Voice Over IP Goes Global, The DNS Way

awehttam writes "A couple of geeks have setup a non-profit public DNS root designed to map phone numbers to Internet protocols. These days we're hearing lots about Skype, and Voice over IP. Asterisk - the open source PBX - is nearing its version 1.00 release, Free World Dialup has applied to run the .tel top level domain, Good old Bell's are migrating to native IP, private sector layer 2 clearing houses are exchanging bits between companies the like of Packet8, China Telecom, MIT and Harvard and even the various regulatory agencies are pondering just what to do about things. In the mean time, consumer SIP phones are dropping in price, and free and open source software is helping to drive a new generation of provide the services networks." Read on for more.

"You just knew the other shoe had to drop. E164.org let's people register their existing phone numbers, and aim various services including VoIP towards a URL on the Internet. Now you can have your calls sent to your Free World Dialup account, or routed to your home Asterisk PBX instead, possibly where you have a $20 card attached to your phone line letting you make and receive calls through both your regular phone line and the Internet. E164.org isn't just about VoIP though, it can also map phone numbers to Email addresses, Instant Messager URL's, or any other protocol that fits in the "foo://bar" scheme of the 'net. :)"

Comments

End of Long distance calls?

[NeurAlien6]

could this be the end of long distance charges?

Re:End of Long distance calls?

[Tarpan]

Slashdot needs a "+1, Hopefully" ;)

Re:End of Long distance calls?

[PeterPumpkin]

The phone and cable companies could always start charging by the hour for internet access to specific technogies/ports/whatever (such as whatever VoIP uses). Then we'd effectively have long distance charges for every call, next door or Thimbuktu

Re:End of Long distance calls?

[Anonymous Coward]

could this be the end of long distance charges?

If you're still paying long distance charges you seriously need to switch carriers. I just use my cell phone and pay nothing for long distance or roaming.

Re:End of Long distance calls?

[azuretek]

then I would just route it through port 80... it's not to tough to beat the system...

and anyway if my cable company started charging by port I would surely find a different provider

Re:End of Long distance calls?

[peter hoffman]

Unless your cell phone bill is $0.00 you are actually paying for those calls. Your provider has simply determined they can charge you a flat rate at your usage level and still make a profit.

Re:End of Long distance calls?

[NanoGator]

"could this be the end of long distance charges? "

Back in 1999 I bought a cell phone that ended my long distance charges. I still pay for the phone (kinda like how you still pay for an ISP), but I can call from anywhere in the USA to anywhere in the USA. It could be the end to other stuff, but if you really really hate paying long distance charges....

Re:End of Long distance calls?

[ShadowRage]

for a short while, then as the major telecoms get more involved, they get the fcc in to start making long distance charges for voip, or charging long distance for their own services. and paying the fcc to make it mandatory, etc..

ah, too late!

[Mad_Rain]

Does this mean that just when I figured out how to whistle at 2600 hertz, it's become useless? ;)

Re:ah, too late!

[StuWho]

Not if you can do it through your ass!

TeamSpeak

[Seek_1]

Personally, I prefer TeamSpeak to Skype. The interface isn't quite as nice, but for group conversations it works alot better (IMO). Less bandwidth too, which is better if you're using it for VoIP while gaming..

Re:TeamSpeak

[NeurAlien6]

i've been using Skype for quite some time now and i really like it. i don't know about TeamSpeak but i don't find that Skype uses to much of my resources. I just hope they won't start charging.

Re:TeamSpeak

[walt-sjc]

I'm a little baffeled as to why skype was even mentioned. It completely different and not at all compatible with everything else listed. It's not "phone over internet" even though the authors of skype imply that it is. There is no PSTN connection anywhere. VoIP has certain stadards such as SIP, H.323, etc that are designed to interoperate and pass real phone calls over an IP network. Skype is proprietary and only talks to skype. Ditto with Teamspeak. It's not that they're bad, but they are not IP Telephony.

Re:TeamSpeak

[MrWorf]

If I'm not misstaken, TeamSpeak isn't VoIP compliant, nor is Skype (if VoIP = H323).

Skype is more a replacement for your phone (thats why it uses more bandwidth, better sound and quality... also less latency) while teamspeak is more of a voicebased irc, perfect for games, but not something for my parents.

Re:TeamSpeak

[anethema]

I STRONGLY disagree.

While teamspeak is nice for some stuff(channels, lower bandwidth, selectable codec), I have never been able to eliminate lag with any combination of options.

I set it up on my home lan (just 3 computers) and it still had about 3/4 of a second to a second's lag. I also tryed it on a DS3 teamspeak/IL2-FB server, and it still had around a second of lag.

This makes it utterly worthless in my book, no matter its other benifets. End up telling someone which server to join...dont hear a answer and ask if they hear you while you get their response, then they get yours and get mad, while you're getitng mad. God it's annoying. Makes it totally unusable for very fast action fps, where you need your team mate to do something NOW! Not 1 second+reaction time from now.

Also, skype is p2p and encrypted, nice interface, and best of all, VERY low latency. I tryed on my cable connection to my friends cable connection, while also beeing on the phone..lag is maybe 1/10th of a second. That and the nice interface ...plus whatever other boons, make it definatly my top choice for voip software.

If anyone has some special settings to make teamspeak nearly lag free, let me know. I'm pretty sure I tryed it all, I screwed around with it for hours. I couldnt even get good latency on a LAN and thats not at all acceptable. Any help is appreciated :)

Re:TeamSpeak

[Anonymous Coward]

TeamSpeak and Skype are both VoIP (Voice over IP). There are proprietary (TS, Skype, ...) and open (SIP/RTP, H.323, ...) protocols, but VoIP just means you're digitizing speech and sending it over an IP network.

Re:TeamSpeak

[reality-bytes]

There must be *something* wrong with your setup as my server pings 2ms locally and my friend connects from Germany (to UK) at around 25-30ms.

Try using the 7.2 speex codec (good quality) as it requires less CPU and consider using a dedicated box if you typically run high priority processes on your existing one - don't forget, by the nature of TS, it can be having to encode dozens of different ways to cater for multiple clients.

All of my experience with TS is on Linux so I can't comment on whether the Windows TS server would be better or worse.

Re:TeamSpeak

[anethema]

Weird.

I have no idea the actual latency, my numbers are jsut from best guesses for tests ive done on local lans. After writing this I had a friend download it and try it on his lan with a server and two clients. Same results. These arent slow computers, and the lan is 100mbit.

Lag seems to be around a second..or a touch less.

What kind of settings do you use?

I have tryed all codecs, lag doesnt change.

Also, the server for Forgotten battles on the DS3 is frequently used by a good few clans, and the people i played with seemed to think this amount of talking-lag was normal. I dunno, skype works. Almost imperceptable lag. Encrypted stream (tho its not only closed source, there is very little info out there, so i dont really trust it)

Re:TeamSpeak

[reality-bytes]

I don't really know what to suggest as its always worked 'out of the box' for me. The lag is imperceptible to us, sometimes when someone comes on channel with a bit of feedback, we sometimes hear ourselves 3/4 second later but this is due to the 'double round trip' (codec>xfer>server>xfer>codec>codec>xfer>server>xf er>codec) which can give a false impression of lag; normal trip is (codec>xfer>server>xfer>codec).

Incidentals:
Our box runs Linux 2.4.x
Our box is pretty unloaded
Our box only has a 10Mbit NIC
Our box is on the end of a 128kbs/800kbps link.
We use v2.0.20.1 of the TS server
We use the Speex 7.2 Codec

My friend was previously on Skype too but we noticed that they had ominously removed a graphic which said '100% spyware free' from their web-page and that the Skype software made a lengthy transaction with an unknown central server on a couple of occasions. We (tin-foil hats at the ready) suspected skype of sending more than just the audio in its encrypted packet stream.

When it comes to all-round (telephone-like) comms tho, you could take a look at Gnomemeeting (cross platform) - I'm quite partial to it :)

Re:TeamSpeak

[anethema]

Ah, these are all windows servers and clients for teamspeak. Maybe the workings on windows xp arent very good.

Also, while i havent done any packet sniffing at all, skype still has a

"NO spyware/adware" graphic on their website.

Ah well, ill skip with skype for now. I just tryed again with that codec and its still laggy. Ah well, thanks for your help.

Re:TeamSpeak

[curne]

Just a thought: How is the sound played on your system?
Directly to /dev/dsp or through Esound (Gnome) or ARTS (KDE)? The two latter options can add lag from the application to the soundcard, especially esd, is my experience.

Re:TeamSpeak

[anethema]

It's actually a windows box, windows server.

And i dont think any sound wrapper or API would introduce a second of lag.

This is only the beginning

[TheMadPenguin]

Obviously we are in the beginning stages of something very large. Telecom as we know it today will change... it's only a matter of time.

Progress is good but wha

[Tandoori Haggis]

t ab*5t dr*p ou^ts affe$ting cal$ qu^lity?

Re:Progress is good but wha

[Anonymous Coward]

Is that how they create those headers for spam:)

Re:Progress is good but wha

[ReverendRyan]

I know you were joking, but I recently decied to try out Vonage. With a cable connection you cant even tell its using the internet, and the ability to get a real phone number anywhere (well, almost) in the US beats a landline hands down.

After 2 weeks with it, I've decided I'm going to drop my landline.

Re:Progress is good but wha

[Tandoori Haggis]

Thanks for the info. I must admit I'm starting to hear more and more positive stories about VOIP. Its certainly picking up in Europe and I have met engineers who assured me that in practice it can work really well. One pointed out that the quality seems much better than mobile phones and we don't complain about them too often.

Those Tariffs are interesting 2c / minute is seriously cheap. If I was living across the pond I think I could be tempted.

Cheers

Re:Progress is good but wha

[Anonymous Coward]

> After 2 weeks with it, I've decided I'm going to drop my landline.

How's their 911 service during a local power outage, or do you have your home PC on a UPS?..

Re:Progress is good but wha

[The Vulture]

We have Vonage at our office, and while it saves a lot of money on calls (especially to Taiwan, which we call a lot, but we use it for our long distance U.S. calls too), we don't feel that it's good enough to drop out landline.

Don't get me wrong, I think Vonage is great, and I really like the service - just that sometimes it will break up, or give us strange echos, or other things.

The Vonage MTA is connected straight to our DSL router (I don't remember the brand), on our SBC DSL line. There's no firewalls or NAT that it's going through.

Despite the glitches (which don't happen very often, but always seem to happen at the wrong times), it's a pretty good service. But not good enough to replace a primary landline.

-- Joe

Re:Progress is good but wha

[jrockway]

Even if you don't pay for phone service, the phone company has to provide 911. So when the power is out just use that phone that's still plugged in for the purpose of 911. Isn't government regulation great :)

Re:Progress is good but wha

[jrockway]

Your DSL router is probably NAT.

I would recommend setting up a Linux (or OpenBSD) box with three ethernet cards such that eth0 is your internal network, eth1 is your vonage box, and eth2 is the WAN. Then configure iptables/pf to drop packets on the eth0eth2 route when the eth1eth2 route is "full". This is called QoS, I'm sure there are some nice HOWTOs that will help with this.

Re:Progress is good but wha

[The Vulture]

The DSL router is not running NAT. NAT was turned off in the router explicitly for two reasons:
1. We have five static IPs (we're using business class DSL), and the MTA is configured with one of those static IPs.
2. If I was using NAT with the MTA (a Motorola device), then I'd have to forward a few ports (the manual for the Motorola MTA mentions this).

NAT is running on the firewall, which has one of the five static IP addresses. The corporate LAN uses this. Although the MTA supports a LAN behind it (using NAT), there is nothing connected to the MTA (there is no need).

The firewall (which is not attached to the MTA in any way) is a Linux PC using IPCop.

-- Joe

Link collection great, but more info?

[Jonah Hex]

Thanks, I've already linked most of those sites and the only useful info I get is that there's a cheaper Asterisk compatible card out now. :( How about a bit more info on how these different methods actually stack up against each other? Maybe some success stories?

Jonah Hex

Re:Link collection great, but more info?

[scottj]

I had never even heard of Asterisk, so this link collection was quite valuable to me. And that $16.99 FXO card that is compatible with Asterisk is extremely interesting. It looks like I may need to look into gettin an Asterisk system up at the house in the next few weeks.

Re:Link collection great, but more info?

[walt-sjc]

The reason you buy the real Digium FXO cards is because it's a great way to support the company that provides most of the coding for Asterisk. You will also get support which can be very valuable when playing with something as complicated as Asterisk for the first time.

Re:Link collection great, but more info?

[tmasssey]

Two things about this:

1) Buying the Asterisk-compatible card does not help out the company who has done 95% of the development, both hardware and software, that makes that board do something: Digium.

2) Digium has announced an FXO module for the TDM400 board that replaces the X100P. In other words, you can add up to 12 FXO (talks to telco) or FXS (talks to telephones) interfaces in the same computer, instead of just a couple.

There's nothing immoral about buying the off-brand X100P's, but it doesn't help you to get the next version of the X100P developed, or the software to make it work...

Re:Link collection great, but more info?

[Anonymous Coward]

First there was the POTS (Plain Old Telephone System). Now there are VoIP (Voice over IP) phones and PBXs (Public Branch eXchange). VoIP comes in several flavors: SIP (Session Initiation Protocol) with RTP (Real-time Transport Protocol) is the most common today, but there are others (H.323, IAX1/2, ...).

One of the problems with VoIP is the user interface: How do you tell the system whom you want to talk to? Phones usually have a very limited UI: numbers, asterisk and pound sign. SIP URLs on the other hand contain dots, characters, at-signs and numbers. To map one onto the other, ENUM was devised. ENUM maps phone numbers to service URLs. Not just SIP urls, but also email-, web- and instant messaging URLs. So, once ENUM is operational, you can "send email to a phone number", and you can place a VoIP call to a regular phone number. User interface problem solved.

Asterisk is an Open Source PBX running on Linux which is developed by Digium, makers of telephony interface hardware. There are interface (channel) drivers for other hardware, too, including CAPI ISDN cards, HFC-S based ISDN cards, ALSA soundcards and more.

The systems work, albeit chaotically documented, so the next step is to connect them. Right now you have to look up what prefix to dial when you are on one (VoIP) network and want to dial into another network. The networks are making their prefixes up as they go (cherry-picking: FWD mostly uses 393, F~3,W~9,D~3), they allocate numbers differently and they use different ways of indicating whether you're going to call a POTS number, a network-internal number or a number on a peering network. This is a considerable drawback compared to the POTS. All of this could be solved by a registry of number to VoIP url mappings, in which providers could have their numbers allocated. There's obviously a huge financial opportunity here. You're looking at the .com for telephones, so reaching consensus won't be easy. In fact, the ENUM RFC is from 2000. That should give you an idea of the controversies. e164.org is the ENUM equivalent of the alternate DNS roots: The official domain for ENUM, under which POTS companies will register their number spaces, is e164.arpa.

Re:Link collection great, but more info?

[Anonymous Coward]

FYI, The guy who did most(if not all) of the early work on the wildcard(previously known as the Tormenta card) does not work for Digium.

Re:Link collection great, but more info?

[Tony Hoyle]

The FX100 module is a rebadged Intel Winmodem worth about $9.99. They're taking the p*ss wanting $100 for it.

Even the 'cheap' ones are expensive when you realize what they actually are... Altruism only gets you so far. If they wanted $30 I'd buy from them, but $100? No way.

Re:Link collection great, but more info?

[n8twj]

You are supporting the primary developer of Asterisk buy buying the hardware from them. After all they outright gave the world an absolutely amazing piece of software to begin with.

Digit networks is not going to support you. They are in this for a quick buck.

Re:Link collection great, but more info?

[n8twj]

No, but Jim Dixon is a personal friend of Mark Spencer. Then Mark took it upon himself to improove on the design and also build a single port version.

Why shouldn't Mark be able to profit off of his work?

Re:Link collection great, but more info?

[tmasssey]

I do not claim to know the intimate history of the Tormenta boards, their evolution into the Wildcard, etc. I do know, however, that the 4-port T1 or E1 cards that you're talking about are now several versions old. In fact, they've been completely replaced with new busmastering PCI cards. Details here (Warning: PDF).

Also, the Wildcard X100P is totally unrelated to the Tormenta cards you mention.

Finally, without the Asterisk software, most definitely written in vast majority by Digium, those Wildcards don't exactly do much...

Re:Link collection great, but more info?

[rootnl]

At this moment I have an Asterisk PBX running at home. I am using an AVM Fritz ISDN adapter with capi drivers in order to dial out to a landline.

The biggest challange with VOIP is getting your current telephones connected. Using Softphones or VOIP hardphones is an solution but costs money.

For me the challange was obtaining an ISDN card with a HFC type chipset. These are some the few chipsets known to enable an ISDN card into NT mode which is required for connecting TA's (telephones). These type of cards are still cheaply available and I recently bought one at ebay for 2 euro's.

Slashdot...

[Anonymous Coward]

Please, oh please, it would be son nice if you let us know when you link to a pdf.

Re:Slashdot...

[dave1g]

may I ask why?

Not that I dont have my own gripes about acrobat, mainly the fact that it takes forever to load up the full version or even just the plugin. And I have an Athlon XP 3200+!

Even longer since I switched to Firefox, though that might just be my imaginatin.

Re:Slashdot...

[iminplaya]

may I ask why? ...mainly the fact that it takes forever to load up the full version or even just the plugin.

You just answered your own question, and that's why I'd like a heads up. Then I can download it for my perusal at a more convenient time. When I'm expecting a pdf, I don't mind the wait so much. Otherwise it looks like my machine locked up. There's no warning on how big the file is, and the time waiting for it to download(especially on dail-up) can be very distressing. I don't always catch the info in the status bar. And focusing on the tiny print there can be painful to the old eyes.

Re:Slashdot...

[dave1g]

Oh ok, I thought maybe there was another reason that I wasn't aware of.

On this point I agree with you, infact my computer did lock up for more than a few seconds when I opened that pdf.

Is that adobe's fault or what? Seems like they have a knack for bloatware...

Re:Slashdot...

[iminplaya]

That and I believe the machine puts itself on "hold" while the file is downloading. A 1 MB file can hold you up for over five minutes on dial-up. I just need to be more careful, and check the status bar. Won't be a problem then. :-)

Re:Slashdot...

[Anonymous Coward]

Sorry for sounding trite but... it looks like the problem is with your browser and not slashdot. First, I always try to check the linked URL before clicking on it, just be safe. If your browser or your setup doesn't let you do that easily, something is wrong. Second, even if I follow the link without checking, my browser doesn't lock up at all and soon after connecting prompts me for a choice to save or open the pdf file. That gives me a chance to continue or abort.

Re:Slashdot...

[Anonymous Coward]

It's all good. I just need a new set of eyes :-)

Skype to POTS idea

[dave1g]

I was thinking about this the earlier today, and since it is relevant to this story I thought I would post it.

Currently Skype is a computer to computer only program, but I have an idea to make it somewhat better.

At signup each user should put in his local calling area/country codes whatever they all are.

Then each user has the option of donating his phone line by hooking up his modem to the phone cord.

Whenever someone wants to call a phone number they could try Skype, or a similar program, and the Skype network would then go search for any available modems in that area code. If there is one then your call would be sent over the net to that computer, and out on his modem.

Now obviously this is a pretty generous donation on the part of the users. There fore there would have to be controls given to the user such as only allow people to call through your line if they are sharing their line. And there needs to be a polite "get off my phone I need it" button for when a Skype user is using your phone and you need it. Something like a message interrupting the call saying "the host modem owner want to use his phone line, please end your call in 30 seconds or it will be disconnected for you."

And if there are no modems available in the area code you wish, you will get a message telling you so.

I think this is would be a nice way of getting around long distance even to people who don't own a computer and/or use Skype.

Are there any technical problems with routing audio info taken from the internet into the modem in the computer without the host having to listen to it over his speakers?

What do you guys think?

Re:Skype to POTS idea

[i_am_pi]

Great idea, but one flaw.

modem dials
person on the other end picks up
"EEEEEEEEEEEEEEEEEEEERRRRRRRR PTSSSSSSSSSSSS"

You'd need a PBX with a FXO card to transmit voice over a regular phone line.

Re:Skype to POTS idea

[digitalvengeance]

Privacy is a big issue here. What's to stop me from hooking a $30 voice activated phone recorder to the line and record all of your conversations? I don't know how much I trust someone I've never met and who has no real oversight to protect my privacy. Scratch that.. I do know that I don't trust them at all.

Re:Skype to POTS idea

[MajorDick]

Oh telemarketers would love this then they could deluge people all around the world with their mazaine subscriptions.

Re:Skype to POTS idea

[dave1g]

well obviously some encryption could be used until the last possible second before offloading it to the modem.

If that turns out not to be possible, you wouldnt use this for a call that required privacy. If that is the case it sounds like paying long distance would be worth it for the privacy that you want. therefore you wouldnt even be in the market for this use.

Re:Skype to POTS idea

[Realistic_Dragon]

Might work in the US but here in the UK we have to pay (rather a lot) for local calls.

I'm not going to pay $0.08/min (peak) for someone in the US to talk to someone down the road.

And that's assuming the software is infalliable. Immagine if it broke and you started placing calls to Azejabstan for £2.50 a minute? Given the average buggyness of most software combined with the general attractions of phone systems to phreakers I would expect a call charge after the first month that could be accidentally confused with a Chinese phone number, international code and all.

Re:Skype to POTS idea

[gnu-generation-one]

"Privacy is a big issue here. What's to stop me from hooking a $30 voice activated phone recorder to the line and record all of your conversations?"

F.F.S.! People still use email for conversations. You know, where you send plaintext to your ISP, through the mailservers of a load of people you don't know, to the mailserver of someone you don't trust.

And we're worried about privacy of phones? (hint: text-searching is easier on peoples' emails)

No, encryption isn't used. I've never once received a PGP-signed or encrypted email from someone that I didn't specifically set-up their email programs myself.

If you want to chat securely on the internet, download PGPFone. It's obsolete, unsupported, and non-free, so we need a good replacement. If anyone knows a good replacement, just email it to yourself, I'll be listening.

Re:Skype to POTS idea

[dave1g]

um... I can transmit voice over my phone line with my modem by simply using the microphone on my computer. I have done this before, even my old compaq from...1996 had this capability in it's 33.6 modem.

What am I missing here? Instead of having the modem use the mic input it could use an audio signal that your computer took off the internet.

Help me to understand the issue you raised. If it turns out this idea sucks as, well it was worth a shot anyways :-)

Re:Skype to POTS idea

[dave1g]

yeah...I thought of that. I think the client program that everyone runs could easily make sure that the phone number is in their area code and wont have any charges.

You could put a list of the area codes that you can call for free to into the computer and it is pretty easy to make sure only those area codes are used.

I guess it was short sighted of me to think that everyone had free local call...sucks to not be in the US. Companies know that we wont pay for anything by the minute.....except cell phones... and even then you get a block of minutes for a set price before per minute pricing kicks in.

Even if this feature was only limited to the US that is still a large demographic that could make use of the feature.

Re:Skype to POTS idea

[Anonymous Coward]

Interesting idea, but one problem. It costs MORE to call from within one area code to another location in that area code (that is not local) than it does to call from out of state to that same, non-local number. It's called "local long-distance" vs "out of state long-distance", just another way the telcos screw customers.

Re:Skype to POTS idea

[tmasssey]

Two things:

1) The AT standard for voice is half-duplex: it can't listen and play at the same time. That won't work for a PBX.

2) There are plenty of softmodems that can do full-duplex voice. However, their API's are not documented, and therefore are not supported.

An X100P or an FXO module for the TDM400P, in fact, are little different than a normal softmodem: they just have documented API's for dealing with voice. And, of course, buying the products from Digium helps support the people who make all of this possible...

Re:Skype to POTS idea

[dossen]

Encryption? How? The attack described is against the analog line, after the call leaves the computer. At that point there is nothing you can do to protect the signal, you are trusting the person providing the modem to be a good guy.

Re:Skype to POTS idea

[Tony Hoyle]

FWD already this, for free... from here in the UK I can call many locations in the US without charge.

I also have a 'real' IP phone not an MSN clone - works when the PC is off.

Re:Skype to POTS idea

[dave1g]

No, don't attempt to trust the guy at all.

Only use the FREE service if

A) you are not a privacy freak

or

B) your call doesn't require privacy

Re:Skype to POTS idea

[michrech]

This wouldn't be any different than me walking up to your house in the middle of the night and hooking a small recorder right up to your phone line. Just because you aren't sharing your landline doesn't mean you are automatically any more safe from people listening to your conversation.

On a side note, I don't even half to go onto your property. If I were that dedicated, I could get a fake phone truck, a little orange vest, and just break right into the junction boxes on the sides of the road and listen into any number of phone conversations -- including yours.

Is that any more likely to happen than someone listening to your conversation through the means you suggest? Probably not. It can still happen, though.

Re:Skype to POTS idea

[Jetifi]

That's pretty interesting, but as another reply said, local calls aren't free in many places. That's related to the biggest hurdle: there's no incentive for people to hook up a modem to their broadband-equipped PC in the first place.

What could possibly work is wiring the VoIP<-->POTS software bridge with a built-in micropayment infrastructre, e.g. using the PayPal web interface, ignoring for the moment the PayPal-being-evil issue.

That way, people could charge for the use of their POTS line, and might even make money off their software - and there'd be nothing wrong with that since they're still saving the caller some cash.

Yours is a really interesting idea, and if you follow through at all with it I'd be interested in knowing about it - my email is this nickname at gmx dot net.

Re:Skype to POTS idea

[BrynM]

I had a related thought: Does anyone know how to jumper any of this software to your home phone? Run your server and tie in a regular handset so you could use a cheap cordless phone?

Re:Skype to POTS idea

[dave1g]

well most computers still come with modems. And the ability to screen based on the caller having his modem hooked up also would provide an incentive.

But your paypal idea could easily be worked into the equation. but in order for it to be a good deal it would need to be like $0.001/min

and I think there would be enough users in each area code who decided to go for free that the paypal people would nver get used.

Most people barely use their telephones, so you would only need a few skype users (10 pretty good uptime, 100 infinite up time) in each area code with their modem's hooked up to provide a good chance of their being a modem available at all times.

And in the end the idea is basically just a gimmick. No guaruntees are made, its just a cool little way to get around paying long distance.

Someone mentioned that free world dial up already has this ability but since it is controled by that company and not the users, and that company (from the admittedly little that I know about them) seems to not have a business model eventually they will switch to a pay service like all the others.

The paypal idea should work really well in places that do charge pre minute. In that case there will be no incentive to do a free modem and everyone will set up paypal accounts and it wont be hard to beat the crappy rates of most phone companies.

Re:Skype to POTS idea

[dave1g]

I'm confused.

Do you want to hook up a regular cordless phone base station to your computer and use the accompanying cordless phone to dial out to skype users?

I think the best you can do in this situation is have one of the phones that supports something like an intercom.

Then use skype to call some one, and have your modem forward the audio down the phone cord to the base station and then through the air to your cordless.

Im thinking that this isnt actually possible though. Because I'm not sure if phones can work like that. It is the equivalent of taking 2phones and wiring them together directly. Stand in 2 rooms and see if you can talk to each other.

If this is possible then I think what you want is also possible.

Re:Skype to POTS idea

[Anonymous Coward]

Had the same idea some time ago - the big (dick)head in the sky said no dice.

Re:Skype to POTS idea

[Jetifi]

One thing is that if you want to screen based on the caller having his modem hooked up, then you have to provide some verification of the presence of the callers modem via the 'net, which is difficult unless you actually use the hardware, i.e. call someone with his modem.

Even if you did this, assuming the software got popular, someone would eventually make a patched/hacked version of it that let you use other peoples lines even if you didn't have a modem hooked up to your own PC.

You're right that given the presence of free lines, eventually the users who charge for access would stop being used, again assuming the software existed and that it took off. At this point you might find area-code micro-cartels popping up to artifically raise prices :-)

Finally, your idea is not just a gimmick, it's a lot more than that. If it's feasible, it could end up as a decentralised grass-roots/P2P alternative to the VoIP solutions from Vonage, Net2Phone etc., and it would be totally impossible for the FCC or anyone else to regulate it. In addition, because it lets you get around paying long distance, you're hitting the big telcos where it hurts, which is always cool. And the end result is that you allow people to talk long-distance/transnational longer for less money and that's fundamentally a Good Thing.

Re:Skype to POTS idea

[NanoGator]

"Privacy is a big issue here. What's to stop me from hooking a $30 voice activated phone recorder to the line and record all of your conversations? I don't know how much I trust someone I've never met and who has no real oversight to protect my privacy. Scratch that.. I do know that I don't trust them at all."

Then why do you have a landline or cell phone? Sorry if I'm missing your point here, it's not intentional, I just don't understand why this concern is different.

Re:Skype to POTS idea

[Anonymous Coward]

What you are trying to do is a pretty standard way of using Asterisk. It is a software PBX after all. However, you'll have to "settle" for standardized protocols, like SIP, not proprietary Skype stuff. There are cheap interface cards to which you can attach regular phones, including cordless ones.

Re:Skype to POTS idea

[Anonymous Coward]

...sucks to not be in the US

Not long until we will be...

Re:Skype to POTS idea

[Huogo]

I've been talking over modems on occasion for many many years, and I've never had a single problem with half duplex or compatability problems.

Re:Skype to POTS idea

[tmasssey]

Do you mean talking in the idea of voice chat over a data connection? If so, that's not what I was talking about. In that case, you don't need a modem as anything more than a data transport: no different than Ethernet, wireless, etc.

But if you're talking about talking to a person using a plain old telephone, then the information above is correct.

For more information, check out the Asterisk mailing list archives.

Re:Skype to POTS idea

[Huogo]

I'm talking about using my modem to interface with the POTS network, not through the internet. All I know is that I've been talking through my modems since around 1995, and have never had any duplex problems. Whatever undocumented APIs there are must be understood quite well.

Re:Skype to POTS idea

[lga]

here in the UK we have to pay (rather a lot) for local calls

That's changing. For a while now BT has offered a tarif with free calls up to an hour. (Local or national) It costs 25.5UKP / month. From July even the basic line rental of 10.50UKP will give you evening calls up to an hour long for a 6p fixed charge.

Anyway, there is no excuse for putting up with BT's expensive call options: you can change to a cheaper BT tarif, or you can join one of the hundreds of alternative phone companies such as Onetel.

Re:Skype to POTS idea

[steve_bryan]

It doesn't appear you are paying attention. Skype is a replacement for PGPFone. The voice and chat traffic is encrypted end to end. That makes connecting to the PSTN a rather questionable idea though I have heard some talk about this from the phone companies recently. I wonder how the encryption issue will be handled. Anyhow, if you want a replacement for PGPFone without the debilitating lag try Skype. The audio quality is also much better.

Re:Skype to POTS idea

[afidel]

What are you talking about, at least on Windows there most certainly ARE documented methods for doing voice through modems. For Windows 9x that's Unimodem/V for 2k+ it's Unimodem/5 and both are Telephony Service Providers for TAPI (Telephony API). I know this because unfortunatly the voice addon for WhatsUpGold only supports Unimodem/V not Unimodem/5 so I would have to run it on an unstable OS to get that functionality and an unstable OS is a poor choice for a network monitoring station.

Re:Skype to POTS idea

[tmasssey]

That's only half right. Unimodem/V or Unimodem/5 are software API's. Just like NDIS is an API for network cards. Just because a network card has NDIS drivers, does not mean that the hardware interface is documented. The same is true for modems, video cards or anything else. The *driver* implements a documented standard (NDIS, Unimodem, VESA, etc.). The hardware can implement whatever magic interface they see fit: that's the whole point of drivers!

That's the problem here. Under Linux, you can't use vendor-supplied Windows Unimodem drivers. There needs to be a Linux equivilent. Just like you can't use your Windows NDIS drivers (weird software hacks excluded) to talk to a network card, or your Windows video driver for X.

While there *is* a clearly documented AT-style standard for voice, it is half-duplex. While there are *many* full-duplex-capable modems (mostly softmodems), their hardware interfaces are not documented. They don't need to be: the Windows driver does that for you. What few softmodems *are* supported by Linux are supported for data only. There is no support for voice.

That's why you need dedicated hardware for Asterisk. Having said all of that, an X100P is little different than a typical softmodem. It's just documented and supported. If you wanted to, you are free to write an Asterisk driver for a Lucent chipset. You would have a hard time, though, because there's no documentation...

End of Phone Numbers is just the beginning

[digitalvengeance]

I've been using vonage.com for my primary home line and fax line for a while now and I absolutely love it. Not only can one talk a lot more for a lot less money, the other features that are included are remarkably useful. I love transferring calls from my home phone to my cell phone before I walk out the door, and I'm planning to put together some code that interacts with Vonage's web "dashboard" to allow better integration with the rest of my digital life.

Marrying phones and computers in the home is going to open a whole new avenue of technical exploration, and unfortunately, exploitation as well..

Josh.

Re:End of Phone Numbers is just the beginning

[Anonymous Coward]

Shame about the audio delays, the having to dial 1 + your local code nonsense, the network overload busy signal, and not being able to call it from a PC over IP.

Ventrilo

[Daath]

When I play and use voice comm, I use Ventrilo. When I (or some of my colleagues) work from home, we use Ventrilo to conference, it works very well!
Ventrilo is also free.

Re:Ventrilo

[kunudo]

When I play and use voice comm, I use Ventrilo. When I (or some of my colleagues) work from home, we use Ventrilo to conference, it works very well! Ventrilo is also free.

Man, I love ventrilo, too bad there isn't a linux client. It's perfect since you can set up your own server (which runs on linux), and keep it private, unlike skype (i think).

Re:Ventrilo

[kunudo]

Linux i386 - 32bit In development

Yay! I just checked the ventrilo site! :D

Re:Ventrilo

[Anonymous Coward]

Too bad it only works on Windows.

Ugh, I hate developers that just target one platform. In todays age that just doesn't make sense.

Yes, I know they have MacOS/Linux clients planned. I'll believe it when I see it.

anyone using voip?

[trs9000]

is anyone using this? regularly? ..or?

any particular setup anyone recommends?
or troubles?

i think i may get rid of the cell and switch on over...
sounds like this is where things may be headed

Re:anyone using voip?

[grahamsz]

I was one of the first vonage customers since I was in a unique position where I lived in one country but worked and had a fiancee in the US.

Vonage worked out great since it gave me unlimited calls to colorado for a tiny fraction of the price i'd be paying to call internationally.

It had occasional problems, but given that I consumed about 3000 minutes in a high month, it worked great most of the time.

Currently i'm moving to freeworlddialup since everyone i talk to either has broadband or is local.

Re:anyone using voip?

[Tony Hoyle]

TBH I'm surprised slashdot is so late with this...

VOIP rocks. I haven't even got my FXO yet (shipping from the US takes time) and I've already got my PBX configured with cheesy 'on hold' music so I can annoy everyone :)

From work I have X-Lite plugged into my server at home, so if I'm needed it's a free call. I've also got local dialup numbers the US, Germany, etc... real fun.

For now it's totally unreglated and largely free (if you don't count $16 for the FXO, and $50 for the phone).. I doubt it'll last, but let's use it while it's there :)

Excuse me but

[fingerfucker]

Why do you call the Voice Peering Fabric a "clearinghouse"? By functionality, it is clearly a "peering center" and not a clearing house.

Maybe less fancy and more clarity would improve the eloquence of the posts...

kind of worrying?

[Anonymous Coward]

if we end up relying on making phone calls via the internet, what's to stop the next killer virus knocking out our phone 'lines'?

Re:kind of worrying?

[Patrik_AKA_RedX]

[_] Good security measures.
[_] Sane design.
[_] Educated users.
[X] Unbelievable amounts of luck.

Re:kind of worrying?

[Anonymous Coward]

A condom? Virus protection?

Re:kind of worrying?

[Anonymous Coward]

If mankind had worried about every possible virus before taking the next step in progress, we'd still be living in caves.

Yes, we should try to make the system as bullet-proof as possible. No we shouldn't wait until it's 100% safe before using it (it will never be that safe). Remember, phone lines weren't exactly reliable at the beginning either. We could've stuck with written letters since that method was well-established, but we went ahead with phone lines, without which we would've never had the internet.

Re:kind of worrying?

[chill]

The same thing that is stopping them now -- sheer, dumb luck.

There have already been incidents of 911 systems and certain local phone systems going down due to viruses and worms. Not to mention what a good phreaker can do to a corporate PBX.

Hey! If we switch to VOIP, maybe phreaking will go away. What's the point in using someone else's phone when yours is free or almost?

-Charles

Asterisk PBS is awesome

[Anonymous Coward]

Asterisk is an awesome piece of code. They have presented at several hacker cons in the South East (Phreaknic, Interz0ne), and I have been quite impressed. Most PBXs handle things like T1 frame generation, TDM, switching, etc in hardware. Asterisk does it all in software using Linux

I hope patents won't impede these developments

[motown]

I'm afraid telcos are letting their legal departments burn the midnight oil in patenting anything obvious and even remotely related to this.

The combination of two obvious technologies (telephony and TCP/IP) should not be patentable.

We'll see how this turn out.

The various Linux telephony projects being mentioned left and right are hopeful developments.

Now that (soon to be "legacy") regelar phone networks are being migrated to TCP/IP, I think the time has really come to finally seriously start making the switch to ipv6.

Re:I hope patents won't impede these developments

[shakah]

I'm afraid telcos are letting their legal departments burn the midnight oil in patenting anything obvious and even remotely related to this.

VoIP is really pretty old (Net2phone popped up around 1996 or so), all the patent-related land grabbing is either over or nearly so.

this story is boring

[Anonymous Coward]

anyone got any links to some good porn? no goatse's or tubgirls please.

TeamSpeak = audible IRC

[reality-bytes]

Teamspeak is great!

A group of my friends have a Teamspeak server set up (on my cable line) and we leave it going 24/7 so anyone can join when they like and see if anyone else is around; typically if we are not on TS, we don't wont to be bothered or can be 'knocked' by IM.

Teamspeak is more like IRC in that you can have lots of people on channel at the same time. Its really reliable too, our uptime record on a dodgey old 350Mhz SuSE box is 64days - only interrupted by a blown PSU.

Packet 8

[Scorchen]

I've been using Packet8.net for my telephone service for about a half a year now, and I've been thoroughly impressed. For 20 dollars a month, I get completely unlimited long distance, call waiting, voice mail.

There are a few glitches every now and then. Somedays it won't let my calls go through, but usually my voice mail bails me out. It's just as reliable as a cell phone, and much much cheaper.

Re:Packet 8

[Anonymous Coward]

It's just as reliable as a cell phone, and much much cheaper.

Only downside is that thing about portability, eh? ;)

How I Learned to Stop Worrying and Love The Virus

[gfecyk]

what's to stop the next killer virus knocking out our phone 'lines'

Good design, lots of testing and before-the-fact protection. Just make sure the Anti-Virus Cartel doesn't get involved.

coming soon to a legislature near you

[cluge]

Expect to see legislation with some assinine label like "The VoIP pro competition rules". This will be of course a couple of millions lines of regulatory text that will be sold to the public as "deregulation".

What will this legislation accomplish? It will help the baby bells and large phone companies hold on to their market shares and stifle competition.

Jaded? Me ? Naaaaa, just experienceed with more than a decade of being told "We are for de-regulation" while signing bills that give the big guys even more. The FCC unep ruling recently is a perfect example of such double talk.

cluge

Hmm, shame about the name...

[Seydlitz]

Yeah, VOIP is big- but at the moment, let's face it, the name sucks.

I mean... VOIP? Try discussing that in a bar or in a non-technical environment. No, seriously, go try it: how do you even pronounce VOIP? I guarantee that you'll get laughed at, or stares at the very least.

Therefore, I suggest that we replace Voice Over IP with Talking Over The Internet, or TOTI. Think about it!

Instead of discussing VOIP down the pub, you can talk about the latest TOTI that you saw Samsung introduce. Or ask if your fellow geek has checked out the new TOTI down at the phone store. Or if you see a nice looking young lady (or man!) in town, you can whistle and challenge those around you to check out that TOTI.

You see? It's perfect. Sure, it isn't as descriptive as VOIP, but it gets across the main idea, kind of. I mean, yeah, there's going to be the odd pedant that will contest the change, but don't listen to them- they're probably a taxman in real life or something.

Re:Hmm, shame about the name...

[pe1chl]

It is moderated as Funny, but I think he has a point.

Remember how Linux won the battle against *BSD.
It just had a better mascot and a nicer name...

If nothing else

[doormat]

VoIP is very good competition for the POTS phone companies. Its funny now I see plans for $50/mo unlimited long distance. A few years ago I can remember a friend paying gigantic sums of money on long distance phone bills. Now for $50, its an all-you-can-eat buffet.

Re:If nothing else

[Wakkow]

I think the cell phone industry helped this also.. They're the ones that kept trying to outcompete each other with more minutes.

Hi, some good links

[Anonymous Coward]

http://www.peoplesprimary.com the number one source for great p0rn links (not work safe!!!)

Re:Hi, some good links

[NumbThumb]

gah! it's slashdotted!

VoIP Quality

[TgrMan]

As with anything running over the public Internet, there will, at times, be issues which cause disruptions in service; ie. a M$ worm sucking bandwidth, neighbor downloading lots of pr0n and slowing down your cable modem connection, etc.

The key to voice quality with VoIP is latency. Most VoIP endpoints have a built in jitter buffer which is able to recover from some latency, usually around 3ms, but after that is begins to be difficult to carry on a conversation. The latency here is the latency between you and the provider's media gateway, such as an Asterisk box. Normally if you have less than 50ms of latency, then you'll notice no degradation in call quality. If you begin to have more latency, the quality of the call will begin to drop off.

I've been using Nuvio for almost 6 months and it rocks! Latency on my cable modem is around 30ms back to their servers so I rarely, if ever, have a quality issue. Their web interface is pretty cool and they do some cool stuff you can do with regard to voicemail and e-mail. Plus it's not PC based, just plug your regular phone into the adaptor they send you and you can take your adaptor with you when you travel, plug it into any broadband connection and still make and receive calls just like you never left home. This is pretty cool because I travel a lot and if the hotel has broadband in the room, it works great. I even have a few virtual phone numbers across the country so people I know in those cities can call me for free.

As someone mentioned before, it's a lot like cell phones. You just have to look at what you're getting and decide if it's right for you. If you'd rather pay less than with a regular phone provider and be able to make TONS of long distance calls where an occasional dropped call is ok, then go sign up with Nuvio. It's really as good as the reception is with a cell phone, and most of the time lots better. It all depends on the latency.

Re:VoIP Quality

[awehttam]

The beauty of having a DNS root like e164.org is that calls can be entirely peer to peer. You're not sending all of your voice traffic through a single upstream like Vonage or Free World Dialup. If you ring my number, your local voip gateway connects directly with mine. Sure, if there's a hit on my local network or yours, then there will be lag issues. Alas that is the nature of the 'net.

Re:VoIP Quality

[walt-sjc]

Um, Free World Dialup just handles call setup, then your sip devices directly talk to each other. Vonage != FWD != e164.org. They are all very different.

Re:VoIP Quality

[Anonymous Coward]

Specifically, FWD is a signalling-proxy, Vonage is a gateway and e164.org is a registry.

Re:VoIP Quality

[awehttam]

You sure about that? Last time I looked, they didn't allow re-invites.

Non-profit?

[Anonymous Coward]

If you think that the people behind e164.org are in it for the good cause, you're kidding yourself. There already is an official phone number to DNS tree: e164.arpa, as designated in RFC 2916. This is a fight for the root of _the_ registry of all POTS-number to VOIP/email/web mappings. There's money to be made, and lots of it.

Re:Non-profit?

[Anonymous Coward]

You bet there's money to be made, in building networks for communities without service and providing services that consumers actually *need*, not what some telco thinks they need. Do you use e164.arpa? Even if you do, you can still use .org as a supplement. Don't try to monopolize telephony online any more than the PSTN already is. Networks want to be accessable, not fill the pork bellies of telecom.

Re:Non-profit?

[Anonymous Coward]

Phone numbers are the domains of the telephone world. You don't want to end up with competing registries and inevitable collisions. In a sense the situation is even more complicated than with domains: Users are not going to enter 20-digit phone numbers, so the space is much more cramped. Everybody will want a nice prefix for their VoIP network, users will want vanity numbers.

Using e164.org as a supplement is like using ORSC root nameservers. They offer a bigger namespace by filling in some gaps in the "official" namespace. Collisions occur when the official namespace itself fills the same gaps. At that point you have to either live with the ambiguities or the alternate registry has to tell all its customers that they have lost their allocation and need to reapply through the official registry.

how about VOX

[w1r3sp33d]

VOX is an old term that covered any Voice Over type (ip, frame relay, atm, whatever) and it sounds cooler than voip or v,o,i,p.

Re:how about VOX

[RollingThunder]

I've always understood VOX to mean "Voice Operated Transmit" - IE: not having to press the transmit key.

BT's Bluephone..

[KitKitNet]

Someone should check out this new Bluephone! It is the ultimate VoIP package, u can use your cell over Wifi, 3G or GSM networks, probably the killer app for VoIP.

VoIP is cool but...

[v5out]

There's no way for the little guy to make any $ with it so it will be owned by the big corps forever. No, Asterisk is not the answer because, while it works, it is totally not a commercial product. And even if you productize Asterisk how do you sell it? Companies are not going to put their mission critical phones on some open source thing. Would you? (IT and telephony guys only please)

Re:VoIP is cool but...

[Thomas Shaddack]

Companies are not going to put their mission critical phones on some open source thing. Would you?

I would. I would also put a mission critical email, web, and server systems on some open-source thing. Then I won't be so damn dependent on the vendor's mercy.

I prefer architecture designed to do things well over an architecture where doing things well is only a method to make money and where things that don't make money (or enough money to be profitable for the vendor) are neglected (and where doing things not-that-well is preferable over doing them better if the vendor's business plan asks for it).

Re:VoIP is cool but...

[Anonymous Coward]

No. This is completely incorrect. SIP/IAX/etc are open standards which include the "little guy". With Asterisk, you _CAN_ get completely comercial support. I've be doing completely PBX installation using asterisk for a little while now. Mostly at small banks. People just want a working phone system, they typically done care what it is. As long as it works. Asterisk is a excellent peice of code, and I'd put it up to any other PBX out there. Would I? I do it every do. The end user saves money (because they don't have to pay out the ass for a really lame PBX). I make money by charging for the hours for installation and support. Everyone is happy.

Re:VoIP is cool but...

[v5out]

You must be a really good sales guy or these banks must really trust you. I can't get to first base with anybody I talk to. Typical comments - if I can even get somebody on the phone - 1. Asterisk is not fault tolerant, secure, [fill in the blank], etc. 2. Who the heck are you?

Re:VoIP is cool but...

[mindstrm]

I've seen a number of medium sized businesses using asterisk, or planning to.

It's not just "some open source thing". It's solid. It's very well designed. And it is commercially support by the core developers if you want it to be. (www.digium.com)

Companies will tend to put their "mission critical" systems on stuff that works, open source or not. (Apache?)If they can pitch to the bosses this new VOIP product they can roll out for a couple thousand bucks instead of $50,000 or easily much more, most IT managers would say "That's worth a shot! Let's see how it goes."

Re:VoIP is cool but...

[knewter]

The company (an IT/CLEC) I previously worked full time at, and currently manage some projects for part time, is rolling out an Asterisk box over which they intend to begin sending an awful lot of long distance calls. We've been doing VoIP internally and for some customers for easily 1.5 years, and they have complete faith in open source software. This coming from a shop whose web servers are all windows-based, we're about to receive Microsoft Gold Partner Status, etc...you just do what works, and makes you the most money.

Asterisk open source?

[Anonymous Coward]

From libpri/pri.c:

/*
* libpri: An implementation of Primary Rate ISDN
*
* Written by Mark Spencer
*
* This program is confidential
*
* Copyright (C) 2001, Linux Support Services, Inc.
* All Rights Reserved.
*
*/

And from zaptel/fasthdlc.h:

* This table based HDLC technology is PATENT PENDING, but will always be
* remain freely distributable under the terms of the GPL version 2.0.
*
* For non-GPL licensing, please contact Mark Spencer at
* the below e-mail address.

Re:Asterisk open source?

[raphae1]

If it wasn't open source how did you get the code?

SIPPhone

[Inexile2002]

Picked up a SIP Phone Call-in-One and I've got to say, I'm really impressed. It does exactly what they say it'll do, it does it well and it does it cheap. Living in Madrid, until now, meant I was sort of cut off from my brother. Now, I call him whenever I want, talk as long as I want and it doesn't cost me anything other than the bandwidtch charges for the DSL connection that I was already paying. Definitely impressed.

Also, we've been using the SIP Minutes thing to make calls to Canada, Chile and and a couple of other countries - all to PSTN numbers - and we're really happy with the cost (cheap as dirt and no monthly fee or anything stupid like that), the sound quality and ease of use are excellent.

My only complaint is that they won't hurry up and set up Virtual Numbers. I'm thinking of starting a small business and being able to have a local number in Paris, NY, Toronto and Vancouver would really help me out. Having all of those numbers ring through to the same phone... beautiful.

Played a little with Skype (litterally, we were roleplaying and brought a friend in from another town) and I wasn't as generally impressed. Sound got wonky too often for my tastes although it could have been a bum mic.

Also, been toying around with this and I have to say this just makes my SIP Phone better. Friends from all over can call me, leave me a voice mail if I'm not there (which gets emailed to me of course) or, with this, I can (and have) set up some cool conference calls.

Scenario.

[mindstrm]

My mother has a SIP phone (actually, a SIP adapter) at home.

So does my uncle.

So does my little sister.

So do a half dozen friends (spread out over several countries in both hemispheres).

All these SIP devices connect to a very small linux box colocated in the US, running asterix (which is an excellent piece of software, btw).

Through that, I issued everyone an extension, voicemail, etc.

Further to that, anyone who wants an account at iaxtel, free world dialup, voicepulse, well, the asterix box can connect to those for them, and route calls to the appropriate extensions. It can also share the outgoing services (Which may violate some TOS, who knows) among all the users (or selected ones).

I can also use end-user phone systems as dialin/out lines, which does require some slightly different adapters.

My point?

VOIP is not a heavyweight thing. The only big scary part is locating the correct parties... this is where POTS works well.. because we have a global dialplan that everyone agrees to.

Eventually, everyone with any bandwidth will just HAVE phone service to each other, with some kind of globally understood dialplan beneath it (which will not be based on numbers at all) and links back to POTS will be fewer and rarer.

Re:Scenario.

[mindstrm]

Of course.. I meant asterisk.

No matter how many times I type it, my fingers just won't listen sometimes.

Re:Scenario.

[Anonymous Coward]

With ENUM you can eliminate the colocated Asterisk box. SIP phones don't need a PBX in order to talk to eachother, they just need a way to find eachother. That's where ENUM steps in: Simply put all call setup information (IOW a SIP-URL) into the DNS and the phones can look it up when you dial a phone number.

ENUM is the "global dialplan" which you fantasize about, and yes, it will be based on numbers. Otherwise you could (can actually) use SIP URLs directly.

DNS to phone

[captaineo]

I would REALLY like to see a standard for finding phone numbers via DNS. We use DNS precisely because it's too hard to remember the correct up-to-date IP address of the services we want. Why should we continue to remember peoples' phone numbers? Now that many cell phones are internet-connected, you should be able to type in e.g. "call.joesmith.com" and the phone should do a DNS lookup to get the phone number.

Maybe this has been done already? Or is the "typing in" the problem?

Re:DNS to phone

[Anonymous Coward]

This article is about the exact opposite. VoIP phones can use textual addresses, but unless you're at a PC with a soft-phone, there's no practical way to enter something like jane@sip.the-does.com. ENUM offers a lookup from phone numbers to URLs, so when you dial Jane's regular phone number with your VoIP phone, it will look up the SIP URL and connect to sip.the-does.com and ask for Jane.

Re:DNS to phone

[muonzoo]

This is contemplated in SIP (RFC3261-3265), in particular, RFC 3263 Location of SIP Servers. You can register your SIP URI (sip:user@domain.com) and have a static registration to many contact URIs, like for example:

• mailto:user@mailservice.domain.com
• sip:mymobilephone.wirelesscarrier.net:5060
• tel:+12125551212

When someone 'calls' user@domain.com, they have a choice of how to contact you. Typically the service provider will match caller capabilities to the registrations in the service database.

The end result could be REALLY cool. You might not get my phone, but you could automagically send me email. Or I could divert you (if you calling device was capable) to a blog / presence URI that explained where I was and what I was up to. Never mind the ACTUAL implementation of presence and instant messaging that ALSO leverages this infratstructure.

Every time I hear about proprietary solutions to VOIP (like Skype) or people going on about Jabber I sort of shake my head and wonder why?

SIP provides an amazing opportunity to provide integration rich-content and services over a standard infrastructure. I cannot wait for this to start being deployed.

People who have opinions about NATs, firewalls and connectivity issues haven't done their homework. Commercial solutions exist today that skirt the NAT issues and standards based solutions are nearly RFC'ed at the IETF.

Re:DNS to phone

[LWATCDR]

you should be able to type in e.g. "call.joesmith.com" and the phone should do a DNS lookup to get the phone number.

probably because there are too many joe smiths.
Phone numbers pretty well for the most part.
I wonder if google has thought about google phone? Make it part of the new Google IM service and offer a free gig of on-line voice mail. Comppressed using Speex from the Ogg project just to make it all as geek cool as possible. Of course the should make a Linux i386 version as well as a Linux PPC version to keep it all geek cool :)

Re:DNS to phone

[awehttam]

DNS SRV records will do that. _sip._udp.yourdomain.com. 3600 IN SRV 0 0 5060 sip.yourdomain.com. well, it's not exactly what you're suggesting.. Hmm..

Re:DNS to phone

[stpeter]

> Never mind the ACTUAL implementation of presence
> and instant messaging that ALSO leverages this
> infratstructure.

Except that there's no there there -- does anyone actually use SIMPLE for IM? 10+ million people are using Jabber/XMPP, which is just as "standard" as SIMPLE given that the IETF has approved the XMPP specifications. :-)

> Every time I hear about proprietary solutions
> to VOIP (like Skype) or people going on about
> Jabber I sort of shake my head and wonder why?

You're right regarding Skype, but people are "going on about Jabber" because it is an open standard with tons of open-source code behind it, which enables you do to lots more than IM (it's all just a streaming XML technology -- think SVG whiteboards, real-time RSS/ATOM feeds, and other forms of dynamic content).

As to VoIP, enabling most voice (and video) over the Net is just a question of how you get to RTP or some other bytestreaming technology (you certainly don't do the content exchange via SIP, only the negotiation). You could get there via SIP, via H.323, or via Jabber (expect protocols and implementations soon), but in the end you're just streaming some bytes. Who cares how you get there?

And let's not forget that sometimes people want to exchange text (IM, email, IRC, etc.) rather than voice or video. Personally I rather like having multiple channels going simultaneously -- voice is so single-channel that it makes multitasking hard.

Anyone know

[mindstrm]

of a service offering unlimited calling to the US & Canada (the Canada part being significant here) that will work with asterisk?

Hint: Packet8 and Vonage will not.

Hint: Voicepulse works great, but only has calling in the US.

Re:Anyone know

[Tony Hoyle]

If you know of a way to get free local calls in canada (I'm guessing that that's not the norm) then you just have to setup FWD gateways in the various local areas. AFAIK that's how FWD manages to cover the US so well. Of course how long it'll last before the phone companies catch on is debatable.

Re:Anyone know

[mindstrm]

No... canada works just like the US. Local calls are free.
I meant long distance.

Also.. why does everyone say FWD covers the US? As far as I can tell, FWD only covers toll-free numbers and other FWD users, all VOIP. It doesn't have dialouts all over the US.

Speaking of mapping things using DNS...

[borud]

Here's an idea that me and a friend thought a bit about last year.

Re:Speaking of mapping things using DNS...

[Anonymous Coward]

Check out RFC's 3401-3405 and DDDS URN's

Don't buy cheaper card, buy from Digium, support *

[SimonDorfman.com]

Someone wrote to the Asterisk list asking about these cheaper cards and someone eloquently replied here:

But that's just not "the right thing to do". Asterisk development is paid for in part by sales of this hardware. Buy it from Digium, and you get support as well. I had a problem compiling the zap drivers when I got mine. When I called, the phone was picked up immediately, by a real person who knew exactly what they were talking about. Digium support actually SSHed into my box and fixed it/showed me what I was doing wrong. The support is well worth the price, especially if you are building a production server. Or if your time is worth anything at all for that matter.

Reversal ?

[Naut]

its kinda funny 10 years ago you had to use p.o.t. service to get on the internet , now you'll be talking on your phone thru a i.p. proticol no phone lines need . I can see maybe in the next 10 years there might not be any phone lines left , and if they are it will be in those areas that are hard to reach with broad band now .

Re:Reversal ?

[pe1chl]

Of course you will still need the phone lines!
Unless they are replaced by fiber to the home, or so. But that costs so much that rarely anyone is doing it.

Broadband is also transported over phone lines.

Re:Reversal ?

[Technonotice_Dom]

I've got an ADSL connection right now... so by using VOIP, my phone data is going over the IP link provided by my phone line. Which is kind of strange to think about...

Analogue voice being converted to digital, then back down to analogue to be carried across my phone line, converted back into digital at the phone exchange. Carried hundreds/thousands of miles as digital data only to have the reverse happen to it...!

One down, one to go

[Graabein]

Disclaimer: I'm a partner in a Linux and Asterisk based VoIP startup in Norway.

Thanks to efforts like E164.org, one day phone calls will be just another service running on the Internet. There will be no fees for doing simple peer-to-peer connections (me calling you to say hi), however special content and services will still have fees. Gateways to POTS* will be one such service for the foreseeable future.

What this means is that we're in for a revolution. We're rapidly moving from a model where the Internet is run over phone lines to where the phones are run over the Internet. IOW, basic communications go from a metered service produced by a relatively small group of very lucrative companies, some of which are still state-owned monopolies, to the Internet model which we all know: A basically cooperative network where people purchase the bandwidth they need and agree to connect to their neighbours in order to join the big network. As I'm sure you've already figured, selling that bandwidth will still be big business, but nowhere near as lucrative as selling metered service.

The big thing to get here is that the production of value, i.e. stuff that people are willing to pay for, will move from the center of the network to the edges. That is, from the big (sometimes monopolistic) phone companies to you and me. Welcome to the Internet revolution, you thought it was mostly over but in reality it's just starting. Oh, and if you thought the RIAA was running scared, they have nothing to fear compared to the big ol' phone companies.

So what do I mean by "one down, one to go"? Two things, actually. First, e164 directory services are the first step, the second involves VoIP providers creating an environment where they can exchange traffic and get paid for services they provide to other providers. Say a user on your VoIP provider in the US calls a POTS number in Norway via a VoIP provider in that country with a POTS gateway. Peering the actual voice traffic is just one part of the exchange, peering the business end of things is the other. But this too will fall into place, sooner rather than later.

The other thing I'm thinking of as "one to go" is TV. Today most countries are building separate infrastructures running parallell carrying different services: Voice over phone lines, TV over cable or OTA** and Internet over whatever's available. I'll give you an example:

In Norway the Storting (Parliament) has decided to spend $0.5 billion to build an OTA infrastructure to transmit digital TV signals. Once the new network is in place, the old analog network will be switched off. And you thought the US digital TV mandate was bad...

My prediction is that before the new digital TV network is in place it will already by severly outdated, completely overrun by the Internet revolution.

The world is rapidly becoming digital and the Internet is the enabler. There is no future in building any kind of infrastructure unless it is dedicated to carrying Internet traffic. Wired, wireless, optical, satellite, it just doesn't matter as long as it moves the bits.

Welcome to the revolution, we're just starting.

* POTS: Plain Old Telephone System
** OTA: Over The Air, traditional broadcasting

Re:One down, one to go

[pe1chl]

>In Norway the Storting (Parliament) has decided to spend $0.5 billion to build an OTA infrastructure to transmit digital TV signals. Once the new network is in place, the old analog network will be switched off. And you thought the US digital TV mandate was bad...

>My prediction is that before the new digital TV network is in place it will already by severly outdated, completely overrun by the Internet revolution.

I don't think it will be that bad, unless they take a decade to build it.
Digital TV requires broadcast bandwidth of about 300 Mbit/s for a typical channel package, or multicast bandwidth of 10-20 Mbit/s to each and every home. That is something completely different from Voice over IP!!

Such bandwidth is not in place in todays typical Internet access networks, and not everyone has or wants to have such wideband Internet access.

Even now, most providers have no multicast or broadcast capability AT ALL. Before things like TV broadcasting over Internet are even to be considered, the lowlevel network must first be made multicast-capable, so that not every viewer has to setup a private connection to some streaming server.

Worse, the way many ADSL access networks are built, with ATM throughout the network and PPPoA on top of that, even precludes the efficient use of multicast...

DVB-T is a very efficient broadcast method in such an environment, something that will be hard to beat in a wired network with point-to-point characteristics.

Dialtone isn't a right

[obtuse]

You'd lose your phone.

You think your ISP gets cranky about you running a server? If you did this, and the telco figured it out, (for example by looking for symmetric traffic on your internet line and your phone line, or by subscribing to the service themselves to get a list of numbers) they'd cut you off. Not many people will want to burn down their landline to give away phone calls.

Twenty some-odd years ago, a guy from my high school was caught running a demon dialer. Southwestern Bell explained to his parents that he could have either a live phoneline or a modem, but not both.

This is what I always wanted to do with VOIP too, but I could never convince myself that it was worth the risk.

I've heard that some companies with international networks and PBXs are illegally doing something very similar with VOIP, but only with their own calls. Their lines are expected to be heavily utilized, and they have much more leverage with the telcos.

Re:Dialtone isn't a right

[dave1g]

"for example by looking for symmetric traffic on your internet line"

I use a cable line for internet not DSL...

And number's arent stored in the system, only area codes.

A Telco cant take away your service just for being on the phone line alot.

And in the event that I am wrong and they can disable your service. I can still sign up for an alternative provider...we now have choice for local phone service since the 1996 Telco Act...right?

Re:Dialtone isn't a right

[kurth]

No, I believe that your totally wrong.

VZ reserves the right to terminate your service at a moments notice, if it is believed that you could possibly be doing something that could damage any of thier equipment or cause any other legit users problems.

And no, when you move from d-tone provider to d-tone provider the ILEC is still responsible for installing the lines and what not, they can deny the request for services on your pair, if you owe them money, or what what not. They can also hold number and not release them, so that you can't port your number to another carrier.

The TelCom Act wasn't enacted for ILECS to lose money, it was enacted to ensure that you have a choice in d-tone.

Why not just VIP?

[zerofoo]

VIP sounds pretty swanky to me.

-ted

e164.org is the same as alternic

[Anonymous Coward]

Don't any of you really understand what e164.org is? It's an ALTERNATE ROOT SERVER. That's right, folks, this is just like someone saying that they have a good solution to the .com name problems - they'll just start up their OWN .com root server and have everyone point to it! Then, you can get whatever domain name you want!!! ...as long as everyone points to that root server. We've been here before - don't any of you remember?

e164.arpa is the "real" root zone that e164.org is trying to replace. There are good reasons for wanting an faster/better/more clueful interface to ENUM, because e164.arpa is clogged with political sewage. However, I would want to see an organization with a little more clout behind them before I'd ever consider putting any time or money into an alternate root service; don't be surprised if suddenly you see a "cost recovery" (cough, cough, , cough) charge for usage.

Re:e164.org is the same as alternic

[awehttam]

No, E164.org is not an Alternic. We are not serving a domain called "e164.arpa", we're providing a suplemental domain called "e164.org" that can be used *in*conjunction* with e164.arpa. Using it is no different from using your own enum root for a local dial plan, or FWD using ENUM for their non-PSTN numbers. Secondly, AlterNIC suffers from an ownernship problem. E164.org actually verifies PSTN numbers to avoid conflicts like this. Thirdly, e164.arpa is not in production. There are no records for +1 (North America). Sure some people have it for experimentation, but e164.org is live and live NOW. . However, I would want to see an organization with a little more clout behind them before I'd ever consider putting any time or money into an alternate root service; don't be surprised if suddenly you see a "cost recovery" (cough, cough, , cough) charge for usage. If you even bothered to look at our site you'd know that we do ask for donations, because some areas are more expensive than others to verify. You can't expect us to foot the bill for a $9 per minute charge to Tuvalu or INMARSAT. For the most part verifications are free (and we let people do up to 5), after that, we ask for a donation. I think that's pretty reasonable.

Re:e164.org is the same as alternic

[Anonymous Coward]

Pardon my ignorance, but ENUM maps phone numbers to VoIP addresses, right? How can there be "supplemental" ENUM domains when there is only one world-wide POTS numbering plan? There can only be one authority for +1-202-395-1232. Who's it going to be, 2.3.2.1.5.9.3.2.0.2.1.e164.arpa or 2.3.2.1.5.9.3.2.0.2.1.e164.org?

Re:e164.org is the same as alternic

[awehttam]

It's a configuration thing, use .arpa primarily and fall back to .org. Try it now, you won't get many hits from .arpa. Like wise you can go with .org, and use .arpa if it doesn't exist.

Re:e164.org is the same as alternic

[Anonymous Coward]

See, that is the exact same setup which the alternate DNS roots use to avoid (well, mitigate) collisions. Problems still arise when both registries have data for the same domain: Resolution depends on resolver priority.

Besides, is this not just a scheme to get a foot in the door for assignments of VoIP network prefixes? FWD likes to be 393 (~FWD) in inter-network connections. 393 is an unassigned area code in the North American Numbering Plan. Will Pulver not try to put FWD numbers into e164.org under 3.9.3.1.e164.org? Are you not going to try and establish a "VoIP country code", for which you could be the assigning authority?

Re:e164.org is the same as alternic

[awehttam]

Pulver wouldn't be able to put it into +1 393, unless they actually own the number. E164.org is not a free-for-all, you have to prove you own the number before you can stuff it into the zone. From what I remember of AlterNIC, anyone could register sco.

Re:e164.org is the same as alternic

[Anonymous Coward]

I understand that you recognize the authority of the ITU over country codes and the authority of the FCC over US area codes. Is that correct?

In case both e164.org and e164.arpa contain records for the same phone number, will you remove your records to reestablish unambiguity? In other words, will you technically enforce your role as a supplemental registry?

Re:e164.org is the same as alternic

[awehttam]

I understand that you recognize the authority of the ITU over country codes and the authority of the FCC over US area codes. Is that correct? You know, not everything has to be a regulatory issue. The ITU is the International Telecommunication Union, there's no way around that. We're not interested in compromising the integrity of a numbering system unnecessarily, I'd be pissed if someone re-directed my number too.

In case both e164.org and e164.arpa contain records for the same phone number, will you remove your records to reestablish unambiguity? In other words, will you technically enforce your role as a supplemental registry? That's what we're doing.

Re:e164.org is the same as alternic

[Anonymous Coward]

May I suggest being a little more upfront about it, maybe by putting this information in some sort of mission statement on your website?

What about ENUM?

[CoolGopher]

Without having read the article (this is slashdot after all), what's wrong with ENUM? That already provides phone# to location/service mapping via DNS...

Re:What about ENUM?

[awehttam]

Nothing's wrong with ENUM, E164.org is an ENUM provider. :) The real question is what's wrong with e164.arpa. The answer, well. They don't already provide phone# service mapping via DNS, at least not in production.

Re:What about ENUM?

[Anonymous Coward]

e164.org is an ENUM clone, done by different people under a different domain. The situation is comparable to the relationship between the Alternic DNS rootservers and the ICANN rootservers.

Lame

[n8twj]

Who's giving me odd that Digit networks paid for this story.

Re:Lame

[awehttam]

No, they didn't. The only reason I linked to them is to illustrate the point of how low-cost VoIP solutions can be.

Re:Lame

[n8twj]

digit networks is just trying to make a quick buck of of Mark Spencer's time, money and effort.

The only thing that digit networks is going to do is cause more support requests to go thru Digium, because they are selling inferior hardware that just happens to function, because it has a very similar, driver compatible chipset to the actual chipset that Mark chose for Digium's X100P.

Then another topic is that they just happened to choose a company name very similar to Digium!? Coincidence? I think not.

Everything about their operation is totally shady in my book.

Mark wrote a custom driver and can at least justify the markup by actively supporting the hardware...digit networks can't say that...they have been referring support customers to the asterisk mailing list, irc channel and but of course Digium.

digit networks wouldn't be in business today if they didn't have Digium to step on, just to make a few a points of margin...how lame.

I tell everyone that come across NOT to buy from digit networks or GoVarion, which is yet another topic, even if they aren't looking to buy any hardware, but that's just me.

73 de n8twj

can we call it VoIP?

[Anonymous Coward]

coz thats what im calling it

I can't wait until VOIP becomes a reality

[Anonymous Coward]

I'm so excited!

I wrote a simple app...

[justin00]

I was wondering how long it was going to take someone to set up a DDNS server for this purpose. About time.

I wrote a cool app which streams voice in between two IP addresses after having some bad experiences trying to get some of the more complex app's to work well, and thru a firewall correctly.

It's here...

It just uses one UDP port (51981), and works pretty well. Other things work well, but they require a service, like yahoo IM... and I like the software to be independent of any servers, and be lightweight enough to use in the background.

Actually, it ain't a revolution

[Taco Cowboy]

Not trying to be picky, but the VOIP thingy, with asterisk, e64.org and all others, are still not qualified to be called a "revolution"

A revolution means Industry B replaces Industry A, just like the cars replacing the horse-buggies.

No matter how the VOIP progress, at least in the short term, will NOT replace POTS. This means, two industries will exist side by side, with some bridges in between the two camps.

This scenario is much like the toothbrush and toothpaste symbiosys - without one, you don't need the other.

All I can see - and please correct me if I am wrong - is that VOIP will co-exist with POTS for quite some times, until the time where we have embedded communication devices on our shirts, or the back of our hands (much like some of the sci-fi flicks) and at that point, we might don't need both VOIP and POTS.

Re:e164.org is the same as alternic(mod parent up)

[malx]

Mod parent up: this is an important issue.

I'm disappointed the Slashdot editors didn't notice that e164.org is in essentially competition with e164.arpa; this is very important to understanding what e164.org is about. As the parent says, it's like Alternic or, to avoid the unfair comparison with Altnernic's business practices, New.net.

I don't accept my sibling post's claim that e164.org is not in competition with e164.arpa but is merely "supplementary": that's like saying ".travel" is "supplementary" to .com. The fact is, while any old joe can say that you should register your telephone number in their DNS zone, e164.arpa is the domain the RFC says to do it in. Having two such domains is to have two roots for this mapping, with the possibility of discrepancy between them. Surely that's significant enough for the editors to mention?

PSTN-DNS mapping is not an easy topic. For example, how would you like someone else coming along and registering *your* phone number in an ENUM server (and so being able to receive SIP VoIP calls intended for you)? I'd be interested to hear how e164.org plan to prevent this.

And finally, VoIP is pronounced as a single syllable, with the vowel sound rhyming with "boy" :-)

voip elsewhere

[Anonymous Coward]

guys.. its great that i see you posting about USA and Canada. But VoIP is happening worldwide. It is a reality weather we want it or not. Most carriers are doing VoIP for ILD and we dont even know about it. I work for HablaNow (http://www.hablanow.com), a LatinAmerican VoIP provider, and i must tell you, what is most disturbing is the attitude of the governments down here and the problems deploying VoIP legally due to monopoly telco situations. All of this is changing wether they want it or not. Everybody's doing VoIP, from small cyber cafes, people using SkyPe, Asterisk, iptel, etc all the way to traditional carriers. There's no stopping for VoIP ;-)

leaps n bounds

[neuraloverload]

voip rollout should be held up as a model of development in real world applications that allows for the customization of essentially blank template devices. a healthy mix of different ideas expressed in different languages, across varied platforms, and with extreme flexibility. downside? just what are we going to do with all that twisted pair?

SpamOverVoiceOverIP

[upside]

Great stuff, now we can just expect MILLIONS of spam calls from China.

- Unregulated
- Dirt cheap
- Easy to script a 5 sec call to millions of numbers

Re:Don't buy cheaper card, buy from Digium, suppor

[Anonymous Coward]

I agree. If you want to support Asterisk then buy a Digium card. I can't believe that Slashdot would spit in Digiums ace by advertising that card.

Are there any FREE (speach) ones?

[aliquis]

So, I tried to find anything you could use with a headset, ventrilo server? no problem, allthought client was Windows only. Skype? Only Windows and PocketPC. Teamspeak? Only Windows and Linux and required a lot of weird RPMs to be installed on my beautiful freebsd box so no thanks. So, the question is, are there anyones which actually is FREE and works? I don't care for this proprietarian(?) crap which only run on a single os.

Re:e164.org is the same as alternic(mod parent up)

[Lennie]

I'm disappointed the Slashdot editors didn't notice that ...

you are new here, right ?

Encryption?

[pestie]

I can't seem to find a good answer as to whether SIP phones support encryption or not. I don't feel too terribly comfortable with transmitting my voice conversations unencrypted over the net. From what I've read SIP phones actually use a combination of SIP to initiate sessions and RTP to move the actual voice data, but no straight "yes, it is" or "no, it isn't" answer on whether encryption is supported or not.

Re:Encryption?

[Anonymous Coward]

Not right now. SRTP (secure realtime transport protocol) is the protocol you're looking for. Very few clients are capable of using it.

Re:Encryption?

[awehttam]

The answer, is basically, no. Or rather, not yet.

SIPphone.com has cheapest SIP hardware

[Anonymous Coward]

you can pick up an adapter for just $50 or so.

They also have a "Call in One" which allows you to merge a PSTN phone line and a SIP line so you can send and receive calls SIP or PSTN calls from the same phone system.

SIPphone.com

They also allow free PSTN calls from their phones, but it is limited to 5 1 minutes calls each day. Still, that's better than anyone else in the world.