customers [recent articles returned by google bring up both these numbers, so I don't know which one is correct].
The way this works is you get a set top box from Yahoo which plugs into your DSL and your phone line. Your telephone also plugs in to the box.
Then if somebody calls you on your landline the call will go to your phone as usual. But if somebody who is also a Yahoo customer calls you using your existing phone number, then the call goes VOIP over Yahoo's network and it is free of charge.
If you call somebody who is not a Yahoo customer, you have the choice to use your landline or going through Yahoo's VOIP service at lower tariffs.
Very much like Vonage but it seems to be much better integrated with your existing landline and phone number.
Maybe Yahoo should consider a similar service in the US and other countries. Who knows, maybe they're already working on it.
that's not entirely correct and probably a misconception on your part.
I think you will find that any provider in Italy from whom you are getting a phone number for inbound calls will charge you some form of line rental.
Those who don't charge a recurring fee will generally not provide you with a phone number for inbound calls, they only provide outbound service. An I right or not?!
The only two exceptions are
- personal or premium rate numbers, where the caller pays a higher rate
- prepaid mobile plans, where you pay more on your outgoing calls than you would if you had a monthly plan
In the UK, prepaid mobile plans now make you pay the monthly line rental indirectly by charging more for the first so many minutes per day or per month which in the end adds up to the same amount you would pay for the line rental on a monthly plan.
It used to be very expensive to make calls on prepaid mobiles in the UK, like 50 pence per minute, compared to 10-15 cents per minute on monthly plans.
Then Virgin came along and introduced a new scheme by which you would be charged a premium rate on the first 15 minutes per month and after that you'd be charged a rate competitive with monthly plans. When you added up the premium you paid on the first 15 minutes, you got to 17.50 pounds which was the cost of the lowest monthly plan at that time. This system was so successful that all the other providers have adopted similar schemes.
As a result, in the UK, today, you pay your monthly line rental even on prepaid plans. The only difference is that you pay it indirectly. And of course, if you don't use your prepaid phone for the entire month, then you don't pay the line rental equivalent either.
The fact however remains that somebody has to pay somehow for the phone number. So, it's no difference in principle when you pay for the inbound number with Skype.
They could of course use premium rate numbers and let callers pay, but most people don't like to call those "expensive numbers" and so you wouldn't get many calls then. Kind of defeats the purpose of having a number for incoming calls in the first place.
... but it can't hurt to point out the dangers when going with something like Skype and thereby giving them support.
What I fear is that vendors of telephone equipment will find themselves in a situation where they feel they have to make a deal with Skype to release Skype speaking versions of their exsiting designs in order not to miss out on the huge market share Skype seem to be building up on their closed network in which they have a captive audience of users.
Not only will Skype extract a tax from those vendors for allowing their devices to speak Skype, but the fact that the vast numbers of users on Skype's network who don't realise they still have a choice, will probably buy those devices and thereby creating a foothold for Skype to hold manufacturers ransom, indulging in such practises as we now hear Intel has been involved in, namely exorting pressure on those vendors not to support anything other than Skype.
As, I said, I hope you are right and they will hit a wall somewhere on the way, but you can't really deny that there are significant dangers.
Your analogy doesn't fit for a variety of reasons ...
First of all, AOL do not exercise a monopoly over access to their network as Skype do. For AIM there are several third party clients, both proprietary and open source. Yes, the only client software that is allowed to participate in the Skype network is the Skype client software.
You cannot just hire somebody to write an interface to the Skype network for your company's telephone system in order to let employees use their office phones to make and receive calls through the Skype network. Skype don't allow you to do this. You have to use their software or devices they control from vendors that they cherry pick and tax.
A very significant difference is also that IM is exclusively used through software clients on common platforms such as PCs or PDAs. There is no market for dedicated IM only devices. This is significant because the software clients for accessing IM are all free downloads and there is no investment into a device that will later prevent users from changing service providers.
This is all the more significant when you consider that IM is not anywhere near as important to our daily lives and to doing business as telephony is. Even if AOL did have a monopoly on Internet Messenger client software and services, it would not anywhere near matter as much as ceding control over telephony to a global monopolist.
What we are heading for with Skype is control over access to telephony by a single global monopolist who tells manufacturers of telephone equipment what they are allowed to do, which providers are allowed to be accessed and how much royalties they have to pay to make their phones compatible with the monopolist's ubiquitous network.
There is no such danger in the IM space.
my apologies for the abbreviations, some of which at least I thought would be commonly known to /. readers, for example NAT or VOIP.
anyway, please see below for an explanation of the abbreviations I used ...
link to my other postTypically I don't respond to people who have no other means to support their opinion than by using vulgar language as you did, but since I feel other readers will benefit from my dealing with some aspects of your post, I will make an exception.
Now, first of all, let's look at the name and meaning of Skype.
There are at least two Skypes, just like there are two Fireflies.
One meaning of Skype is the client software that you install on your PC (or Mac).
Likewise, there is a client software called Firefly.
Another meaning of Skype is the Skype network that the Skype client software uses to talk to other Skype clients.
Likewise, Firefly, the client software can use Firefly, the network to connect to other clients.
There is thus little point in complaining about the Firefly network having the same name as the Firefly client, because this is exactly the same way it is with Skype.
You say that downloading Skype (meaning downloading the Skype client software) and using it was easy and free. Well, same applies to Firefly. Downloading the Firefly client software and using it is easy and it's free, too.
However, there is an important difference between the Firefly client software and the Skype client software. The Firefly software gives you a choice which network you want to connect to. One of those networks is the Firefly network, but you have a choice not to use it. With the Skype client software on the other hand you get no choice. The Skype software is deliberately made so you can only use it with the Skype network.
Skype, the company, is trying to coerce us all to speak Klingon and nothing but Klingon, and only through one of their own Klingon interpreters, so that we won't be able to communicate on our own without their control. On the other hand, the people who made Firefly have recognised that there are common languages most people already speak: English and Spanish and they don't get to control who we can talk to.
The people who made X-Lite also happen to let us talk English to others (but not Spanish) using their client software. As a result, we can use the Firefly software to talk to somebody who uses the X-lite software and we can do so peer to peer without any involvement of any third party or we can do so using any service provider we choose to.
The same applies to just about everybody else who makes software or devices for internet telephony.
There is nothing wrong with a proprietary protocol per se in order to solve a particular problem. What is bad, however is forcing people into a lock in where they have no choice but to use just one company's lilcensed products and services.
Take Cisco as an example. They do have their own proprietary protocol. It's called Skinny Client Control Protocol and the acronym is SCCP but it's more commonly called "Skinny". As the full name suggests, this is a special lightweight protocol for connecting telephone clients to servers.
Only Cisco have Skinny support on their IP phones, but you can change the setup and use SIP instead. So you are not stuck, you still have a choice. This is OK.
Another example of a new protocol that did not follow established standards is the Inter Asterisk eXchange protocol, or IAX in short. Again, this is a special purpose protocol, designed to hook up two telephone systems over the internet. In this case the designer of the protocol decided not to keep the protocol specs for himself, like Cisco did with Skinny. Instead he published the specs and released an open source library for the protocol under the GPL so that everybody who wanted to could take the protocol and use it. Again, this is OK.
This is a great example of a protocol that started life as a proprietary novelty given into the public domain and getting widely adopted. IAX has become very popular, not only because it is open and there's an open source library, but also because it is very compact, very efficient, extremely resilient against bad network condition
That's fair enough but you do pay a monthly line rental for your phone line, don't you?!
Paying for a phone number that passes calls through a gateway service via VOIP to your IP phone is the equivalent of paying the line rental.
> I'm saying that Skype joining the interop means Skype opening their APIs, and their protocols
It doesn' mean that at all.
Skype have not opened their protocol to anybody, nor are they likely to do so in the future, unless forced as a result of some anti-trust action.
What they have done fits perfectly well into their strategy of keeping their protocol closed. What they are doing is control access to the gateway into the POTS network. They can only control that if they keep their protocol closed. This is why they do it.
They lure people with free p2p calls, but in return those people give up their choice of service providers. The only service provider for going in and out to and from Skype's p2p network is Skype and will remain Skype. People will not be able to choose another provider. There is no competition.
Skype's strategy will be to get more and more phone equipment manufacturers to make Skype devices (of course they have to pay Skype a tax for that) and thereby locking in end users even more because once you have spend 50, 100 or 200 bucks on that Skype device, you are unlikely to try any other provider since the device will only work with Skype and you would have to buy yet another device to change providers.
Skype offering POTS gateway service is the very reason why they keep their protocol closed. It is not a sign of opening up.
There is quite a lot of information on open VOIP standards. The benefits and possibilities of open standards are so manyfold, it wouldn't fit on a single page. However, the VOIP Wiki is a good source of information ...
http://www.voip-info.org
BTW, there are several standard protocols, not just SIP. There is H.323 which is widely used as a protocol between carriers. There is MGCP, which is also used for interconnects between carriers but also for connecting customer premises equipment.
SIP was meant to consolidate but it has become somewhat bloated and it is mostly used for connecting customer premises equipment. One of the cool things about SIP is that it uses plain text, like HTTP and thus it is much easier to read even without special debugging tools.
All of these VOIP protocols are troublesome in NAT and firewall situations. For carrier interconnects this doesn't matter much because carriers will often set up tunnels between them anyway. However, with SIP being predominantly used for delivery to the end user, this hurts. There are several work arounds for NAT traversal and all of them have trade offs, mostly security trade offs.
Skype and Apple have taken SIP and customised them so that those work arounds are permanently bolted on. That's why Skype can only talk to Skype and iChat can only talk to iChat. They are both based on SIP but incompatible. But worse than being incompatible is the fact that both Skype and Apple keep their communications protocols secret. It's one thing to close APIs, it is quite another to do so with comms protocols.
It would have been better to solve the NAT problem by creating a new open protocol, designed with NAT traversal right from the start, avoiding the need for work arounds.
Well, I said "would have been" but actually somebody did just that and the protocol is called IAX, designed by Mark Spencer of Digium, who also released an IAX library as open source. IAX was meant to be a protocol to connect PBXes over the internet, but it has become popular as an alternative to SIP. Various softphones support IAX and at least one chip manufacturer has implemented IAX into their chipsets and many IP phone manufacturers have released hardware using those chips supporting IAX. Many VOIP providers also support IAX now.
The benefit is the same as with all other open standards technologies. You can choose amongst devices from different vendors who do not have to beg to be allowed to use the standards, nor have to pay a tax to use the standards. Likewise, you can choose amongst services from different service providers. Your investment into any particular device is protected. You can take your device and change your service provider.
With Skype, all this is in jeopardy. If they get a certain critical mass and by virtue of market share coerce vendors to make Skype devices paying the Skype tax, eventually to only make Skype devices, then there is the danger of another Microsoft, this time in the telephone space. If you thought the AT&T monopoly was bad, you ain't seen nothing yet.
Again, I am not talking about Skype's APIs.
I am talking about the fact that they are using a closed communications protocol in order to lock customers into Skype only devices so they cannot easily change providers.
This is like the Palladium chip that would make sure that only the big guys who licensed the technology would be able to sell meaningful software.
What would you say if TCP/IP was owned by some corporation and everybody who wanted to use it needed to buy hardware from OEMs who paid the TCP/IP tax, anybody else was locked out?
I repeat, communications protocols must be open, not closed. It's the very nature of communications that it needs open protocols. APIs are a different matter. This one is not about APIs.
... why not run Asterisk as a SIP/IAX gateway?
Then you can run X-Lite on OSX under MoL to talk SIP to Asterisk (on the same Mac) which then talks IAX to the rest of the world.
In your case, you could run Asterisk either on Linux or on OSX, doesn't really matter. If you run it on OSX, there are native setup assistants, since you said you don't like editing config files.
Anyway, you can find the SIP/IAX gateway setup at
http://www.voip-info.org/wiki-Localhost+Gateway
You are mistaken. The fact that Skype now offers you POTS phone numbers for rental which they feed into their proprietary network does not mean that they are any more interoperable than before.
It sure matters that the protocols are different as long as Skype control those protocols which they do. This means that for any interconnect with POTS, whether inbound or outbound, there is only one place you can go and that is Skype. A classic lock in.
Also, we are not talking APIs here, we are talking communications protocols. To use a networking analogy, what Skype are using is something like the equivalent of Token Ring while anybody else has long settled for Ethernet.
It's one thing to come up with a proprietary API, but it is quite another to come up with a closed comms protocol. Closed comms protocols are just evil. Period.
I said use the Firefly SOFTPHONE, not their service.
_ nu mbers
Firefly (the softphone) supports both SIP and IAX, which give you access to hundreds of VOIP providers and devices. It doesn't get more multi-platform than that. Using the Skype client you have only one provider you can use, Skype.
IAX is preferable in a NAT situation because it doesn't compromise the security of your NAT firewall. SIP and its proprietary derivatives (ie Skype and Apple's iChat) do NAT traversal by punching holes into your firewall and other such work arounds that sacrifice security. IAX doesn't, it only uses a single UDP port. You call out on that port and you receive calls back in on that port. Signalling and video also go in and out on the same port.
ILBC is a free (as in free beer) codec at present, but that's only free as in "free lunch", just like Skype is free as in "free lunch". Anybody with a little reasoning knows that there ain't no such thing as a free lunch aka TANSTAAFL. Once a market dominance is established, those lunches will become expensive and at that point you won't have anywhwere else left to go.
Besides, ILBC is available in any OSS solution, too. The difference is that you are not stuck with just one single codec. IP phones are designed in such a way that they negotiate codecs when the connection between them is established. If you have a phone that can only speak one single codec, then you are limited to talking to devices which have ILBC, so if somebody has a phone that doesn't speak ILBC, you won't be able to talk to them. More importantly, if you are talking over a connection with characteristics that make ILBC less favourable, - for example, ILBC doesn't cope so well with packetloss - then you would get better quality if the phones can negotiate a more suitable codec. With Skype, they can't.
As far as FWD goes, you do get POTS service through LibreTel directly via your FWD account. Moreover, since you are using a standard IP phone with FWD, you can use any of several hundred VOIP providers who support SIP or IAX. Most IP phones can be configured to use multiple providers without reconfiguration. For example, with Firefly (the softphone) you could tell the phone to use FWD for "internal" calls and NuFone, or Voicepulse or VoipJet for "external" calls. The fact that you are using a device which supports open standards means that you are free to take your business anywhere you like. No lock in.
The FWD interconnects are listed at
http://www.freeworlddialup.com/advanced/peering
FWD also have a new service called FWDout. This allows you to make free POTS calls in return for letting other FWDout users make local calls on your phone line (you determine the limits).
Finally, If there is any "cobbled together frankensolution" as you have put it, then it is Skype, both in terms of the hacked together technology they use and in terms of the final outcome.
You do have plenty of choices on the Mac.
c al host%20gateway
If you want to use the NAT friendly IAX protocol, you can use a cross platform softphone called iaxComm, it's somewhere on sourceforge. Granted, it doesn't look as stylish as Skype or Firefly but it is as easy to use and it works just fine.
If you want a more fancy softphone, but still don't want to struggle with NAT traversal, you can download Asterisk for OSX and run X-Lite together with Asterisk on your Mac.
X-Lite will talk SIP to Asterisk and Asterisk will talk IAX to the internet.
This may sound complicated but is actually straight forward. There is a wiki that shows you how to do this at
http://www.voip-info.org/tiki-index.php?page=Lo
Talking between FWD users is free, too.
Moreover, calling from FWD to Vonage is free, calling from FWD to Packet8 is free, etc etc etc
FWD supports SIP and IAX, both open standards. You can use any device from any vendor or software author who support those protocols.
Call quality is better, too.
I didn't think that Skype runs on LinuxPPC, but if you say so ...
In any event, there are quite a number of softphones for Linux, and there are quite a few for OSX, too.
For easiest NAT traversal, you can use a phone that supports IAX. Such phones are available for Linux, OSX and Windows. Many VOIP providers now support IAX, including the free FWD network.
The most user friendly IAX softphone is probably Firefly (currently only for Windows, Linux version is under development). It's just as easy as the Skype client and it supports both SIP and IAX.
On your Linux Mac, you'd have to use iaxComm, which is open source and easy to use but doesn't look as nice as Skype or Firefly.
In any event, you'll get something that's just as easy to install and use as Skype but based on open standards, which means you can also interconnect with devices and services that don't play with Skype.
Get yourself that Sipura and sign up with FWD, it's free, no lock-in and it's based on open standards.
http://www.freeworlddialup.com
FWD aka Free World Dialup is a free VOIP network with access numbers for inbound access in various countries. Goto http://www.freeworlddialup.com for more details.
Goto http://www.voip-info.org and search the site for VOIP providers. There are hundreds of them. All offer termination (what Skype call SkypeOut) and many offer origination (what Skype call SkypeIn).
Agreed, Skype aim to lure us into lock-in with their "free lunch". Their closed and proprietary ways are reason for concern.
Yet there is nothing Skype does that OSS doesn't match or beat.
Ease of use? Take a look at the Firefly softphone which supports both SIP and IAX2. It's just as easy as Skype.
P2P? Take a look at E164.org or DUNDi. Those don't require an organisation that finally calls the shots and can hold us all for ransom one day.
NAT Traversal? Skype's protocol has to use the very same bad hacks SIP needs to do NAT traversal. Not really a big surprise because it's a derivative of SIP. Take a look at IAX. This is a protocol that is NAT friendly by design. It doesn't need work arounds to travers NATs or firwalls and it's open source.
Codec? Skype use the ILBC codec which is widely available in other proprietary and OSS solutions. Then again, take a look at Speex. It's at least as good as ILBC if not better and it's open source.
Then look at interconnects and you'll find that Skype is not such a nice netizen as they try to have us believe. For example, can you call a Vonage subscriber from Skype? Can you call a Skype user from Vonage? Replace Vonage with a variety of other VOIP networks and ask the same questions again. You will find that there are no interconnects. Skype want it all.
Take a look at FWD. You can call a whole bunch of subscribers of other VOIP networks and vice versa. Many VOIP networks, commercial or not, have interconnects with each other. Skype don't.
Besides, their gateway service to POTS is pretty bad. Most other VOIP services deliver much better quality.