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Communications The Internet

VoIP Regulation, SIP Insurrection 117

Posted by michael
from the insurgents-vs-incumbents dept.
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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VoIP Regulation, SIP Insurrection

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  • by numbski (515011) * <numbskiNO@SPAMhksilver.net> on Friday January 21, 2005 @11:13AM (#11431821) Homepage Journal
    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.)
  • SIP behind Nat (Score:5, Interesting)

    by Albanach (527650) on Friday January 21, 2005 @11:17AM (#11431871) Homepage
    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.

    • 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 (Score:3, Interesting)

      by luvirini (753157)
      (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 s

    • Re:SIP behind Nat (Score:4, Insightful)

      by wolf31o2 (778801) on Friday January 21, 2005 @12:28PM (#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.
      • I agree with you completely. Now write a letter to my network's upstream provider and tell them
      • 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 na
        • 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 resi

          • 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

            I don't know about you, but my ISP doesn't give my a nice ethernet connection to plug into. I have a DSL router which talks to the ISP and routes traffic from my local subnet (which is a real-world subnet) to the isp over the DSL. I know this varies from country to country, but here in the UK our DSL is
            • I don't know about you, but my ISP doesn't give my a nice ethernet connection to plug into. I have a DSL router which talks to the ISP and routes traffic from my local subnet (which is a real-world subnet) to the isp over the DSL. I know this varies from country to country, but here in the UK our DSL is entirely PPPoA which means that the DSL router really does need to understand the protocol you're using - most only understand IPv4 (i.e. noone has made a PPPoA bridge yet).

              In fact, my ISP does give me a n

        • 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 ha

    • 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 (Score:3, Informative)

      by valmont (3573)
      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.
      • 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.
      • 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.
    • 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
    • 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
    • SIP works fine behind most NAT/firewalls, what are you talking about? And IPV6 isn't NAT replacement, no thanks, I don't want to open my private localnet to the world.
  • by chris09876 (643289) on Friday January 21, 2005 @11:19AM (#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.
    • > Open source and open protocols are an excellent way

      Actually it's not important that those technologies are open source as long as they use open protocols (because you can use any kind of technology - both closed and open - to communicate using open standards). If the closed source technology is better (cheaper, etc.) it will win and there's nothing wrong with that.
      This is one of those things where Sun's Schwartz is absolutely right.

      > If the government steps in and starts regulating everything like
  • 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 :)
    • 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.
    • 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
    • I'm a developer on SIP-based PBX software and on proprietary PBX software also (both from a vendor you'd recognize).

      SIP has opened up some options, but the implementation has revealed lots of little gotchas along the way.

      And I was just (in the middle of this email) speaking to a SIP architect from another vendor. His comment:

      "SIP is great in the network, but not so good to the desktop".

      My own observation was that SIP puts a lot of brains often into the end device, and the effect of that is to distribute
  • Spam (Score:1, Interesting)

    by awhelan (781773)
    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 (Score:3, Interesting)

      by luvirini (753157)
      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

  • 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 (Score:3, Informative)

      by stratjakt (596332)
      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.
      • 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!
        • 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.
      • 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,
  • Because everyone is sitting in front of their computer with their IM client of choice.
  • Hooray! (Score:4, Interesting)

    by drewzhrodague (606182) <drew@zhDALIrodague.net minus painter> on Friday January 21, 2005 @11:30AM (#11432026) Homepage Journal
    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...
    • May I recommend you look at http://connect.voicepulse.com if you will be playing with asterisk.
    • Hey I've just moved cites and I decided that instead of getting a regular line, I would try Vonage. Is this the wrong place to go for VoIP? SIP? Asterisk? If so what hardware is suggested. I'm planning to make this my primary phone so I want something that is rock solid (as long as my internet connection is up). Also, I don't want a computer to be requried. I simply want to plug a regular phone into a router (which I want to purchase, not build) and go! I basically want this to work as a regular phon
  • by Anonymous Coward

    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!
  • 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
  • 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.
  • Magic Beans (Score:4, Interesting)

    by Bookwyrm (3535) on Friday January 21, 2005 @11:44AM (#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.
    • 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 s
      • 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, 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
    • SIP isn't a magic protocol, there is only one magic protocol, and that is XML. :)

      SIP over XML might be a magic bullet. ;)
      • 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 (Score:2, Interesting)

    by ebbyfish (759832)
    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 pe
    • 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.
      • 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.
        • 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

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

      by Big_Al_B (743369)
      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.
    • 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 (Score:1, Informative)

        by Anonymous Coward
        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 91
    • 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 (Score:3, Insightful)

      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". On
    • Re:No 9-1-1 (Score:1, Redundant)

      by spacefrog (313816)
      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 (Score:3, Informative)

        by Scyber (539694)
        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 marke

    • No accountability (Score:3, Interesting)

      by Omega (1602)
      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 t
      • 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.
      • 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 phone service goes out, how can you contact the Public Utility Commission in a manner which would bring about results "right away"? In all honesty though, don't you imagine that phone companies had large periods of outage in the early years? Like all advancement, the downtime of VoIP will become insignificant as infrastructure develops.
      • while many would argue to the contrary, i'm not exactly seeing VoIP completely replace POTS in any foreseeable future. I'm thinking it will be a compelling alternative for people to communicate over longer distances, at cheaper costs, in location-agnostic ways. For example, while many people no-longer have a home phone line because they're satisfied with the coverage they get from their mobile phones, most people do still retain their POTS phone line. But I for one, don't spend money on long distance calls
  • 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]
  • 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 messagin
    • 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.

    • 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 tho
  • by akajerry (702712) <`moc.hcetaka' `ta' `yrrejaka'> on Friday January 21, 2005 @12:27PM (#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.
    • 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.
  • 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 comput

  • 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.
    • No, SIP provides call signaling.
    • 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.
  • by Jailbrekr (73837) <jailbrekr@digitaladdiction.net> on Friday January 21, 2005 @01:34PM (#11433412) Homepage
    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.

    • 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

    • 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.

    • 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 i

    • It is ironic that I was going to mention the UPS in my original post, but figured that it would have made it too wordy.

      The main point I was making was the VoIP is a nice alternative for the technically minded, but it is not in any way shape or form a replacement for traditional telephony nor is it a product ready for the ignorant masses. There are too many things which must be implemented on the users side in order to make it almost as stable as a traditional phone line. It is nowhere near 'idiot proof'.
      • Man, did you get hammered with the UPS replies. :^)

        What they, and you, also missed is that the average cordless phone also rolls over dead with no wallpower. Consumers are used to this. My parents, in their 60s, don't expect their phones to work when the power is out anymore.

        The days of drawing *all* your current from the RBOC/LEC FXS jack are disappearing quickly into the sunset, so power is a nonstarter in the IP/POTS debate.

        And, for SOHO-grade IPT, network stability is not nearly the factor you made
    • 1. UPS. 'nuff said.
      2) Use a backup device -- say, a cellphone -- if you need high-availability. Also, look at the disaster-resistant ability of VoIP phones -- after September 11th, the phone networks in New York were utterly swamped with calls, including calls to emergency services, "routine" calls, and calls to/from concerned family members. Internet traffic, with the exception of the destroyed nodes in the WTC and surrounding buildings was largely unaffacted. VoIP phones would be able to work normally whi
  • by Anonymous Coward
    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.
  • 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/w [microsoft.com]
    • It is shame that the ISA server development team didn't get the "We now like SIP" memo, because there is no Microsoft way [microsoft.com] of passing audio or video SIP calls through their own firewall server (VPN or opening 1000's of ports doesn't count).

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