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Michael Robertson Unveils SIPphone

JimCricket writes "After almost a year of preparation, the person behind MP3.com and Lindows has unveiled his latest venture: SIPphone. According to a CNET article, the new company sells VoIP-based telephones. I wonder what kind of latency you get with these devices." Interestingly, the CNET article reveals the telephones "...can only call other phones that use the same technology."

4 of 236 comments (clear)

  1. Just another nexus by Jdodge99 · · Score: 4, Interesting

    FWD is one way for SIP users to interconnect, this could well be another. Grandstream's are cheapie ip phones, but from what I read on asterisk's mailing list - they do work pretty well for the simple stuff. VOIP /will/ happen -- the protocols need to catch up -- with QOS priority usage, nat traversal issues etc. IPV6 would make these things easy, but even without it people will find a way to make things work. Keep in mind -- people may well buy a pair of these as a "free talk" solution on a temporary basis, but then move on to more sophisticated usage. Grab one of these, set up asterisk http://asterisk.org w/ a one port fxo (connects computer to a phone line for incoming/outgoing calls) card - and you instantly gain a lot of flexibility. This WILL happen -- it's only a matter of WHEN it will happen.

  2. Re:If it can only call similar phones... by Anonymous Coward · · Score: 3, Interesting
    You're right -- if it can only call similar phones, it's doomed. That's exactly the kind of truism comment I'd expect from here -- have you ever actually used VoIP?

    Here's why this might be reasonably successful:

    1. What they are selling is a Directory Service + Grandstream phones which support the SIP protocol -- which is *the* standard for VoIP signalling -- oh, which is also supported by the Cisco ATA 186, Cisco 7960, MS IM, X-Lite, Asterisk, etc. -- i.e. basically anyone playing in the VoIP space who doesn't have a legacy H323 or proprietary protocol already deployed.
    2. They've already got an interconnect agreement with FWD which has circa 40,000 users signed up. (albeit not fee paying)
    3. The phones aren't locked to being used for this particular service -- nothing to stop you taking the phones and pointing them at FWD/your own Gatekeeper etc. (Refer: Michael Roberton's comments)

    They've also had the smarts to set their SIP phone numbers as a "US area code" (don't know if they've actually been allocated it, who knows) -- no doubt PSTN access is in their plans at some stage like Vonage.

    Doomed? I doubt it. While nothing here is revolutionary, the genius is in offering the total package (phone + directory service) for a one-off fee that even your grandmother could figure out how to use. All they need to add is PSTN access. If you'd like to learn some more about VoIP, I'd suggest FWD is a nice easy learning curve.

  3. Existing... by daaku · · Score: 3, Interesting

    Making international call's is pretty expensive, especially in developing contries. In India, we get a device which goes between the phone cable and the phone. You press # twice (on your regular phone), it calls your ISP, goes online, checks your account balance and rings. All in about 2 mins. And then you can use your existing phone and call from anywhere to the USA for about 1.9 cents a minute. Hell, even within the US 1.9 cents a minute is pretty good.

    Oh yeah, they've got a ethernet version too. No need to wait for the 2 odd minutes. And the connection is crystal clear. The callee never realizes how 'cheap' the caller is!

  4. SIP isn't new at all - neither are SIP phones by potcrackpot · · Score: 3, Interesting

    I thought slashdot was supposed to be full of people who knew what they were on about? More than half of this thread is rubbish! "Doomed, we're all Doomed!"

    Saying that SIP is dead is like saying that, ooh, UDP will never take off.

    It's been around for ages, and it's not just used for phones; it's a generic session establishing protocol.

    Essentially, you want to set up some kind of media session between two endpoints; what you do is you send a SIP INVITE message through proxies etc. and attach another kind of protocol message (such as SDP) which describes the requirements for the session. The endpoint receives this, and establishes the session directly (without the proxies etc. in the loop). In very short.

    It's just another protocol, like DAP.

    SIP phones have been around for ages too - Pingtel's offering is probably the best one.

    That said, I don't see SIP phones as taking off in the home, or for personal use; they're much more suited to being used in small enterprises etc.; much lower cost than a PBX. You'd have to have some kind of PSTN interface with the outside world - perhaps phone companies will start providing softswitch capabilities so that people can make their VoIP network speak to the outside world?

    What you CAN do at the moment is have a mixed network - VoIP which talks via a router (Cisco 2600 for example) which then talks to the PBX which talks to the PSTN in the usual fashion.