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Saturday, August 27. 2005Google Talk And Skype Are Boring, And Here's Why
It seems that everywhere I turn on the web these days, people are excited about Google Talk and/or Skype. "Real voice conversations over the Internet for free," people say.
True. But, at least in the case of Skype, you are hooked into a single vendor, and at the mercy of that vendor. (I don't know about Google Talk's network specifically.) Neither one has done anything really innovative, either. Neither has any features that SIP doesn't. Really, both Google Talk and Skype are "been there, done that" technologies. And what's more, far superior technologies exist. So in response to all this undeserved hype, I would like to take this opportunity to talk about SIP and why I find it much more exciting for voice conversations. What SIP Is Put very simply, SIP, in its most common incarnation, is a protocol for carrying voice conversations over IP networks. It is used for everything from calling the kids in college from a home PC, to massive telecommunications infrastructure by some of the largest multinational corporations in the world. Wikipedia has a technical description of SIP, and you may note that some other protocols (such as RTP) are involved. For most purposes, though, these types of calls are called just "SIP". SIP is an open industry standard. They are players of all sizes in the SIP marketplace -- from giants like Cisco, to "home VOIP" companies like Vonage, to small non-profits offering free calling services. First, let's take a look at some use cases for SIP. Then, I'll describe all the variety available in the SIP marketplace, and give some examples of why it beats out Skype so easily. SIP for free IP-to-IP calling If you are using Skype or Google Talk on your PC, to talk to someone else over the Internet also using a PC, you are using what I'll call "IP-to-IP" calling. That is, your call never hits the traditional (PSTN -- Public Switched Telephone Network) phone system. It is carried 100% on the Internet. This is trivial with SIP. At its most rudimentary, you can connect to someone else by giving their IP address and a "phone number" to connect to at that IP. In practice, this is rather inconvenient and is rarely used. Rather, most people use a service such as Free World Dialup (FWD) for IP-to-IP calls. It's completely free and works with any SIP phone. You just sign up and get how ever many FWD "phone numbers" you want. You can be as simple as you like, and treat it like Skype, or you can set up voicemail, PBX systems, whatever. A little-known fact about FWD is that they have peering agreements with many commercial VOIP providers, letting you call into and out of the FWD network for free if you're a customer of one of these providers (or know one). SIP for IP-to-PSTN calling Many people are using SkypeOut, which lets you place a call from your PC to a phone on the PSTN (traditional phone network). They charge approximately $0.023/min USD for this service. With Skype, this is pretty much your only choice. If you're using SIP, you have -- and this is no exaggeration -- hundreds or thousands of choices. Personally, I use and recommend Voxee, which charges $0.011/min USD for this service, and that's without any sort of bulk buying arrangements. SkypeOut is more than twice as expensive. There are plenty of other options -- Vonage is one that targets residential customers, and there are others that target large corporations that sell minutes in blocks of 1 million. In SIP lingo, a company that provides IP-to-PSTN calling is known as a "termination provider" -- that is, they "terminate" your call onto the PSTN. SIP for PSTN-to-IP calling As far as I know, Skype does not support this. (Update: there are some sketchy details on SkypeIn; see comments below.) Many of the SIP providers linked to above also provide the reverse: they will, for a small monthly rate (often under $5), sell you a local number. When someone calls that number, the call is routed over the Internet to your VOIP system as if someone had called from another VOIP node on the Internet. Rates are usually quite low on these as well, even for 800 numbers. In SIP lingo, companies that provide this service are said to provide "DID" (Direct Inward Dial) service. In addition to the above, there are some other general purpose PSTN-to-IP gateways. IPKall, for instance, links regular PSTN numbers into the FWD network. SIP Phones With Skype, any device that you are using must be used with a PC that is running the Skype software. That may not seem like a problem to you, but it can be a surprising limitation. Of course, there are plenty of PC-based phones available for SIP. They are called "softphones", and there are dozens available. You can also find "hardphones", which are implemented in hardware. These phones require no PC to operate. They typically have an RJ-45 Ethernet jack only, and all they need is a network connection and the name of a server (or two) to go. A person could have no PC in the house at all and have a working SIP hardphone, no problem. There are dozens of these available as well. I personally have several Sipura SPA-841 phones, which sell for about $80. Very nice. Work right out of the box, but also highly configurable. There are also a few of SIP hardphones that use 802.11b Wifi instead of Ethernet. This market is growing. SIP Analog Telephone Adapters Perhaps you'd rather keep the traditional analog phone that you already have, instead of buying a new phone. No problem, you can buy SIP Analog Telephone Adapters (ATAs). ATAs for personal use sell for under $100. They typically have two ports: an RJ-45 Ethernet port, and an RJ-11 phone port. Plug in the Ethernet and phone, point it at the appropriate server(s), and you're set. They provide dialtone, caller ID, ring voltage, etc. as appropriate to the analog phones. Once again, no need for a PC. In more advanced settings (see below), you can also use a different type of ATA to go the other way: to integrate a line from the PSTN into your VOIP setup. More on that below. Be Your Own Provider You can run professional, industrial-strength PBX systems in your own home with a program such as Asterisk. This gives you all the features you'd expect from a "business" phone system: multiple extensions, voice mail, automatic call routing using least expensive lines, etc. I've written about my Asterisk setup and given some details on Asterisk before, so I won't go into a whole lot of detail here. Suffice it to say that you can really go places with SIP and Asterisk. You can use an ATA to let your outgoing VOIP calls get routed over a PSTN line, or the Internet, depending on costs and whether the appropriate lines are available. You can get bulk concentrators, connecting dozens or hundreds of analog phone lines to a VOIP server at once. There are a bunch of options. Common Questions Here are some questions you may have: What is the sound quality of SIP? There are many different protocols, some compressed, some not, that are frequently used with SIP. In general, over my DSL link, I find SIP calls to sound better than calls I place with a traditional phone. Is it hard to set up? A basic setup, similar to what you would use with Google Talk or Skype, is really very easy. FWD has a nice Quick Start Guide that you might want to check out. Obviously, if you are setting up a PBX with Asterisk on Linux, you will need a bit more expertise. Is SIP here to stay? Unquestionably yes. There are very big names (Linksys, Cisco, Nortel, Earthlink, SBC, etc.) using SIP. There are also community organizations that use it. It will be here for a long time. Where can I find more information? If you have some basic questions, you can post comments here and I will try to help you out. (I will not reply to e-mailed questions.) I highly recommend the VOIP Wiki at Voip-Info.Org. It is a thorough and amazing resource that I have referred to on many occasions. Is it available globally? Yes, all over the world. VoIP User, for instance, is a community-run PSTN termination service in the UK. Public and private organizations exist all over, though. Trackbacks
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I was under the impression that Google were going to be using SIP (albeit there own implementation..)?
I've heard that rumor, but no confirmation anywhere, and also no indication if they've somehow locked it down. It would be nifty if it were a standard SIP service, for sure, though alphanumeric usernames, it would be difficult to place a call to a Google user from a SIP hardphone.
It's not a rumor.
<a href="http://www.google.com/talk/developer.html">Google Talk's Developer info"</a> clearly mentions it: <blockquote><b>5. What protocols are used for voice calls?</b> Google Talk supports a custom XMPP-based signaling protocol and peer-to-peer communication mechanism. We will fully document this protocol. In the near future, we plan to support SIP signaling.</blockquote> Oh and BTW, I'm glad that someone finally mentions that this <i>isn't</i> innovative. VoIP has existed for *years*, even before SIP (although it was ugly). It's very unfortunate that it's becoming popular by the means of proprietary protocols...
This is another reason why the convergence of Video, Audio, IM, Identity and Presence is important.
Skype provides SkypeOut for PSTN to IP calls.
The real advantage of Skype is that I can sit down on a random NATed connection, and call someone behind another random NATed connection, without having to beg whoever runs it to open ports, etc. Nothing else seems to manage this.
Thanks, I wasn't aware of SkypeIn. I checked their page about it, though, and it says it's in beta test, and also wouldn't give me specific rates for my country. I will correct the article.
Anyway, I have never had trouble with SIP and NATs. Most SIP phones/equipment these days should Just Work. I once opened up Asterisk on my home firewall, even carried my SIP phone from my own house to my parents' place, connected it to their ethernet (behind a NAT), and it hooked right up to my home network. Calling my home phone made it, as well as the phone at my parents' place, ring, and it all worked just fine. I know that in the earlier days, this was a problem, but really, I just don't see it anymore.
SkypeIn doesn't cost anything over what they charge per year, which is pretty reasonable at ~40 USD a year. The call comes in like it was a normal Skype call and doesn't cost me anything. Of course the calling party pays whatever they normally pay to dial the number which you selected (so my parents pay LD charges).
I'm pretty interested in this SIP thing and may set it up at my new place, especially because it has a wide range of phones you can use. Big disadvantage with Skype is the limited amount of hardware out there.
<p>Re SkypeIn (from the Skype site):</p>
<blockquote>SkypeIn numbers are currently available in the US, UK, France, Hong Kong, Denmark, Finland, Sweden and Poland.</blockquote> <p>Boo. The countries I would have been interested in are Australia, Papua New Guinea and UAE.</p>
If you compare Skype with MSN messenger or Yahoo! Messenger it comes out pretty well. It's got some features that beat them hands down. So is Skype a good IM program with great voice, or a VOIP program with good IM. Trying..
Two other things to mention.
One is the IAX protocol. This protocol was originally designed by the Asterisk team, but is really starting to catch on. It's not nearly as popular as SIP for hard or soft phones, but a number of providers (including FWD and Voxee) are supporting it these days. IAX was designed from the ground up to be NAT-friendly. The other is that if you are using SIP to dial-by-IP, then yes, NAT will still pose a problem. I don't know of anybody actually doing that anymore, though. It's so much more convenient to use an agent such as FWD. (Of course, these agents are just used to establish the connection; it's still a direct peer-to-peer link once done.)
I consider the people who run skype (corporately) to be unethical.
The way that skype manages this (user behind firewall calls another user behind a different firewall) is it uses other users to route those calls. If you are a skype user and you are not behind a firewall, or if you are behind a home firewall that they can reconfigure with uPnP, you might become a supernode. Becoming a supernode means that skype users who are not supernodes (perhaps because they are behind a firewall) will use you as a relay. So, using the connection protocol, eventually, both machines make inside-out connections to the supernode, and the supernode (which is just a capable ordinary user who might not know that they have been elected) relays the traffic. It looks like a single call will saturate a dial-up. Thus, without your knowledge or specific permission, skype will automatically use some of your bandwidth to route other people's calls, people you don't know. This is not ordinary practice - it would be theft of service except for the clause in the user agreement. I've never met a skype user who knew that this happened. Most of them just know that firewalls present no impediment to skype - and they do not know why this is. This is unethical - it is, to me, as if they had a paragraph hidden inside their user agreement that allowed them to stop by your house and use your car while they were in town - and it was hidden in paragraph 63 of the fine print where you expect to find the legal disclaimers of liability, rather than being up front as would, say, the cost. People expect the bandwidth that they pay for to be there to use. No doubt many would volunteer to be supernodes - and many would not to. It is also true that many people have user agreements that specifically disallow them from running servers on their systems - and what skype does is a violation of this end user agreement. It would be ethical of them to do this if they made it clear that you were going to be routing stranger's calls even when you were not making use of the product other than simply being logged in bandwidth - putting this in bold at the beginning of the user agreement - making sure that people knew of this - allowing the function to be turned on or off - and perhaps only allowing people who left the function on to do the behind firewall to behind firewall connection. It is NOT ordinary practice to use people's memory or bandwidth to route your unrelated traffic, simply because they use your system. If skype wanted to do this in the standard manner - they would need some servers - and they would have to route the firewall to firewall calls through their servers. Of course, initially they were a free service, and they did not have the budget to do this. And, if they had banks of paid-for relay servers running as supernodes, firewall owners could block access to these servers and thus block skype. No one else does this - gets such good firewall to firewall routing - because all the other players in the game are ethical. To put this another way - supposing you were using a file sharing service - and they not only used your downloaded copies to feed other people who wanted copies of the files you downloaded (while and shortly after you were downloading them), they also downloaded popular files to you - and they used you as a conduit so that they could get the files to places that were intentionally blocking the sites you could access. This might make their file sharing service work real well. It would not be ethical unless they made it clear that they would be using your hard disk and bandwidth to share files you had no interest in. Furthermore. skype tries to pick ports that are not likely to be blocked - I've seen skype running on the port that https normally uses - a network administrator might have a legal requirement to record the destination of all calls made from his internal net - if he blocks all the regular sip ports and requires that those ports run through a proxy, well, skype will just pick some random port that is likely to be trusted and run its traffic on that port. Or so the rumors go. It might be that they use port 80 and 443 more or less exclusively. The simple thing to do is simply to fire any employee who installs skype on a computer that is company owned or plugged in behind the firewall - if your corporate policy restricts skype. But the reality is that network admins will get told "enforce this" and not be given any authority to have personnel actions taken when people break corporate policy. This usually is not a problem - firewall vendors can determine how protocols work and provide filters on ports and contents. Skype has taken steps to make this impossible - just so that people can say, "gosh, any firewall I am behind - it does not matter, I can make a connection to anyone." There are a couple firewall vendors who claim that they can block skype. They do this by doing packet analyss of all packets. http://www1.cs.columbia.edu/~library/TR-repository/reports/reports-2004/cucs-039-04.pdf is a paper that describes what it can of the skype protocol. They don't attempt to dig in to the protocol or decode the traffic (which would be possible with reverse compilation - they could trap the packets at the exit of the AES routines, for example. The nodes that are used to relay others' calls are called "supernodes" and you can't stop yourself from being elected as a supernode. From this protocol analysis, it looks as if there might be a simple way to do that - just bind to ports 80 and 443 - since it looks as if the initial connections are made via UDP and then 80 and then 443 to a supernode when someone connects to the net - so if you are already bound to those ports before you start skype, you can block the people behind firewalls from connecting to you.
That's very well put, and a good point.
You should also note that there are instant messengers out there which look similar to the user as google talk and skype but use SIP. One example is gizmo project (although not OSS).
Im in VoIP sea too :) Enjoying all that i would be able to
The reason Skype is popular is because it does builtin conferencing and allows two people behind NAT to talk to each other over IP. VoIP has been around for aaaaages, but nobody really used it until recently because it wasn't so convenient.
SIP does all of that as well. Once again, if you don't believe me, try downloading the Communicator from fwdnet.net. It Just Works also.
I agree with a lot of what you say. And perhaps you should do a little more research about Skype. But I think you should also recognise that Skype is far and away better than any SIP softphone. I've yet to find any SIP softphone that works as easily or requires so little configuration. The key things I find missing in SIP softphones are:-
- Common directory - Presence - Chat and IM features - Group chat - Voice conferencing And that's before we get to NAT and Firewall busting. If nothing else Skype has raised the bar in Voice software that "just works" without any sysadmin involvement to open ports or deal with STUN. But there's also an element of comparing apples with oranges here. All the major IM packages now support voice. Does that make them VOIP programs? If you compare Skype with MSN messenger or Yahoo! Messenger it comes out pretty well. It's got some features that beat them hands down. So is Skype a good IM program with great voice, or a VOIP program with good IM? Don't get me wrong, I wish Skype was more open and used SIP or at least had a SIP gateway. And I wish the SIP community would come up with clients that are as good. But right now it's got momentum and it's a damn good product. As for Google Talk. It's clearly unfinished and effectively an alpha, not a beta. *If* they ship early and ship often. *If* they play nicely with the rest of the jabber community and join the network. *If* they support SIP and interoperate with other SIP phone network. *If* they beef up the IM chat client. *If* they link up with POTS IN and POTS out services. Then maybe it will become interesting. But right now it's a toy.
It's difficult to do research about Skype because they don't publish their source code, nor binaries for my platform, nor make much information available for people that aren't customers. So you do make a somewhat fair point there.
At the same time, it seems to me that you haven't researched SIP all that much. The fact that SIP phones don't have a single directory is, IMHO, a *feature*. I can use FWD with my phone, or I can choose not to and use someone else, as I wish. That's a big plus to me. I have choice, and I can use whomever I want. Most of these places, though, do have directories. Conferencing, etc. are commonplace features in SIPland and work fine. As I've already said, NAT and firewall issues are rare with SIP these days, especially if you are using a central server like FWD (in a manner similar to Skype). It is true that most SIP phones don't support text. But OTOH, why would I need text if I'm talking to someone already? I really don't see the point for this integration. If you have audio (and even video), why bother with text as well? As for free/busy information, that's up to the SIP provider(s) you choose. FWD does publish that info.
Skype don't publish source, agreed. But what platform? Skype is available now for Windows, Mac, Linux. So what are you using? FreeBSD? Last time I looked they had info on their website about Skype-in, Skype-out, voicemail. All things you seemed to have trouble finding information on. Apart for my account data, I don't think there's any information that is only available to customers.
Directory is not integrated into the client in SIP. I'm on FWD somewhere but a quick google didn't turn up the directory. So on to the website. No mention of the directory in the FAQ. At that point I gave up. So then I went to check out configuration; Public IP Address (FAQ), STUN (FAQ), Outbound Proxy (FAQ) Ugh! Never mind all that geeky goodness, tell me how to call someone. My pointy haired boss would take half a look at that, give up immediately and call me. By contrast he installed Skype and started using it with zero help. Have you looked at the option pages on Xten recently? Would you describe them as dumb consumer friendly? NAT and firewall. One solution to this is to use a proxy reflector like FWD which you mention. This approach doesn't scale unless you have the resources of MS or Yahoo!. And it often involves round trips right across the net so two UK clients have to go via Seattle. My experience with this has not been good. By contrast, Skype does not use centralised proxys but distributed call setup supernodes. And once the call is setup, the link is P2P and no longer involves the supernode/proxy. So not in a manner similar to Skype at all, at all. "why would I need text if I'm talking to someone" To pass them a URL, among other things. IM and Voice go well together. They are both real time communication. Presence is more than just free/busy. And it's important. This is another reason why the convergence of Video, Audio, IM, Identity and Presence is important. It seems to me that the SIP clients seen so far (with the possible exception of Gizmo and FWD.Communicator) only really deal with one of these. This is beginning to feel like the usual back and forth of zealots. And I said earlier, perhaps it's because we're talking apples and oranges. Skype isn't VoIP, it's IM chat with good voice. SIP isn't real time communications or IM, it's POTS telephone replacement. And that last sentence is a sound bite and only partly true. As for Google Talk, I'm waiting for v0.3 because v0.1 is little more than proof of concept. A loose cartel of Google, AOL, Apple, Jabber, SIP could change the game completely. But not yet. They've got to prove they can ship early and often and fill in the blanks first.
Skype only publishes i386 binaries for Linux. Don't forget, Linux runs more than just i386. It supports more than a dozen other platforms. Everything from ARM to amd64 to m68k. It is extremely irritating to find people that assume that all Linux users are running on i386. (It's just as irritating to find software publishers that assume this.)
It's true I missed SkypeIn to start with, but I did not miss SkypeOut. It's also true that the info on Skype's site for both is rather sketchy, and for SkypeIn is so sketchy that it is almost useless. The rest of your comments reflect numerous misconceptions. If you think SIP is that hard, why not just download the Communicator from FWD and watch it Just Work? You are comparing apples to oranges. You are comparing the Skype client, which comes pre-configured out of the box to work with a particular service, to a generic SIP client that does not come so configured. *Of course* you will have to configure a generic SIP client. Or, you can get one that's already configured (such as the FWD one) and it will be no more difficult than Skype. As for dialing, you press the numbers and the call connects. Just like any other phone. Can't get any simpler. You are also misrepresenting how SIP pierces firewalls. It uses the central server only for *connecting* the call. After that, the traffic goes point-to-point. There is no fuss with sending data streams through a centralized server. I can see your point for text messaging, although personally I have *never* felt the need for that. SIP already is used for video. A little research revealed that SIP supports RFC2793 for text messaging, and some programs, and even Polycom hardphones, support it today. Still, I'd rather use my preferred IRC client for my text messaging, and my hardphone for voice, and if I have them both open simultaneously, fine. I don't want to be forced to use the same client for both. If Skype is just "IM chat with good voice", then why are so many people still so excited about it? This is not unique in the marketplace by any means. Apple's iChat does it over AIM, Microsoft's programs have been doing it for ages, and GnomeMeeting does it on Linux (with H.323). So you're left with my original assertion that Skype is boring because it brings a whole bunch of restrictions and nothing novel to the modern table. SIP is not the only alternative to Skype (I even mentioned IAX in my article). SIP is not the protocol to end all protocols, of course. It has its weaknesses, and I personally prefer IAX. But let's have a fair, apples-to-apples discussion, shall we? A lot of people are hyping up Skype as a POTS replacement. Probably far more than are commenting about being able to do text chat with it. Even Skype's own homepage discusses only the voice features. It seems that voice is what people are excited about, so voice is what I've discussed. Make no mistake, text chat is nothing exciting either. After all, 4.2BSD, released in 1983, contained talk(1).
Great post, thank you for taking the time to spell out what options we have.
My question is: what should I use? I have been w/o a landline for two years but now have the need for one. I don't have any phone infrastructure to worry about abandoning, so anything is game. I have broadband and a wireless network. What do you recommend?
That depends on what you're wanting to do and your level of technical expertise.
Let's say that you just want to do pure-IP conversation. That is, both you and the people you speak with have Internet phones. The easiest way there is to go over to <a href="http://www.fwdnet.net/">FWD</a> and download their communicator. It's pretty much a click-and-go proposition. Perhaps you want to be making calls over the Internet to regular (PSTN) phones. The first thing to do is find a <a href="http://www.voip-info.org/tiki-index.php?page=VOIP+Service+Providers">provider</a>. I like <a href="http://www.voxee.com/">Voxee</a>, and you can get started with them for as little as $5. They're more oriented towards technical folks though, so if you're not technically inclined, you might try someone like <a href="http://www.broadvoice.com/">Broadvoice</a> instead. Next, you need a SIP phone. Good starter phones would be the <a href="http://www.sipura.com/">Sipura</a> SPA-841 for a hardphone (about $90 <a href="http://store.voxilla.com/customer/product.php?productid=16151&cat=254&page=1">from Voxilla</a>). For a <a href="http://www.voip-info.org/tiki-index.php?page=VOIP+Phones#id363973">softphone</a>, I recommend the free <a href="http://www.voip-info.org/tiki-index.php?page=SJphone">SJPhone</a>.
I was considering buying a SPA-841, but aster a long consideration and by the same price I got my SPA 3000, which allows me to connect any number of analog phones in parallel to the Internet, and at the same time I can pick up land-line call with the same phone. And of course I can choose routing conveniently by Internet or by land-line.
The SPA 3000 is an intelligent an very configurable telephony router. It has three connector. 1) Analog phone 2) Ethernet 3) Analog land line. The limits are your imagination, and you can make it work with asterisk for extra possibilities. My 2 cents.
Yup, we use one to integrate our PSTN line into our Asterisk system.
Make sure you upgrade the firmware to the latest version. It has some features that make this nicer.
Our product, PhoneGnome, is an option as well. It is a simple plug-and-play device that brings SIP to your plain phone and existing phone service. Simply plug PhoneGnome in and join the SIP community. SIP calls and your existing plain calls on the same phone, and next generation SIP features even for your old POTS calls. like voicemail to email, online call logs (including POTS calls), and online phonebook with click-to-dial. PhoneGnome is easy enough for non-technical users to use, but still makes SIP to SIP calling a reality for anyone, and makes it easy to use a SIP PSTN termination provider of your choice (optional -- you can just keep using your landline service for non-free calls). Plug PhoneGnome in. Call any number just as you do now and some will be free. It's really that simple. Dial SIP addresses too.
I find it odd that you fail to talk about voipbuster.com which hands down beats skype and google talk. It's free to use after you send 1 euro via paypal and works great. Was this done on purpouse?
I'm not trying to list all the possible VOIP providers here; there are hundreds or thousands of them. I did, however, link to the voip-info wiki pages about providers, and VoipBuster is indeed on their <a href="http://www.voip-info.org/tiki-print.php?page=VOIP+Service+Providers+Residential>"residential</a> page.
I bought 11 EUROS, used only 4 EUROS and my balance tells 1.80 EUROS.
Besides I cannot used any more, you are teling me " there are problems with your account" and thats all. WHAT CAN I DO? to you need my money, Im very sorprised, this is the 5 letter I send you.NO ANSWERS>>>>>>> bye. Federico Kristal
What program are you talking about? I most certainly have no program that I'm selling here. And I have not heard from you before.
I've been using voipbuster with a Cisco SIP phone to make free calls (to certain countries) for a couple of months. Except for once accidentally calling a Swiss mobile phone, my 1 Euro is almost intact.
It will certainly not stay free forever, but it sure is nice for now.
SIP is the <a href="http://en.wikipedia.org/wiki/Session_Initiation_Protocol">Session Initiation Protocol</a>, and as the name implies, is primarily responsible for controlling calls -- placing them, handling transfers, etc. Other protocols are responsible for encoding actual audio data and streaming the audio data (RTP is used for the latter). It's sort of a family of protocols.
I currently have Skype and Vonage in the caribbean with a DSL connection. Very often (almost always lately), I get garbled voice on Vonage but Skype is working fine. Can anyone recommend another solution so i don't have to keep the PC on all the time.
You might try an ATA such as the <a href="http://www.sipura.com/">Sipura</a> 100. There may also be patches out there to hack the vonage ATA to work with other providers. With other providers, you can use different codecs that use less bandwidth and are more tolerant of network issues.
check out <a href="http://www.openwengo.com">OpenWengo</a> - this french based startup is pushing FOSS skype alternative. sources available of course, and builds for windows and several *nix flavors.
OpenWengo looks good. Finally an open source one... and I read on their forum they plan to interoperate with Gizmo.
Gizmo being SIP based (it even gives you a SIP number), along with Google Talk's plans to be, is good news for potential interoperability between all these systems. The main selling point for these systems is they are "open unlike Skype". Google has also expressed interest in interoperating with Skype. I hope that puts some pressure on the Skype project.
I heard Jabber is implementing SIP. Does anyone know how this is being approached? I'm guessing we will still need to hook it into a SIP provider somehow?
The founders of Skype are certainly laughing all the way to the bank after re-packaging an old idea in a fancy dress and selling it to eBay for millions. The way to make an insane amount of money is by creating a closed standard which becomes the only standard.
I think that voip with SIP is only very slowly gaining ground with regular users. Things like WiFi phones, analog telephone adapters (ATAs), routers with voip functionality and PDA soft phones are making it more accessible. Only by educating people about the importance of open formats like SIP can the voip industry flourish and users see a lot of advancements. So once again thank you for your great post. In some way Skype is making people aware of voip and hopefully makes them interested in open SIP voip.
I see a lot of missing things.
- Voip island - If you subscribe with any provider should be able to call any user at any provider. ( The way actually email is working ) - Most SIP phones don't support text - Try do dial in a phone bugs-bunny@whatever.you.like.com - Easy Dial - If the above its a nighmare give us a working method "now" to convert text to number , realm to as number and so on. - NAT and Firewall issue - People dont want to bother with tecnical expertize. - Convergence of Video, Audio, IM, Identity and Presence - People want to comunicate in any ways.
This is amazingly useful and comprehensive information, for which the author deserves a lot of credit.
One comment, though: why the totally superior tone? Skype, Google etc. are made up of ordinary people trying to do creative things. They may not get them totally right or flavour them to everyone's liking, but there is no good reason to diss other people or businesses (especially if they are successful, becuase then it looks like "sour grapes"?) Thanks for the great info!
Have you check www.efonica.com. They use sip , good voice quality
I think Skype just has better audio compression algorithms then ones used for SIP. I tried calling VoIP using several telephony cards, I believe big companies use SIP. And the sound quality is far worse. Same for my company internal phones which also uses VoIP, I can talk better to the same people using Skype, especially when network load is noticable.
how to block google talk using linux proxy server on client machine
I used SIP side-by-side with Skype for more than a year until I got sick of all the problems I had with XTen where it wouldn't hang up properly, calls that were coming into my land line number wouldn't be received properly. I don't think it was at the time (maybe two years ago) it was ready for the mainstream. I agree with earlier posts saying that Skype's biggest contribution is that it has made VoIP easy to use for the end user. If something takes more than clicking on the 'Next' button to configure, it is too much to the average user, let alone having a client that doesn't seem stable at the best of times. I agree that things should be open standards, but technology like this usually takes a corporate like Skype or (and I shudder to say this) Microsoft to promote the hell out of something and get users understanding what is possible.
I think the guys over at the Gizmo Project have got the right idea behind that their mix of open standards and design. It must be remembered that Skype is popular now, because it provides something that is rarely available in the open source world (and I am a big user and fan of open source), which is a consistent user experience. Users don't care what standard is underneath a piece of software they are using as long as it makes their lives easier and it works ALL the time. I haven't tried the packages that you mentioned and I do believe you when you say they "Just Work", but unless they can, from the users POV, demonstrate a CLEAR advantage to the lay man over Skype, you are not going to see a shift away from the commercial packages. Firefox pulled market share away from IE by providing features and speed that were OBVIOUSLY superior to IE.. but even it still does not hold an enormous market share because most people still don't see enough of an advantage to stuff around downloading and installing something unfamiliar. This is even more pronounced for Open Office, where people would prefer to pay Microsoft's high price than switch away for something technically better, but functionally the same (with worse graphics) from the average user's perspective. The fact that SIP didn't have a good a user experience when Skype came on the scene, regardless of what it is now, is why Skype has the market share. So until SIP can demonstrate to the END USER that it is markedly better than Skype it will not draw the public's attention in any major way. Google's entrance into the market complete with SIP is probably the best thing that could happen for the protocol as they have enough of a user orientated philosophy to make the innovations needed to create a clear distinction between Skype and the rest.
Google talk will be free soon to the PSTN powered by a parntership with Bandwidth.com using their SIP gateways. Bandwidth.com is using Googles network for transport.
I have a skype account password is trapper 14 snopping cart # 115677909 my nami is kennetwillardocherty
How amazing that somebody with your level of knowledge can miss the point of these two programs entirely.
Or maybe it's because of it. You know, kind of "not seeing the forest for the trees "thing But good luck with your marketing. J. |
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