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."

If it can only call similar phones...

...then it's doomed already.

What kind of service is that?

"...can only call other phones that use the same technology."

I nkow some cell phone companies have offers when calling within their network (no use of minutes, extra minutes, etc.), but not being able to call out of the network at all?

As someone said, the thing is dead already.

The only thing I can think of that it might work well for is buisnesses. Think of something like a Nextel walkie-talkie cellular service, but without the 'fear' of employees calling other people.

Other than possibly that, however, this thing will never sell.

Thursdae
600 minutes/month, free nights/weekends, and free long distance on my cel plan... and I can call anyone I want.

It's SIP service, silly

As the name implies and the article explains, the phone uses SIP, or Session Initiation Protocol [columbia.edu]. I did some research on SIP last year and found it to be somewhat intruiging.

SIP is basically used for setting up the endpoints of a human communication channel over an IP-based network. It negotiates what kinds of communcations are supported on each end, and what protocols to use. So if a video-SIP-phone calls a regular analog phone via a SIP-PSTN proxy, the proxy would only support audio certain codecs. The calling video-SIP-phone and the proxy would negotiate to use only audio using a matching protocol and the cal would go through.

And since SIP is a protocol just like SMTP or HTTP, it is very controllable. There are dozens of SIP products popping up from SIP servers to SIP proxies... and now SIP phones. For example, you can have a SIP proxy/server be concious of where a user is logged in and re-route SIP calls to their present location. As a Java programmer, I'm looking forward to the day when I find a reason to write a SIP Servlet.

Furthermore, the latest version of Messenger in Windows XP supports SIP. I would think that this means a SIPPhone could call someone using Microsoft's Messenger on Windows XP. However, I was not able to confirm this with a breif perusal of the SIPPhone site, and they also state this only works with other SIPPhones. That may be an over-generalization to keep people from thinking it works with regular phones, or maybe they did something crazy with it.

I'm crossing my fingers that it is a generic SIP endpoint that can contact any SIP-enabled device.

Re:It's SIP service, silly

I would think that this means a SIPPhone could call someone using Microsoft's Messenger on Windows XP. However, I was not able to confirm this with a breif perusal of the SIPPhone site, and they also state this only works with other SIPPhones.

Check out the product spec [grandstream.com] from the manufacturer.
The SIPPhone page states the make and model.

Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products (e.g., MS Messenger, Cisco IP phone and gateway, etc)

Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, G.728, and wide-band G.722 (Model 102D). Dynamic negotiation of codec and voice payload length

G.711 is the granddaddy of the voice codecs. It doesn't say it uses H.323, but I'm guessing it does, seeing as it interoperates with cisco and msn messenger voip.

You can probably even use a different directory service than SIPPhone.com's ; the phone has a web interface for configuring it.

American area code for an international system?

After reading the how it works page [sipphone.com], it looks like all these phones will use the US/Canadian area code 747. While this jumbo number is easy to remember, I'm asking myself if it would have been wiser to use a new country code instead. Imagine asian people exchanging their phone number, and one of the two has a number starting with +1747... it just doesn't sound right...

Must we support Michael Robertson??

Lindows is good advertisement for Linux, but for all wrong reasons - root login, anti-virus etc.. Secondly, Lindows has subsidised the SCO vultures. Despite all the hype about XBox cracking, no one but Microsoft has benefitted from the hack.

It's tough supporting someone who paid SCO.

-

Internetwork gateways?

Even if you could only call people using phones with the same technology locally, surely a solution could be devised to call other networks, as we do with areacodes and whatnot already?

The question rather becomens, does it make sense to do so?

I like it

I already read about that a few hours ago on a German newspage and am currently waiting for SIPPhone's sales dep. to answer my questions :)

As you get 2 phones for $129 its not that big problem that you can only call other SIPPhones with that... I might buy a pair and give one phone to my girlfriend as she lives over 200km away from me and a priceless phone connection to her would lower my bill alot ;)

If anyone is curious about the quality, there is some info about that on their homepage [sipphone.com] saying:

SIP calls typically have very high audio quality. Call quality is much better than cell phones and may even be better than land line phones you're used to - especially over long distances and between countries. SIP uses the latest compression techniques which allow calls to sound their best.

Sounds ok for me :)

Re:I like it

Your standard analog land line uses a stream of 64kbps (ulaw/alaw compression) but this is related to the quality of the audio not the latency of the network it goes over. The telecom network traditionally allocates a complete end-to-end circuit for your call - wasteful but guarantees QoS.

I definitely think this [VoIP] is the way to go. This is an preview of the future and as such will have limitations.

Just another nexus

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.

A step in the right direction

This business model seems very much like the way we as consumers should be heading.

I am reminded of the failed business plan when faw machines were first commercial (before they were common) FedEx offered a service called ZapMail, whereby they offered 2 hour delivery of documents rather than 1 day. They did this by faxing the documents around FedEx offices.
Of course people realised that for a small initial investment (buy a fax machine) they could do they same thing themselves, cheaper.

This seems a small venture at the moment and may be ulitmately unsuccessful due to the limitiation of only being able to call other SIPphones, but it is a step in the right direction and may pave the way for other businesses to operate using a similar model.

I see uses for not only businesses but for travellers and ex-patriots. It is increasingly easy and cheap too access broadband internet while costs of international phone calls are still high.

Can only call other VoIP phones that use SIP

From the FAQ:

Q: Can I use software or what is called a softphone to make and receive calls with SIPphone?
A: Although it may work, at this time we cannot offer support for anything but a certified SIP phone.

Q: Are there other SIP phones I can order besides those offered at SIPphone?
A: The SIP phones offered at SIPphone are designed to work out of the box with SIPphone with zero or minimal configuration. We also work to offer the most affordable SIP phones available in the world. Many SIP phones cost hundreds of dollars. SIPphone sells 2 phones for just $129.99. It may be possible to use the SIPphone directory with other phones, but no technical support is available at this time to support this.

Q: I already own a SIP phone and I would like to use your SIPphone directory service. What should I do?
A: First, you need to sign up with our service at SIPphone Sign Up. These are the settings that you will want to use:

SIP Server: proxy01.sipphone.com (130.94.123.252)
STUN Server: stun01.sipphone.com (69.0.208.27)
NTP Server: ntp01.sipphone.com
TFTP Server: tftp01.sipphone.com (130.94.123.253)

Currently the SIPphone directory service has only been tested with the Grandstream BudgeTone 100 phone. Please check back for further updates on "SIPphone friendly" SIP phones.

SIP Phones

SIP is an IETF standard for voip, surely he just meant it could only phone other SIP phones! no need for any conspiracy theories! SIP is an open standard, and you can even get linux software linphone [linphone.org] to use it... Just need a gateway to the traditional phone system and yer sorted.

latency?

"I wonder what kind of latency you get with these.."

I would assume the same latency you would have with any application that would have taken the same network path as the 'net phone's packets?

Perhaps you are talking about an audio delay? In that case, assuming your ISP has proper routing, there should be no significant delay (around the same as many cell phones) when speaking to someone else in your same country.

I've set up vbrick devices to use two T1's bridged for LAAtlanta conferences and the delay was barely noticable.

Not since dialup on a 28.8k modem have I noticed much problem with audio communications on the web. Definitely better than the telco's international service back in the 80's. I remember talking to friends in Germany and Japan and having to stop for long periods of time between sentences to prevent cross talk.

I think this product is so-so, though. Without a subscription based access from the voip phone to a telco bridge and a real phone number, it's not going to explode in popularity regardless of it's audio quality.

SIP is the way to go

SIP is not limited to just VoIP, as the name says it is Session Initiation Protocol. There already is a reasonable GNU SIP library, so let's make that better, and then we can create an open source SIP capable VoIP-phone that could interoperate with this system as well as others.

Other uses for SIP that could/should happen IMO are (starting a session of) multi-player games and messaging, conferencing software for sharing pictures, etc.

Since SIP is basically just a handshake protocol, doing all those things shouldn't be impossible. Wanna play a game of chess or go with a pal? Just initiate a SIP connection, if their end supports your game and they are available, you've got a connection. No more application specific ports to configure to get a multiuser application work.

Existing...

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!

This is Doomed.

Creative tried it. with he VoIP blaster. While being fantastic devices and I snapped up as many as I could to use not with creative's services but with fobbit so I could use them without relying on their servers for routing and connecting.

Most people were not interesed in it because it was semi-difficult to use and made you think you needed to buy their service, which you didn't.

I use VoIP all the time. My GF lives 100KM away so that makes it cheaper IF both ends have DSL or CABLE modem. VoIP completely sucks over a dial up.... which over 60% of internet users still have as their only way of getting online.

I wish him luck, but there is cheaper and better hardware out there already (The VoIP blaster is still sold under the origional manufacturer's brand) that is cheaper and much more flexible.

Cisco has them

Our company have several Cisco VoIP phones deployed in various departments. We even have the ability for them to interface with the PTSN through special hardware attached to a 5ESS switch.

The only thing that prevents us from doing any massive rollouts is the utter fact that price per user and the nature of data networks make the phones more subject to unusability due to network problems than a normal phone.. This is not latency issues were more worried about something like a OSPF/Firewall or something along those lines wiping out a whole department's ability to communicate.

But who has the directory?

You can pick up SIP phones, and even nice H323 videoconferencing hardware cheap these days from Taiwanese OEMs. Companies like vonage.com [vonage.com] or pilmo.nl [pilmo.nl] will even hook them up to the plain old telephone system for you.

The main problem is that each company that sells these things to end users uses it's own LDAP directory. So you can call other people who use the same brand easily by tapping a 'phone number' that's the same regardless of their everchanging IP number, but don't expect to call your buddy who's using netmeeting so easily. Also, if you place a call from one VOIP telco to another, chances are it will travel some distance over PSTN and will be billed in stead of free, despite the fact it could have been an end-to-end-over-IP connection which is usually free of charge.

Of course SIP can work over the real dns just beautifully (using SRV records), but do these phones support entering alphanumeric user/hostnames? And will hotmail support SIP? (Answer, yes it will, and it will tie in with MSN video/voiceconferencing and Microsoft SIP phones...)

Not a problem

"...can only call other phones that use the same technology."

The article seem to have forgotten to mention that (almost) all 3G mobile phones have native SIP support. It means that in near future all mobile phones, atleast in Europe can call via SIP.

Since Microsoft Netmeeting has SIP support, and Linux has its own SIP stacks, you might be expecting a SIP boom soon.

SIP is probably the future of IP calling. It has some very nice features in it that make it work well with other messaging applications like "InstantMessaging". I'd say put your money on SIP now.

SIP isn't new at all - neither are SIP phones

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.