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VoIP Regulation, SIP Insurrection

Posted by michael on Fri Jan 21, 2005 11:11 AM
from the insurgents-vs-incumbents dept.
Communications The Internet
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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  • 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) <wolf31o2@gentoo.org> on Friday January 21 2005, @12:28PM (#11432713) Homepage
      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

    • 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.
    • 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

  • It should be great... (Score:5, Insightful)

    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.
  • 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

  • Hooray! (Score:4, Interesting)

    by drewzhrodague (606182) <.ten.eugadorhz. .ta. .werd.> 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...
  • 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

  • Magic Beans (Score:4, Interesting)

    by Bookwyrm (3535) on Friday January 21 2005, @11:44AM (#11432196) Homepage
    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. ;)

  • 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

    • 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:3, Insightful)

      by Gaewyn L Knight (16566)
      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

    • 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.

      • 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

  • 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

  • SIP needs to opened up (Score:5, Insightful)

    by akajerry (702712) <akajerry@akate[ ]com ['ch.' in gap]> 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.
  • 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'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 (Score:3, Interesting)

          by Afrosheen (42464)
          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.

    • 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.
      • 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,
        • 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: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

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