Project Gizmo Challenges Skype

valmont writes "The Register is offering an interesting introduction to Project Gizmo, a new player in the Voice over IP field, poised to challenge Skype with its ability to interoperate with others thanks to the SIP protocol it complies to. Whereas Skype has selectively licensed usage of an API that offers limited insight into a closed protocol, a closed ecosystem solely controlled by one organization, the SIP protocol is open. Free open-source proxy/server implementations are sprouting up, and many developers are actively working on SIP clients. The Gizmo Project is the first to bring a truly-usable, user-friendly, cross-platform SIP client (Mac, Windows, Linux coming soon) to market. Meanwhile, theappleblog.com is already offering a Gizmo Project Wish-List to promote better interoperability between current and upcoming SIP providers, to make it more practical for users of disparate SIP clients to communicate with one another."

Comments

Gizmo Interoperability

[valmont]

I've also posted a couple of tips and tricks on getting Gizmo working with other SIP systems, and also listing some other players in the SIP field.

Re:Gizmo Interoperability

[valmont]

you might consider reading-up on the STUN protocol, which came out in 2003, and solves the VAST MAJORITY of NAT traversal issues, albeit, not *all* possible issues. But this is where your SIP provider comes in, and offer to relay RTP (voice data essentially) traffic for you, which is a similar practice as Skype using people as "supernodes" without their knowledge or conscious consent. Except that in the SIP model, it's the SIP provider who spends the extra network resources, not some hapless random person Skype may have picked for you.

Re:Gizmo Interoperability

[bigzigga]

Actually, SIP is only a signaling protocol. It has nothing to do with voice quality. Voice quality is determined by the compression codec. Most SIP systems in North America use G.711 uLaw for maximum interoperability with public telephone systems.

Also see Ineen and P2P SIP

[Wesley Felter]

Ineen is similar to Gizmo but it also has IM and video. It seems like companies are falling over themselves to give away SIP clients these days.

Ineen and Gizmo are still client-server SIP systems, but to truly compete with Skype, the IETF is working on P2P extensions to SIP.

Conflict of "Gizmo" name?

[Jim_Maryland]

SunRocket, a VOIP company, has a hardware device called "The Gizmo" (see SunRocket How to Install and you'll see their adapter named "The Gizmo"). I can't find anything to indicate which came first though.

Re:Conflict of "Gizmo" name?

[Ulven]

According to The Register they are looking for a new name anyway.

"Indeed, Gizmo is even asking for suggestions on a "cooler" name under which to operate."

Re:Explanation

[ZephyrXero]

Didn't you ever hear the rule that once you have to explain a joke, it's no longer funny? ..heh...

Re:Stanaphone is SIP too

[Rosco P. Coltrane]

Stanaphone.com is the FIRST usable multi-platform SIP application. They offer more free things than Skype does (it's a real telephone number for example, for free). Quality is not that great though, I prefer Gizmo's.

Skype works well enough for me, the sound quality is more than okay for telephony, and the latency really isn't bad at all considering. And the Linux client is stable, and is a snap to install. However, I don't use it because:

1 - There's outgoing traffic from my box all the time. I know it's the P2P routing, but it's a bit unsettling to watch it chew away that much bandwidth

2 - It's made by the Kazaa people, and therefore who know what the hell it does behind your back (spyware?). I straced it, and didn't see anything nasty a-priori, but I don't trust Kazaa folks in that respect, period.

3 - The most disquieting feature: it goes some UDP magic to punch its way out of the firewall without telling you anything of what it does. That's sneaky. I know they claim it's for user friendliness and easy of installation and yada, but I say the networking code is too damn sneaky for my strict Unix-bred style of making sure software I install behaves predictably.

So I can't *wait* for an open-source (or at least open-standard, as Gizmo pegs itself) Skype replacement. The biggest problem of course, now, will be to have a compelling enough alternative that the millions of Skype users are willing to switch over, which isn't too likely.

some lint

[The Creator]

It was'nt a gremlin, it was i belive, a "mogvai"(no i don't know how to spell it). It did'nt turn into a gremlin until you fed it after midnight.

Re:SIP and NAT

[valmont]

WRONG, NAT was SIP's weakness ONLY until the STUN protocol came out in 2003. Since then, all SIP clients use the STUN protocol to traverse NAT at least as well as Skype does. I've used EarthLink SIP, sipphone.com SIP, FWD through multiple layers of NATs without a glitch.

STUN is a major enabler of SIP

Not FREE

[jtosburn]

Free beer, yes.

Free speech, no. There's a EULA (http://www.gizmoproject.com/gizmo-end-user.html) which grants:

License

This Agreement grants you a personal, non-exclusive, non-transferable, non-sublicensable right to use one copy of the Software for your own personal use on a single computer and/or workstation. SIPphone reserves all rights in the Software not expressly granted herein, including without limitation ownership and proprietary rights.

Among other items is this:

EMERGENCY/911 CALLS

YOU EXPRESSLY ACKNOWLEDGE THAT EMERGENCY 9-1-1 CALLS ARE NOT INTENDED TO BE CARRIED/SUPPORTED BY THE SOFTWARE OR SERVICES AND THAT NEITHER SIPphone NOR ANY OF ITS AFFILIATES, SUBSIDIARIES, PARENT COMPANIES, AGENTS, NETWORK SERVICE PROVIDERS, PARTNERS, OR EMPLOYEES ARE OR WILL BE LIABLE FOR SUCH CALLS.

Re:Not FREE

[SirPrize]

I don't see anything wrong with either of those statements. They let you use the software, you can't sell it, and they retain rights to it. Nothing wrong there. And about E911 calls, all VoIP providers have to provide such a declaration, to make users aware that the 911 service via VoIP does not provide location information, etc., to the 911 call center, which standard calls via PSTN would. I think Vonage is one of the few services that actually DO provide a 911 service.

Re:SIP SCHMIP

[valmont]

SIP by its very definition defines and enables P2P based on open standards. The SIP protocol is a signaling protocol, which enables two machines to find each-other on the internet and start exchanging data in real-time without going through a 3rd-party.

SIP opens the doors to far more than just real-time communications (text, voice, video), it also opens the doors to file sharing. It's a matter of someone writing a client that does file sharing over SIP. OH WAIT, someone already did as a proof of concept, and released the source code under a BSD-style license.

Voice quality is absolutely fantastic with Gizmo, but it entirely depends on which SIP client you use, and which SIP client the party you're calling uses. The SIP protocol, is only a signaling protocol. There are a buttload of other open protocols such as SDP (Session Description Protocol) that come into play and allow for infinite layers of interoperability based on users' computing and network resources. Through handshaking, two SIP clients can easily agree to the best codec to use.

Re:We use skype and

[chrome]

Well, I just installed it on my PC and my Mac.

PC version: Worked initially. Started up, created an account, and I edited my profile. Saving my profile however made it lock up and I had to kill it. Starting it again just makes it spin at the Login window.

Mac version: Initially brought up login window, but now dies silently while trying to start. No error log to system.log.

Oops, I guess getting exposure on /. wasn't such a good idea just yet. They will need to get the stability working otherwise a lot of people like myself will try it and be turned off by its quality before it's really ready.

Mandatory FreeWorldDialup comparison

[MCRocker]

FreeWorldDialup has been doing pretty much the same thing for years now. They even have their own pre-configured software. If you don't like the Pulver Communicator you can always get one of half a dozen other SIP phones (soft or otherwise) that work with the service. They even offer a pre-configured version of The SJPhone, which is essentially what the Gizmo folks are offering. If you want a dial-out service FWD offers a choice of affiliates who can give you that portion a-la-carte!

So, why would you want to use Gizmo?

Re:Mandatory FreeWorldDialup comparison

[Colol]

So, why would you want to use Gizmo?

Having played around with Gizmo for the past few hours, the answer seems to be "the interface."

For lack of a better way to describe it, it's Skype, but with all the interoperable goodness using SIP offers. It's a little easier to work with than the majority of SIP softphones out there. You get the ease-of-use of Skype with the flexibility of being able to dial any SIP URI. I've already used it to call into my Asterisk box, and it's all hunky dory.

For the telephony geeks (myself included), Gizmo's probably not a big deal. There's nothing I gain from using Gizmo that I can't do with my Asterisk server and a softphone (except the ability to call Gizmo users, which I'm sure will be possible in the near future).

On the other hand, Gizmo would be just the thing for people like my parents. Big friendly buttons, an easy-to-use interface and phonebook, and none of the bizarre things X-Lite likes to do.

Re:Stanaphone is SIP too

[Chandon Seldon]

I'm not a big fan of Skype, but the "UDP Magic" you speak of is a normal technique used for bypassing firewalls in modern UDP applications. On the protocal level, it's no more of a dirty trick than network address translation is to begin with, and on the user level it's no more relevent than TCP vs. UDP. There's no security issue here.

Re:SIP and NAT

[Anonymous Coward]

Unless you happen to use full cone NAT (Basically NAT with connection tracking used for security)in which case STUN doen't work. TURN (IIRC) can do this, but at the expense of tunneling all of the peer to peer part of the call (i.e., RTP and its ilk) though far fewer paths than would be nice (basically, from TURN server to TURN server), which can suck if the TURN server endpoints are in the US and South Africa, but you are trying to call Hong Kong.

Don't get me wrong, Skype sucks due to it's proprietary nature and it's creators (does anybody really trust these guys?), but STUN is only part of the solution, and most SIP clients do not yet support TURN.

Re:Not as Free as You Mighty Think

[Anonymous Coward]

Gizmo Project is not a pay service. It's free. You just have to buy CallOut credits to call the PSTN (like everywhere else). Gizmo to Gizmo calls are free and Gizmo to networks that Gizmo is peered with are also free (UCSD, UCI, UCSC, SIPgate, etc).

Re:SIP and NAT

[JanneM]

You seem to be confusing Skype with Kazaa. The founders of Skype founded Kazaa as well, but sold it and have long since nothing to do with Kazaa the company.

Re:Why I won't switch from Skype!

[Anonymous Coward]

This is the EXACT same policy as Skype. It is just an issue of accounting. At some point, the company has to be able to know when an account is no longer in use, which allows them to right off the liablilty (potential to make a call) and recognize the revenue. You are such a fool. You won't switch from Skype because they have the exact same policy?

Re:SIP and NAT

[valmont]

It depends on who your SIP provider is. When all else fails, your SIP providers might relay RTP traffic between you and your party, which is similar to practices employed by Skype whereby they'll get around network limitations by routing calls through various members, without members' knowledge. Which I find insanely unethical. Do give Gizmo a try and see for yourself instances where you're unable to place or receive calls where Skype can. I have personally not run into such instance, but it'd be interesting to see people document those instances.

I've used various SIP clients (XLite/SJPhone/Gizmo) through 2 levels of NAT in my own home (192.0.0.* network plugged to 10.0.0.* network hooked to my DSL WAN), calling a free world dial-up user over his SIP URL, who sits behind asterisk at home, and one big NAT. It works just fine for me. I've SIPed countless users who sit behind your average linksys router, or behind your average 2wire home-network-in-a-box kinda thingie. Upon inspecting tcpdumps and SIP client logs, In no case did i ever see the various SIP providers actually routing the RTP traffic for me.

So again, SIP may not work everywhere, but damn, I've tried some really ODD shit with SIP working seamlessly.

More about Gizmo

[Anonymous Coward]

Ok, we admit it's a lame name, but we think the product is solid. Here's why it's significant:

- It is based on SIP so other SIP hardware and software should interoperate. Check out http://www.siphardware.com/ for some of the hardware devices. I have a Call-in-One so a SIP call rings my home phone. Even a wifi SIP phone should work with Gizmo.

- It navigates firewalls and NATS. Skype has done this well, but SIP clients have generally not. Gizmo Project should penetrate just about any firewall or NAT setup. I used it successfully on Boeing's wifi service on flight to Frankfurt 2 weeks ago bouncing calls off satellites.

- It's Mac/Win today and YES, there will be Linux client shortly. No, really.

- We're committed to an open directory meaning that we will interconnect with everyone. We already connect to many big and small networks. See: http://sipphone.com/numbers/. We do enum lookups. We even support the intriguing DUNDI extension for Astericks. I want VOIP world to be like email where one address is reachable by everyone. I hope VOIP does NOT become like IM where AOL can't talk to MSN who can't talk to Y, etc. If you care to read more about why I think this is important, you can go to: http://michaelrobertson.com/.

- All calls on IP network are free as is voicemail, call recording, conference calling, etc. (try calling 1-222-xxx-xxxx and create an instant conference call with unlimited number of callers.) Only have to pay if touching the PSTN. We do give you 25 cents for free to check out the PSTN quality.

- Be nice because it's just beta software! Thx.

-- MR

Bluetooth headset connectivity...

[Critical_]

One of the greatest things about Skype is its API. Someone brilliantly wrote a plug-in for Skype that ties together my bluetooth headset with Skypes calling functions. This way, my headset works like a normal headset. Get it here: http://www.skypeheadset.co.uk/ Case in point:

1. The headset is automatically detected and Skype's sound devices dialog is changed automatically to use the bluetooth headset.

2. The headset sits in standby mode. Before this plug-in, the headset would have to be tied to the machine in its active mode which drained the battery within 4 hours even though there weren't any calls. This plug-in only activates the headset if there is an incoming call or you hit the quickdial button.

3. The quickdial button can be configured to any # you have in your phonebook.

Now show me any SIP client out there integrates bluetooth (even via third party plug-in) as well as Skype. If you can find one, then I'll ditch Skype in a heartbeat.

Don't let anyone snooker you out of your freedom.

[jbn-o]

The license for Gizmo Project software is quite clear, the software doesn't even qualify for the weaker standard of "open source" (weaker than "free software" which requires that the user have permission to make private derivatives):

License Restrictions

You may not reproduce or distribute the Software for any purpose whatsoever. Without limiting the foregoing, you may not copy the Software to any server or location for reproduction or distribution. You may not reverse engineer, decompile or disassemble the Software. The restrictions contained herein apply equally to any updates that may be provided to you by SIPphone.

The company offers what they call "Opensource support", which appears to be little more than name-dropping. But I expect people around here to know better than to take on faith whatever a proprietor says.

At the very least, even if there is no intention to bamboozle users into adopting the software, this is yet another use of the term "Opensource" which shows how people don't understand what that term means. Reminiscent of examples of same from the FSF essay on the differences between the free software and open source movements (examples include Neal Stephenson's writing, the State of Kansas' definition, and a particularly interesting example from a trade show in late 1998).

As a result of a little digging into Project Gizmo, I still see no real competition for GNOMEMeeting. I see one more proprietor asking me to install a mysterious program on my computer so that the proprietor can do something on my computer that I am prohibited from inspecting, sharing, or modifying.

Re:Why I won't switch from Skype!

[blixblix]

Form the Skype Announcement:

Luxembourg, November 26, 2004:
When Skype first launched our SkypeOut service, we didnt realise exactly how long 10 Euro would last. At our launch, we announced a policy that unused credits would expire after 180 days (6 months). Well, we're soon reaching 180 days from our beta launch, and can see that many of you still haven't used all of your original credit. As a result, we are pleased to tell you that Skype has revised our policy. We won't be clearing any balances until 6 months after you have made your last SkypeOut call. So, if you make just one paid call every 6 months, we'll not touch your account of unused credit.
However, if you don't use SkypeOut (that is, don't make any additional purchases nor place any SkypeOut calls in 6 months), we'll zero your Skype balance, simply to ensure that we aren't using expensive systems to maintain accounts for people who don't need it. Your Skype Name will be unaffected and you will be able to still use free Skype-to-Skype calling, file transfer, and all of the other free services you've come to enjoy.
We believe this policy is both market leading and the right way to keep our costs down, so we can pass savings on to you.

Re:Er, other way around.

[Anonymous Coward]

http://info.lindows.com/inthenews/nytimes12302002. html

"An upstart company, Lindows.com, is trying to persuade the Federal District Court in Seattle to invalidate Microsoft's trademark on Windows."

has noone here actually used this yet?

[intro]

I often use skype, but in the quest for a SIP# along with my skype experience, I tried this yesterday.

I is a fairly hopeless alternative to skype, at least for the time being. Gizmo gives you "free" minutes of which you can use to test the system, but, try as I might, I was unable to make a call outside of the US, as 001, the country code for the US was prefixed onto any number I entered in.

Add the random shutdowns when editing certain options (of which there are a distinct lack of) and odd, finicky menus which stay open and dont have a close button, and you have yourself Gizmo.

Having said that, if they can sort those problems out then they will be onto a winner. It is certainly far easier setup than any other SIP/softphone system Ive used.

Re:SIP security

[elgaard]

> Why don't all these VOIP companies use some sort of encryption?

Some of them do (eg. musimi.dk which i use). They use S-RTP AES.

Not all clients support it though (Sipura, Zultrys do).

Re:SIP SCHMIP

[level2trader]

P2P - Answer is NO. But neither is Skype -- Skype uses "special" super nodes as a central meeting for regualr nodes. If you ask me -- this is the definition of a CLIENT -- SERVER setup. Encryption: In BETA stage encryption makes troubleshooting a nightmare hence answer is a "NO" for now. But its a matter of setting a boolean flag to "true" for us to enable it. Takes 5 minutes. Comparable Voice quality: Since we do not rely on "super nodes" running on end user computers / net connections overall you will find Gizmo call quality better. have fun with VOIP. regards, PM.