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VoIP Regulation, SIP Insurrection

michael posted more than 9 years ago | from the insurgents-vs-incumbents dept.

Communications 117

Chris Holland writes "As voice communications are evolving beyond traditional phone systems and making better use of the Internet, Aswath Rao is offering regulation-advocating counterpoints to Dr. Daniel Ryan's original analysis of various VoIP industry players' arguments for deregulation. Many of the above discussions revolve around closed, regulatory-scrutiny-fostering voice communications ecosystems reserved to a small, resourceful elite. Meanwhile, an open Internet protocol which provides support for all forms of real-time communications including Text, Voice and Video, with a few open-sourced server implementations and free client solutions is starting to gain serious ground: The Session Initiation Protocol enables just about anybody with little resources to become their own Real-Time Communications Giant."

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117 comments

FP (-1, Offtopic)

Anonymous Coward | more than 9 years ago | (#11431820)

FP MCK

It's the asterisk drinking game! (3, Funny)

numbski (515011) | more than 9 years ago | (#11431821)

Okay, here's the rules.

Every time someone mentions the word "Asterisk" in this page, you have to take a shot. ;)

(Note that I'm building 2 of the 'A' Boxes right now. One for my home, and one at the office, a third will go at the ISP.)

Re:It's the asterisk drinking game! (0)

thegameiam (671961) | more than 9 years ago | (#11432206)

Asterisk Asterisk Asterisk Asterisk Asterisk Asterisk Asterisk Ashteritzk Ashtreick Ashtrick Ashkick Ashtray Ash... too drunk... must stop...

Re:It's the asterisk drinking game! (1)

wankledot (712148) | more than 9 years ago | (#11432218)

3 so far, and it's not even 8am... *hick* it's going to be a long day at work if I have to keep this up. Good thing it's Friday.

Re:It's the asterisk drinking game! (1)

bored_lurker (788136) | more than 9 years ago | (#11432221)

Well, I've been reazing - reabing - reading these post for a while now and... and... and... I'm not as think as you drunk I am.

Re:It's the asterisk drinking game! (-1)

Anonymous Coward | more than 9 years ago | (#11432348)

As my dear fathhherr said: i'm not hafl thunkl drinkl peep I am. I've only had one tee many martooonies...

how long you been playing already today? (1)

way2trivial (601132) | more than 9 years ago | (#11432572)

I'm building 2 ...... a third will

hoisted a few already today have ya?

Re:It's the asterisk drinking game! (0)

Anonymous Coward | more than 9 years ago | (#11434212)

Crap. out of ammunition already.

Want to get started with free VoIP... (0)

Anonymous Coward | more than 9 years ago | (#11431851)

I want to get started with free VoIP. What clients are best for windows and for linux?

Re:Want to get started with free VoIP... (1)

R.D.Olivaw (826349) | more than 9 years ago | (#11431910)

try Skype [skype.com]

Have to agree (1)

grahamsz (150076) | more than 9 years ago | (#11431995)

I've used a lot of VoIP services

Skype just works

I can take my laptop to work and it just works and figures out appropriate proxy settings.

My cisco hardware seems a lot harder to get working and keep working.

Re:Have to agree (2, Interesting)

Afrosheen (42464) | more than 9 years ago | (#11433431)

Sucks going with a proprietary, closed vendor sometimes. We've been very happy with our Sip-enabled Polycom phones though, we have an office full of them now and they work like champions. Nobody has even noticed that there are no phone lines in the new cubes and that the handsfree is full duplex now. I like it when new tech makes you take things for granted.

MOD HIM DOWN TO HELL! (-1)

Anonymous Coward | more than 9 years ago | (#11432157)

Skype sucks, it's NOT OPEN SOURCE and its authors are CRIMINALS.

Re:MOD HIM DOWN TO HELL! (0)

matthiasvangorp (576254) | more than 9 years ago | (#11432259)

boohoo

AC Troll (0)

Anonymous Coward | more than 9 years ago | (#11432877)

Care to give an account to that post, you little AC troll?

SIP behind Nat (4, Interesting)

Albanach (527650) | more than 9 years ago | (#11431871)

Sip works well, but doesn't like NAT'd connections though it can be made to work. IPv4 and forcing customers to use NAT are the technologies that will continue to be used to keep provision of a lot of these technologies in the hands of the ISP's with the potential to bill customers.

The ability to circumvent NAT is why programs like Skype have such popularity and why Linux users looking for more control have been quick to investigate Asterisk and it's IAX2 protocol.

Open standards are all very well, but for the time being at least, SIP is going to be a good technology so we can connect our computers to big carrriers and interoperate with the POTS. Other technologies have the potential to completely circumnavigate POTS and the big carriers - you cna bet your life they'll do everything they can to make sure they're not adopted.

Re:SIP behind Nat (1)

luvirini (753157) | more than 9 years ago | (#11432173)

Well, our quite Small( Recently we embarked on a trial project to connect directly to some of the people we do a lot of business with. We sent out an inquiry about 2 months ago to around 100-120 companies and if I am correct(not directly involved with day to day on this) allready few (10) dial rules go directly to some other company's PBXs bypassing the POTS.

Re:SIP behind Nat (2, Interesting)

luvirini (753157) | more than 9 years ago | (#11432215)

(argh html formatting, disregard previous)

Well, our quite Small( less than 250 employees) but international(18 countries) company is allready circumventing the POTS systems a lot. We actually have soft PBX in all our locations and thus allow us to talk within the organisation without charges. Also the callout rules use a combination of local calling from nearest office and VOIP terminations.

Recently we embarked on a trial project to connect directly to some of the people we do a lot of business with. We sent out an inquiry about 2 months ago to around 100-120 companies and if I am correct(not directly involved with day to day on this) allready few (way below 10) dial rules go directly to some other company's PBXs bypassing the POTS.

Re:SIP behind Nat (4, Insightful)

wolf31o2 (778801) | more than 9 years ago | (#11432713)

This is one of the primary reasons for dumping IPv4 and going IPv6.

I have been working on setting up my own IPv6 network. I am even investigating the possibility of getting true native IPv6 addressing along side IPv4 from my ISP.

The real problem for us is going to be all of the jokers out there that are so short-sighted that they ignore IPv6 claiming that "IPv4 and NAT are good enough for anything you want to do."

Well, those people are simply wrong. There are lots of reasons for IPv6. Cheap, or even free, global phone service is just one of them. Let's all work to re-establish the Internet as the peer-to-peer network that it was originally, and not the client-server network where the content is provided by big business and multi-national media conglomerates.

Re:SIP behind Nat (1)

MightyMartian (840721) | more than 9 years ago | (#11433339)

I agree with you completely. Now write a letter to my network's upstream provider and tell them

Re:SIP behind Nat (0)

Anonymous Coward | more than 9 years ago | (#11433745)

please mod parent up--the adoption of IPv6 /is/ a benefit to us all.

Re:SIP behind Nat (1)

FireFury03 (653718) | more than 9 years ago | (#11435114)

This is one of the primary reasons for dumping IPv4 and going IPv6.

I have been working on setting up my own IPv6 network. I am even investigating the possibility of getting true native IPv6 addressing along side IPv4 from my ISP.


I too have been using IPv6 for a while, unfortunately Asterisk currently doesn't support it.

You don't actually need a native IPv6 connection from your ISP - you can get away with using 6-to-4 dynamic tunnelling, which is what I do. Infact a big problem with rolling out IPv6 naviely is that I am not aware of any consumer grade DSL routers that support IPv6, so the PC to ISP connection at least would have to be tunnelled. Of course I'm hoping most ISPs wake up soon - if the ISP I use installed their own 6-to-4 gateway (and preferably advertised it using the anycast address) then I would be very happy.

By far the fastest way to roll out IPv6 would be to have the next version of Windows configured to use it by default. The shear number of people who wouldn't know anything about it would increase the size of the IPv6 network massively.

Re:SIP behind Nat (1)

ae (16342) | more than 9 years ago | (#11435793)

Infact a big problem with rolling out IPv6 naviely is that I am not aware of any consumer grade DSL routers that support IPv6, so the PC to ISP connection at least would have to be tunnelled.

If you receive real, globally routed IPv6 addresses and your ISP behaves as it should and gives you a /48 subnet or similar, you won't need a router of your own. Just connect all your machines to a switch and let them get their own addresses from your ISP's DHCP server. If you think you need a firewall, it should reside on each host anyway, where it does the least damage.

Re:SIP behind Nat (1)

nikkoslack (739901) | more than 9 years ago | (#11432988)

I totally agree that IAX2 is the next big protocol. We can't run SIP anywhere but on the local network due to several issues: Security: SIP requires many ports to be open IAX uses one port, and eve trunks multiple calls on one connection. NAT: SIP does a terrible job traversing consumer firewalls. Overhead: SIP - lots IAX2 lost less ALso, I run 3 asterisk boxes in a production environment. uptime is measured in months

Re:SIP behind Nat (2, Informative)

valmont (3573) | more than 9 years ago | (#11433041)

SIP was recently made to work behind NAT just fine thanks to STUN. read the article 'till the end. STUN was introduced in 2003, while SIP's been around for nearly a decade. I've even recently pushed the envelope to verify how well STUN works by making and receiving SIP calls from/to my earthlink SIP account behind 2 layers of NAT: 192.168.1.* network, linked to a 10.0.0.* network, linked to my earthlink (verizon) dsl.

Re:SIP behind Nat (1)

MightyMartian (840721) | more than 9 years ago | (#11433372)

With Linux iptables, it's reasonably easy to write helper apps for those protocols (IRC and FTP come to mind) that bust in a NAT firewall. I don't know anything specific about SIP, but I'm sure that a helper app could be written for those guys running 2.4 or greater Linux firewalls.

Re:SIP behind Nat (1)

Moofie (22272) | more than 9 years ago | (#11434185)

I know the REAL way to make money off of this deal. Come up with the next acronym, and copyright it. I'm becoming more convinced that there's a savant somewhere with a serial port grafted into their skull who comes up with all these damn things.

Re:SIP behind Nat (1)

Snocone (158524) | more than 9 years ago | (#11433130)

There's a variety of ways to get around the NAT/firewall issues, but to completely eliminate them under all possible circumstances you pretty much need to have a server at a dedicated public IP. It just so happens that there is one out there called X-Tunnels, and it's open source too, which Xten of X-Lite/X-Pro/eyeBeam SIP softphone fame has made available here:

http://www.xtunnels.org/

which you could always look into if you're trying to set up a genuinely universally accessible from absolutely anywhere at all SIP network.

Disclaimer: I wrote it. So I might be a bit biased here.

Re:SIP behind Nat (1)

FireFury03 (653718) | more than 9 years ago | (#11434993)

The ability to circumvent NAT is why programs like Skype have such popularity and why Linux users looking for more control have been quick to investigate Asterisk and it's IAX2 protocol.

I think IAX2 is definately the way forward because of it's external simplicity (one fixed UDP port carries everything).

I believe Skype uses a TCP session to carry the traffic, which makes it a fundamentally bad design (not to mention closed and propriatory). Unfortunately it's easy for complete eejuts to set up and they have good marketting so they're getting some business. (of course there is nothing that they're doing which a IAX to PSTN gateway couldn't do with some decent software and a marketting budget).

Speaking of which, does anyone know of any decent IAX2 or SIP softphones for Linux? ATM I'm using IAXComm which is not without it's problems. Kphone was unstable and almost totally feature-free when I tried it. GnomePhone is IAX1, not IAX2. GnomeMeeting is H.323 only (although they've been saying the next version will have SIP for the past year)...

(I'm also looking around for a Symbian UIQ softphone to run on my P900 but as far as I can tell, none exists).

Open standards are all very well, but for the time being at least, SIP is going to be a good technology so we can connect our computers to big carrriers and interoperate with the POTS.

In the end there will be no such thing as a telephone service provider - the PSTN will eventually die so there will be no need to gateway to it and then it's only a matter of time before everyone using Skype, etc realises they can do everything with direct peer-to-peer communications and skip the 3rd party completely.

The whole phone system is likely to move to a setup similar to email, where you will just enter an email style address and it'll connect you (we will need something similar to MX records in the long run I think). We're already part way there - i.e. you can call IAX2/pabx.nexusuk.org/slashdot to get to the speaking clock on my Asterisk server. :)

It should be great... (4, Insightful)

chris09876 (643289) | more than 9 years ago | (#11431907)

The whole VoIP technology has the ability to revolutionize communications. We just need to make sure that the industry is kept open enough, so everyone has a chance to innovate. Open source and open protocols are an excellent way to help do that. If the government steps in and starts regulating everything like they did with POTS, then we'll end up with a few huge monopolies that offer horrible service and horrible prices again.

What's the big deal? (1)

jaymzter (452402) | more than 9 years ago | (#11431952)

I am at this moment sitting in a class covering my company's SIP enabled devices (fortunately running on Linux), but I have yet to see the big deal.
Honest question, what does SIP, an all in one protocal, offer you that traditional implementations don't?
Note: I'm not referring to home users, so please no replies about calling porn services in Rumania for free :)

Re:What's the big deal? Thanks dude! (-1)

Anonymous Coward | more than 9 years ago | (#11432105)

Note: I'm not referring to home users, so please no replies about calling porn services in Rumania for free :)

Thanks for the tip, dude! Bored out of my mind I was just thinking what I was going to do this fine Friday evening. Any Romanian numbers and sites you can recommend?

Re:What's the big deal? (1)

walt-sjc (145127) | more than 9 years ago | (#11432807)

What do you mean by "traditional implementations"? Proprietary PBX systems like a Nortel or Seimens? Or other VoIP protocols? Or a closed campus that has no other off-site connectivity other than traditional phone service (POTS / PRI, etc?)

I guessing "proprietary systems..." If you think about it for more than 5 seconds or so, or haven't been hiding under a rock for the past couple years, the answers should be obvious. Flexability, open systems, and cost savings are the top three.

Re:What's the big deal? (1)

FireFury03 (653718) | more than 9 years ago | (#11435294)

Honest question, what does SIP, an all in one protocal, offer you that traditional implementations don't?

Ok, I think IAX2 is a far better protocol than SIP because it's not as complex from the networking point of view, so this reply will be based on VoIP in general rather than specifically SIP.

There are 2 areas to consider, the first is an internal (e.g. office-wide) phone system and the second is a replacement for the PSTN:

Office phone system:
1. Less cabling infrastructure - instead of separate cables for phone and data you can run both down the same wiring. This is a big deal in large buildings.
2. You don't necessarilly need to invest in actual physical phones, you may find it advantageous to have a softphone on your workstation instead.
3. People who are out of the office can log into the phone system from home/hotel/wherever and use it as if they were on an internal extension.
4. You can hook all your branch offices into the main phone system over the Internet.
5. I'm sure there are more advantages :)

As a PSTN replacement the big deal is that there is no phone company involved unless you're having to gateway to the PSTN, thus no call charges. This is really good news for anyone who spends vast amounts of money on calls. And even if you're gatewaying to the PSTN you can do least-cost routing to gateways near the destination. i.e. if I run a business with customers in the UK, US, Australia, etc. I could subscribe to gateways in those countries and route the calls appropriately, which is likely to be way cheaper than paying BT to carry international calls.

At some point in the future (hopefully soon), the PSTN won't exist at all - you will phone people over the internet using nice easy to remember phone addresses in the same way as you use email addresses. You won't pay anyone to carry those calls, just your usual internet bandwidth charges. You can already do this now if you're phoning the right people. e.g. you can phone the speaking clock running on my Asterisk server by calling IAX2/pabx.nexusuk.org/slashdot

SIP offers light weight... (1)

msauve (701917) | more than 9 years ago | (#11435712)

"traditional" VoIP implementations, especially H.323 are overly complex. H.323, for instance, was originally intended for videoconferencing, and was a in international "design by committee." "Traditional" VoIP carries a heavy resource cost, making it expensive to implement on light weight devices (think PDA, cell phone, other personal/portable stuff).

SIP can be implemented with a much lower resource requirement.

If you're familiar with IP and OSI protocols, think CMIP (H.323) vs. SNMP (SIP).

Spam (2, Interesting)

awhelan (781773) | more than 9 years ago | (#11431967)

After reading the blog entry, VOIP looks like it is very susceptible to spam. Some of the limits of telemarketers today are paying to make the calls, and accountability. New spammer/telemarketers could use a semi-anonymous SIP address.... or use a virus to control someone else's and send out millions of bulk recorded messages. Also, spam detection software to prevent something like this would be infinately more difficult to create than email filtering software.

Re:Spam (2, Interesting)

luvirini (753157) | more than 9 years ago | (#11432418)

Actually this might help in reducing spam if properly implemented.

As atleast all the "real" revices are programmabel, you just give a voice menu that a human can easily select past.

"You have called the residence of (insert name), the calls here are subject to licence agreemennt, Press 1 to accept the lisence, press 2 to listen to the lisence or hang up."

On 1 it connects.

on 2 it says something like "This is a legal agreement between you, the caller and (insert name), the called party. if you are trying to sell a product or a service, you must provide full company and personal details and to present the product truthfully. You will be billed 20 Dollars a minute for the call. Any lie or omission of fact on your part will result in a 500 dollar fee for for each such instance. By continuing this call you are accepting these conditions. if you do not jave the authority to accept contracts by phone, hang up now." And then continue about everything else you can think of..

Thus everyone can just press 1 to continue and the phone will ring as normal, but try getting a spam to do that.. and all the telemarketeers are promising to pay you 20 dollars a minute to listen to them.. :)

Better yet... (-1, Offtopic)

Anonymous Coward | more than 9 years ago | (#11431974)

Why don't you get the FUCK back to work!

data of VOIP (1)

mzwaterski (802371) | more than 9 years ago | (#11432000)

This may be a silly question, but can you do data for VOIP? I guess what I mean is in relation to a call originating with VOIP and ending at a modem on a POTS. Granted, it would be stupid to go from a high speed digital network to a slow analog telephone system, but is there any way to do this? It would be similar to a VPN type network connection with a virtual VOIP modem.

Re:data of VOIP (3, Informative)

stratjakt (596332) | more than 9 years ago | (#11432022)

Yes, you can send and recieve faxes and dial-out via modem over VOiP.

I dial out over Vonage all the time, since the only access to most of the boxes I support is via dial-up. There are still plenty of computers that aren't on the 'net, especially where privacy/security is key.

Re:data of VOIP (1)

mzwaterski (802371) | more than 9 years ago | (#11432181)

Oh, I realize how silly the question is now. As long as you have a converter box to interface your rj11 connection to the pc you are good. Thanks!

Re:data of VOIP (1)

walt-sjc (145127) | more than 9 years ago | (#11432953)

No. This is not the case. You need to have both an ATA and service providor that supports the very new and rarely implemented standards that allow modems to work.

Re:data of VOIP (1)

walt-sjc (145127) | more than 9 years ago | (#11432930)

So what kind of speeds do you get??? For faxes, you have the T.38 protocol that allows them to work (requires support at both the VoIP provider AND the ATA you are using). Getting modems to work over 9600 is Much more of a trick. First, you can't use any codec that does compression so it sucks a lot of bandwidth, and second, the latency and packetization of the modem signal is going to be quite problematic. See this page [voip-info.org] for more info on modems over VoIP.

If you can get your modem to work at all over VoIP, good for you (I am VERY surprised to find that someone is using it succesfully.) It doesn't work at all for most people at this time however.

No one cares... (1, Troll)

astebbin (836820) | more than 9 years ago | (#11432013)

Because everyone is sitting in front of their computer with their IM client of choice.

Re:No one cares... (3, Interesting)

faedle (114018) | more than 9 years ago | (#11433204)

The irony, of course, being that SIP has MORE [com.com] THAN [ietf.org] ONCE [ietf.org] been suggested as a replacement open IM protocol...

Hooray! (3, Interesting)

drewzhrodague (606182) | more than 9 years ago | (#11432026)

Friend of mine called me from his Asterisk box last nite -- I picked up the call on my cell phone. His voice was clear, crisp, unjittered, no echo -- sounded like he was on a landline handset.

So, I'm now experimenting with Asterisk...

Re:Hooray! (1)

grasshoppa (657393) | more than 9 years ago | (#11432201)

May I recommend you look at http://connect.voicepulse.com if you will be playing with asterisk.

$50 Open Source Wifi SIP Server! (1, Informative)

Anonymous Coward | more than 9 years ago | (#11432035)

These guys [sveasoft.com] are upgrading a $50 Linksys router [linksys.com] with a full SIP server and SIP NAT. Add a wireless Wifi phone you have your own wireless PBX for the house including Wifi, QoS, a killer firewall, and tons more to boot.

And it's based on Linux and open source - whoopee!

Re:$50 Open Source Wifi SIP Server! (1, Interesting)

Anonymous Coward | more than 9 years ago | (#11432403)

It's not quite open source and they are selling it and are happy to use the DMCA to stop anyone from distributing it, I'm too tired to explain, just RTFA here [chillingeffects.org] .

Re:$50 Open Source Wifi SIP Server! (1)

walt-sjc (145127) | more than 9 years ago | (#11433306)

Now you just need to find a WiFi SIP phone that doesn't totally suck. Good luck!!!

OPENWRT.ORG + WRT + ASTERISK (1)

bobsalt (575905) | more than 9 years ago | (#11434124)

use openwrt.org. People have gotten asterisk to build on it and have even made packages for it. more interesting will be the new linksys with the fxo ports on it. They had some oem in the beginning, but now you have to get it bundled with vonage...

In related News... Michael Powell is... (0)

Uptown Joe (819388) | more than 9 years ago | (#11432050)

Stepping Down as head of the FCC... Read all about it: http://online.wsj.com/article_email/0,,SB110627220 789332234-INjfYNilaJ4nZ2pZIKIcKWHm4,00.html

Re:In related News... Michael Powell is... (1)

Arngautr (745196) | more than 9 years ago | (#11432420)

Or for those who don't feel like logging in, and want a clicky link, clikicky clicky. [wsj.com]

Resources? (1)

dsginter (104154) | more than 9 years ago | (#11432053)

The Session Initiation Protocol enables just about anybody with little resources to become their own Real-Time Communications Giant.

What if I have modest resources? Can I still become my own real-time communications giant? /sarcasm

Don't take this guy as the word of god.... (1)

PepeGSay (847429) | more than 9 years ago | (#11432099)

I worked for a Cable company doing a VoIP rollout and I can assure you they do not as he says in #3: "Note that these players not only do not object, but they want regulatory parity with ILECs, because that is their competition." We very cleary wanted parity or to have a regulatory advantage. With the regulatory advantage being much preferred.

Re:Don't take this guy as the word of god.... (0)

Anonymous Coward | more than 9 years ago | (#11435316)

I guess nobody's word should be taken at face value. But ...

I am basing my comment based on statements made by executives of cable companies in public fora. For example, in the FCC Forum of December 2003, John Billock of Time Warner stated that they view their offering to be a facilities based offering and that they are willing to play within a regulatory regime. THe actions by other cable providers go along with his sentiment. For example they are not pricing it compaetitive to other VoIP services, but on par with the incumbents.

Aswath (Anonymous because lazy to open an account)

On becoming my own communications giant (1)

$RANDOMLUSER (804576) | more than 9 years ago | (#11432191)

As noted before, here [slashdot.org] (FCC Asks For Comments On Internet Wiretapping), and here [slashdot.org] (FCC Rules VoIP Must Be Tappable), the federal Communications Assistance for Law Enforcement Act (CALEA) mandates law enforcement "back doors" on these networks to allow wiretapping.

VOIP is the leading edge of big government/big moneys effort to quell the anarchy that is the Internet.

Magic Beans (3, Interesting)

Bookwyrm (3535) | more than 9 years ago | (#11432196)

The Session Initiation Protocol enables just about anybody with little resources to become their own Real-Time Communications Giant.

And anyone with a hoe and a little water can become a Real Farming Industry Giant! Or, If You Have A Few Bucks, You Can Buy This Bridge I Can Sell You.

The ... protocol (sic) does not function as a magic bullet. Just waving the SIP spec at a traditional telcom does not knock them over. (Okay, throwing the entire printed version of all the SIP specs might...) This isn't about anyone with just 'a little resources', this is about people with resources, a lot of technical know-how (SIP is easy only in the sunny day cases), and LOTS OF TIME.

Re:Magic Beans (1)

valmont (3573) | more than 9 years ago | (#11433100)

i'm trying to open minds here, and i know exactly who the target audience here: geeks. The point i'm trying to make is that SIP services are that much harder to set-up as SMTP/POP services, and now that SIP was made to work behind NAT thanks to STUN, and that you have free, open-source implementations of SIP presence/registration servers and STUN servers, it should be QUITE POSSIBLE for anyone with a bit of determination to at least provide SIP services to themselves, even to their friends, and/or add SIP services to an ISP they may already be running, or web hosting services they may already be running. try n' read the article. I'm not suggesting the telcos will be replaced any time soon. I'm merely pointing out that SIP provides for an open real-time communications framework, and we geeks would be silly to not take it for a ride. earthlink did it. pulver did it.

Re:Magic Beans (1)

Bookwyrm (3535) | more than 9 years ago | (#11433885)

The audience here is also system and network admins. Setting up SIP may be about as hard as setting up SMTP/POP service (if I can decode your syntax properly), but diagnosing, debugging, and maintaining SIP is far, far harder and less forgiving with the current state of things. It's one thing to 'take it for a ride' for yourself or friend or for an ISP, it's another thing to 'clean up after it, keep it secure, keep it running month after month, and do tech support' for yourself or your friends or an ISP.

For the first time in history... (1)

bs_02_06_02 (670476) | more than 9 years ago | (#11433841)

For the first time in history, those with the time and a bit of know-how can do it. It's possible. And if the government stays out of it, it's a real grass roots threat to the big corporations.

Legislators are scared of this. Successes in ventures like this prove that we don't need legislators and regulators like they think we do. Legislators want to leave their legacy. They want to make themselves important, justify their own existence. They want to pat themselves on the back and say that they made government work!

Legislators and big government like to, with the media's help, paint a big picture of pirates, desparation, big evil corporations that need regulating, rip-offs, and even death at the hands of unregulated technology. Just you wait and see... the first time someone dies waiting for an ambulance and they don't have a landline or wireless line, and if it's discovered that they had a free internet phone connection, you can't imagine the press coverage that will flood this topic. 60 Minutes, the NY Times, the Washington Post, the Reverend Jesse Jackson... they'll ALL step in to talk about the horror, the sorrow that this poor soul faced, and that we need legislation to help prevent this type of accident, to prevent the treachery that resorted to this person turning to a "pirated" type of internet phone. You'll see 20% taxes on your internet connection within WEEKS! ISPs will be forced to put up filters. Trust me. It'll happen.

But if we're fast about it, and if it happens quickly, it'll go too far and reach too many people before the government can react. TIVO was something like this. TIVO changed people's habits. They aren't tied to the broadcast schedules for their shows. They skip thru commercials. Commercials are now served inline. You don't realize it, but TIVO changed a lot more than people think. SIP and VoIP, if it picks up pace, could do far more. Soon, wireless connections won't just be a phone, it'll be IP, and you'll end up with VoIP over a wireless phone. It's already started. We just need to keep the government out. Get the stupid "industry" lobbyists, regulators, press, and party hacks and keep them locked up somewhere. A year, maybe 2, and let it grow unchecked.

Re:For the first time in history... (0)

Anonymous Coward | more than 9 years ago | (#11433991)

goodness.

For the first time in history, those with the time and a bit of know-how can do it.


cue lighting and dramatic music! first time evar that people who know what they're doing and have time to do something can actually do something.

we need a new moderation "-1 Melodramatic" or "-1 Milking the Giant Cow"

Re:Magic Beans (1)

Frank T. Lofaro Jr. (142215) | more than 9 years ago | (#11434758)

SIP isn't a magic protocol, there is only one magic protocol, and that is XML. :)

SIP over XML might be a magic bullet. ;)

Re:Magic Beans (1)

Bookwyrm (3535) | more than 9 years ago | (#11434826)

Actually, there was a lot of talk about doing an XML encoding for SIP. (Actually, I think the next-gen SDP specification is in XML.) However, to the best of my recollection, that idea triggered lots of religious schisms and convulsions among the SIP faithful.

I'd go with Jabber and enhance it with some voice signalling specific tweaks/messages, probably, before trying to convince the SIP True Believers about doing an XML conversion.

No 9-1-1 (2, Interesting)

ebbyfish (759832) | more than 9 years ago | (#11432284)

VoIP (and similar technologies) does not provide any address information when you call 9-1-1 (I know neither do PBX's, but most people do not have one of those in their houses). That is a really big issue if someone reports his or her address wrong to the 9-1-1 Dispatcher (it happens all of the time, all over the country - I call this the grey side of innovation). Deregulation certainly has its pluses, but what are they worth if you or someone you know doesn't get they help they need? There is a public perception that 9-1-1 will come to your aid if you call them, many people were taught this as children. If these VoIP companies choose not to address this issue, then where does that leave the whole EMS system? How can they assist the public if they do not know where they are calling form? Just some thoughts.

Re:No 9-1-1 -wrong (1)

bored_lurker (788136) | more than 9 years ago | (#11432399)

Um, sorry wrong. Packet8 does [packet8.net] . It is not a big deal to me but if it was I would have gone with them.

Re:No 9-1-1 -wrong (1)

ebbyfish (759832) | more than 9 years ago | (#11432462)

I had no idea, I am involed in the EMS industry, which is why I posted this. But I have never heard of Packet8 before. Maybe in time their service model will be followed by others. Thanks for the tip.

Re:No 9-1-1 -not all 9-1-1 are the same (1)

bored_lurker (788136) | more than 9 years ago | (#11432940)

Not that while many of the child posts point out that many VoIP providers have 9-1-1 they are not all the same. Take Vonage for example [vonage.com] (whom I use). Their 9-1-1 is routed to the PSAP and is not true E911 service. This distinction may be lost on many but what it effectively means is that E911 centers get an address that pops up on their screen when you call them. With Vonage you address may or may not pop up on the operator's screen.

The only service from a major VoIP provider that I am aware of is the afore mentioned packet8. I'm sure this change over time.

BTW, Packet8 charges a $10 setup fee and a $1.50 monthly fee for E911 (RBOCs also charge for this service - but you have no choice)

Re:No 9-1-1 (3, Informative)

Big_Al_B (743369) | more than 9 years ago | (#11432466)

I don't know where you've gotten this "No 911 with VoIP" idea from.

I work for a telco/ISP/VoIP provider, and we've offer 911 services standard with all VoIP services. It's the same E911 [fcc.gov] service that cell carriers are providing.

And most major VoIP industry players offer it as a standard, or at least optional, feature.

Cell carriers are legally bound to provide E911 services (stage 1). VoIP carriers are not, but most serious providers do anyway, to have feature parity with the POTS market.

Re:No 9-1-1 (1)

hab136 (30884) | more than 9 years ago | (#11432482)

VoIP (and similar technologies) does not provide any address information when you call 9-1-1

That used to be true. Vonage supplies your address to 911 [vonage.com]

Re:No 9-1-1 (1, Informative)

Anonymous Coward | more than 9 years ago | (#11432781)

You misread, or posted the wrong link. What that page says is that you need to verify your address with Vonage, and based on that information they'll route your 911 calls to the answering center that covers your address. They do not send your address to the 911 answering center, and in fact your Vonage 911 call doesn't necessarily go to the same lines as POTS 911 calls do.

I believe Vonage is doing a trial of E911 in Rhode Island. E911 means full POTS 911 features, including supplying your address to the 911 center.

Unrelated:
E911 is also slowly being installed for the cellular network -- instead of an address, your phone receives GPS signals and passes those to the cell tower for computation of your position. I believe the towers also do some triangulation and signal-strength calculations to help determine position.

Re:No 9-1-1 (1)

company nuncio (29090) | more than 9 years ago | (#11432519)

There's nothing about VoIP that keeps it from working with regular or enhanced 911 services - as noted Packet 8, Lingo, and Vonage all do stadard 911, as will others. In these cases it does depend on the user entering the correct service address in the provider's database. Newer services will do e-911 with more robust location handling, but any of them can be compromised by the user moving the TA from his or her home to another site.

Re:No 9-1-1 (2, Insightful)

Gaewyn L Knight (16566) | more than 9 years ago | (#11432674)

Incorrect!

VoIP companies can and do provide E911 addressing. Vonage for example has a web page that you can tell them your home address and that will be sent with any calls to 911.

The only place where VoIP does have a downfall in this area is for wireless VoIP phones. Since these phones have no idea where they are your company will be providing your home address as the 911 address even if you are in a hotel halfway around the world.

Hence we hear the cry "Put GPSs in all of them like newer cellphones". Only problem with this is that most of these are used indoors and GPS signals are horrible at penetrating structures.

Then we hear the cry "Place the location in all the access points". Once again the problem with this is... you have to tell the access point the location and you know that someone is going to forget to set/change it. This would also take a LARGE overhaul of how access points work since they would then have to inject this location into the datastream as it passes instead of being a passive bridge of the data.

I think people are just going to have to be taught what the inherent risks are and how they can avoid them beyond that we start getting into the stupid world of blatantly obvious warning labels and such.

Everything has an inherent risk... learn to deal with it people!

Re:No 9-1-1 (0, Redundant)

spacefrog (313816) | more than 9 years ago | (#11433335)

Bullshit.

I have Vonage, and we most certainly do have 911 service. In the case of Vonage, I can directly tell them the EXACT address that the phone is currently located at.

This is important to me, since I have a California area code and billing address, but the phone is in Washington right now.

Re:No 9-1-1 (2, Informative)

Scyber (539694) | more than 9 years ago | (#11433982)

Wrong, I have Vonage too. And if you read their page: http://www.vonage.com/features.php?feature=911 [vonage.com] They even tell you the following:
Your Call Will Go To A General Access Line at the Public Safety Answering Point (PSAP). This is different from the 911 Emergency Response Center where traditional 911 calls go.
This means that your address does not automatically appear on the Call Centers computers. Currently only Packet8 offers this feature. Although I heard that Vonage is beta testing in some markets.

Re:No 9-1-1 (1)

jwilhelm (238084) | more than 9 years ago | (#11435033)

Rhode Island is the only market where E911 is fully implemented by Vonage. Packet8 is much farther along.

Phone number portability (1)

timeOday (582209) | more than 9 years ago | (#11435416)

Thanks for the link. I just followed it, and was poking around, and according to vonage I could transfer my current home phone number to VOIP service!

Now this is something I'm really going to consider.

Re:Phone number portability (1)

timeOday (582209) | more than 9 years ago | (#11435468)

Here's the link [vonage.com] to see if you can keep your number.

No accountability (2, Interesting)

Omega (1602) | more than 9 years ago | (#11433993)

Here's why a lack of regulation for VoIP is A Bad Thing(TM). When you pick up a phone using POTS you always and immediately get a dial tone. If your phone service goes out for any reason, you can contact the Public Utility Commission and they will be on the phone company's ass right away. If your VoIP goes out, you have no recourse. Not to mention the fact ISP's do POP maintenance all the time -- I'm a little uncomfortable with knowing there's a time of day when I might not have phone service. When's the last time your phone company told you that phone service will be taken down between the hours of 11-12?

I'm not trying to impugn the technology -- I think VoIP is great, but if it's going to replace POTS, it needs oversight and regulation as a public service.

Re:No accountability (1)

Scyber (539694) | more than 9 years ago | (#11434370)

In the 2.5 years I had verizon POTS service in NJ, I had at least a dozen outages. One of which was over 24 hours. They apparently disconnected my line b/c they had no record of it being active. While I understand my experiences are unique and rare, it doesn't exactly leave a good impression of the POTS service. And I wish I knew that I could contact a Public Utility Commision, but I had no idea that even existed.

I use asterisk at home and at work... (1)

SlongNY (766017) | more than 9 years ago | (#11432634)

I use asterisk at home and work. It works flawless (once you have it setup correctly) Whats great is my home asterisk box communicates with the work asterisk box. At work we have two seperate offices one in LV and one in NY, those two box's communicate to each other.. In the same way for example broadvoice will probally hook up with packet8 to eliminate the middle man and save mula, so those calls to eachother will cost nada. Get a Free mini mac - http://www.freeminimacs.com/?r=14172807 [freeminimacs.com]

VOIP is too centered around service providers (1)

llefler (184847) | more than 9 years ago | (#11432667)

Why is it necessary to subscribe to any provider to get directory services? Sure, if you want inbound/outbound POTS service you need to subscribe to a gateway. But now that even grandma has the fancy new broadband, why can't we just make direct calls to other VOIP users?

I still think VOIP directories should be available through services like ddns. I don't have to subcribe to any service to do a DNS lookup so I can visit someone's website. Just think how much simpler life would have been if instant messaging had a standard protocol and public directory servers.

Re:VOIP is too centered around service providers (1)

Big_Al_B (743369) | more than 9 years ago | (#11432731)

I still think VOIP directories should be available through services like ddns. I don't have to subcribe to any service to do a DNS lookup

Your ISP runs a DNS service to support other services that they sell you. They do it because they want your money.

DNS is not "a public service." Except for, arguably, the roots.

Re:VOIP is too centered around service providers (1)

FireFury03 (653718) | more than 9 years ago | (#11435428)

I still think VOIP directories should be available through services like ddns.

They can and they are, so long as you're not using a propriatory system. The ENUM system lets you do exactly this (have a look at e164.org). You register your phone number with the system along with details of what VoIP protocol you use and the address of the VoIP phone (or PABX). That address can quite happyilly be handled by a DDNS system somewhere, and people can look up your number on the ENUM DNS servers and then use those details to make a VoIP connection.

But in the long run, there will be no need for this anyway because when the PSTN finally dies there will be no need for phone numbers - you will just have email style addresses.

SIP needs to opened up (5, Insightful)

akajerry (702712) | more than 9 years ago | (#11432699)


I think the biggest thing that the VoIP providers can do to avoid regulation is open up their SIP networks. And the best thing people like AT&T can do to get upstart VoIP players regulated is to open up their SIP networks.

VoIP get's most of the emphasis, but SIP is the killer app that VoIP is riding on, IMHO. The most annoying thing is that the VoIP providers won't allow customers, other VoIP providers or CPE (Customer Premise Equipment) manufactures access to the really cool features of SIP.

What can you do with truly open SIP. For starters it help to understand that SIP is a signaling protocol (like SS7 in the POTS world), not a communication protocol, SIP doesn't bother with encoding, decoding, or routing of the actually bits being communicated. As the name implies Session Initiation Protocol initiates communication session between end-points, once initiated the communication occurs direct between the end-point devices using some other protocol negotiated by SIP when it initiated the connection. However, the word "initiation" is a bit misleading because the SIP server also maintains awareness of the connection once established and can be used to control the connection afterwards and that can include adding/subtracting end-points, add/subtracting layers of communication, re-connecting end-points, etc. Very powerful stuff.

So with open SIP, you could have your cell phone route calls to the ATA in your home when you're home, but directly to your cell phone when away (and visa versa) by having the SIP server of your home ATA tell the SIP server of your cell phone provider that the new end-point device for phone number xxx is here. Also, you could set up complex multi-media connection on the fly. You're chatting over IM with someone and decide you need to up the bandwidth to voice, click, both parties (2 or more actually) phones ring, need to add a data feed to that to send a file, click. Need to add video, click.

The possibilities of what can be done with SIP have just barely been explored because of the limitation imposed by the VoIP providers. If only they understood Metcalf's law: The power of the network increases proportionately with the square of the number of nodes on the network. So by artificially limiting the number of nodes on your VoIP network to only your customers you really do yourself a disservice.

So if AT&T opened up its SIP network first and allowed users to see the power of SIP then the public sentiment could very quickly tilt in favor of regulation on other VoIP providers to do the same. On the other hand, if Vonage opened up its SIP network first then it could maintain the regulatory high-ground that VoIP inherently creates a competitive marketplace without regulations.

Re:SIP needs to opened up (1)

valmont (3573) | more than 9 years ago | (#11434577)

if you read my "fun and frolics" article (last link in the post) and the article prior to that, you'll see there are already quite a few pure SIP providers out there, including pulver.com/fwd and iptel.org, who both are free. i use earthlink's SIP services because it comes with my account, and i often converse with a buddy who's linked his pulver.com account to a home-bound asterisk PBX system. anyway, the articles list a few providers. I'm hoping more will rise.

Just Another Rich Boy's Toy ... (-1, Flamebait)

Anonymous Coward | more than 9 years ago | (#11432741)

... without broadband. As usual, the privileged and lucky have another toy to play with while many will never have the opportunity to use it because they don't have broadband available to them.

Re:Just Another Rich Boy's Toy ... (0)

Anonymous Coward | more than 9 years ago | (#11434885)


Out here in the real world beyond the borders of the USA (where citizen means more than "mark"), broadband is getting pretty darn common. Not a rich boys toy at all...

Simplest "free" SIP server for "personal" use? (1)

Dr.Dubious DDQ (11968) | more than 9 years ago | (#11433290)

Are there any simple (relatively speaking) SIP servers that can be pressed into service as a Voice-over-IP conferencing server, the way OpenH323's OpenMCU [openh323.org] can? I wouldn't really care that it was SIP, except that SIP seems to be the protocol with the greatest selection of open and/or free clients available at the moment.

I'm not thinking here of a full hook-your-telephone-to-the-internet system (which Asterisk seems to be ideal for), just a simple open-standards server for a few people to point their computer-based voice phones at, running on my OWN server, for a casual conference, using readily-acquired free/open software. I've gotten OpenMCU to work for that before, but H.323 seems like it is slowly being replaced by SIP (and there appear to be more SIP clients available than H.323 ones.)

SIP... or IPv6? (1)

nsayer (86181) | more than 9 years ago | (#11433305)

I haven't read the full spec, but from what I see, it sounds like SIP's main purpose is to be a workaround for NAT. Well, instead of that, how about adding support for IPv6? No NAT traversal required.

Re:SIP... or IPv6? (1)

Wesley Felter (138342) | more than 9 years ago | (#11434571)

No, SIP provides call signaling.

Re:SIP... or IPv6? (1)

valmont (3573) | more than 9 years ago | (#11434732)

no no no, please do read the full spec. and try'n'read the articles. SIP's purpose is not to be a workaround for NAT. in fact one of the reasons SIP hadn't had a chance to get many mainstream applications was because of NAT. In 2003, a full spec for STUN was released. STUN is a standard way to work around NAT. Pretty-much all SIP clients have support for STUN.

VoIP is still very much in its infancy (3, Insightful)

Jailbrekr (73837) | more than 9 years ago | (#11433412)

1) When your power goes out, the phone still works. Your computers (and VoIP phone) do not.
2) When your Network connection flakes out (as it is known to do periodically), your VoIP phone goes silent.
3) When your ISP starts to block or throttle back VoIP calls which are not routed through their own VoIP service, your VoIP phone is almost useless. You can thank the lack of regulations for this.

The VoIP industry is very much in bubble mode right now. It will burst, and when it does, I think that VoIP will finally have the opportunity to mature into a product which is actually useable for joe average.

Re:VoIP is still very much in its infancy (1)

jwilhelm (238084) | more than 9 years ago | (#11435082)


1) When your power goes out, the phone still works. Your computers (and VoIP phone) do not.


Not if your networking equipment and your ATA is on an UPS. I know I had a short outage shortly after moving in, and I was still able to use my Vonage service.


2) When your Network connection flakes out (as it is known to do periodically), your VoIP phone goes silent.


True, but most of the time the VoIP provider knows because it cannot contact the ATA, so it reroutes calls. I have my account set to route calls to my cell phone if the ATA is unreachable.


3) When your ISP starts to block or throttle back VoIP calls which are not routed through their own VoIP service, your VoIP phone is almost useless. You can thank the lack of regulations for this.

That is the one thing I am a little bit worried about. I know many ISPs are rolling out their own VoIP service, and they'll want to give that higher precendence over other providers' traffic. That being said, if I am paying for 768 upstream, I expect to get it. I guess this is where the free markets, and competition will kick in. If Provider A throttles back VoIP and Provider B does not, Provider A may lose customers.

Re:VoIP is still very much in its infancy (1)

valmont (3573) | more than 9 years ago | (#11435711)

i highly doubt we'll ever see VoIP replace traditional telephone. I think there'll always be a need for the traditional phone, as it was built from the ground-up to serve very reliable, mission-critical purposes, as a closed ecosystem, with checks and balances.

that doesn't make VoIP any less of a very nice complementary alternative, especially SIP whereby end-to-end SIP communications are 100% free.

Re:VoIP is still very much in its infancy (1)

mla_anderson (578539) | more than 9 years ago | (#11435871)

1) When your power goes out, the phone still works. Your computers (and VoIP phone) do not.

Ever hear of a UPS? My ATA is on the UPS with the DSL modem. If power goes out for too long I can always hook up an inverter to the car for emergencies, or just use the auto-forward.

2) When your Network connection flakes out (as it is known to do periodically), your VoIP phone goes silent.

Speak for your own network connection. I pay a little extra for a good ISP and get good reliability....but once again if it were to go down the auto-forward is still in place.

3) When your ISP starts to block or throttle back VoIP calls which are not routed through their own VoIP service, your VoIP phone is almost useless. You can thank the lack of regulations for this.

And so I chose an ISP with a good Usage Agreement.

If you want to use VOIP as your primary voice transport you have to do some more work to make sure your connection will support more than web browsing. Then VOIP works and works well.

I've been a Vonage customer since 2003, the only problem I ever had with them was with the initial implementation of voice mail. I've been a DSL Extreme customer since DirectTV stopped their DSL service (early 2003?), the only problem I've had with them was a single outage right before a CS match. VOIP is mature and ready for prime time...now if only someone would talk to the Vonage about those TV spots.

VoIP Regulation Panel Discussion in LA (1, Informative)

Anonymous Coward | more than 9 years ago | (#11434430)

Representatives from several VoIP companies will be discussing the future of open-source, regulation, and VoIP at SCALE 3x [socallinuxexpo.org] next month. Panelists [socallinuxexpo.org] will include Louie Mamakos, (Vonage), Jeff Bonforte (SIPphone), Al Brisard (PingTel), and Darryl Strauss (President - Digital Ordnance). In addition there will be talks about setting up your own VoIP systems with Asterisk.

UPload speed (1)

blahbooboo (839709) | more than 9 years ago | (#11435230)

I keep wondering, VOIP is never going to really be very good until ISPs increase the bandwidth on the UPLOAD side. Download is fine, but the uploads speeds are just terrible.

Microsoft Windows Messenger uses SIP (2, Informative)

daern (526012) | more than 9 years ago | (#11435969)

i don't suppose anyone on /. will mention it, but Microsoft have adopted SIP in the latest Windows Messenger client.

Note that this is *not* the same as the .NET Messenger client, which is designed for public Internet use. Windows Messenger is designed to work with a Live Communication Server, integrated into Active Directory and Microsoft Exchange. If you have the whole Microsoft suite, it actually works really well...

More info here... http://www.microsoft.com/technet/prodtechnol/winxp pro/maintain/wmsgrfaq.mspx [microsoft.com]

We use this for corporate IM, voice and video conferencing, as well as remote desktop support (using the "remote assistance" feature) and also for desktop application sharing.
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