SIP vs. Skype, Making the "Open" Choice
techie34290 writes "If you were to make the choice between SIP and Skype for Linux, which one would you go for? Matt Hartley from MadPenguin.org says to opt for SIP. Why? "One tidbit of information that most people are not likely aware of is that when you install the Skype client, it will drain system resources by running as a supernode from time to time. Granted, this is not always the case; however, the very idea of my PC having its resources tied up for someone else's phone call is frankly maddening to me."
Cause I've cybered there with women I met from SWG back in the day K.
The sacrifice you make to have a 'free' phone service is your system resources. Without donating your clock cyles, you would have to pay skype to use servers that they would otherwise have to provide. And yes, if your system can handle it, it will act as a supernode from time to time. It is your choice, but very few things are truly free.
It's getting to the stage now where I know people who say "Skype" when they mean "VoIP/SIP". Admittedly, it's early days in terms of adoption of the technologies, but this is a little worrying. Seems like very few people in the real world have any concept of open standards etc.
of my PC having its resources tied up for someone else's phone call is frankly maddening to me
... but I have no problem with tying up someone else's resources when it's for my convenience ... ?
the very idea of my PC having its resources tied up for someone else's phone call is frankly maddening to me."
So that means you don't use bittorrent either?
"the very idea of my PC having its resources tied up for someone else's phone call is frankly maddening to me."
Well, boo hoo. It's the way the system works. I seriously doubt any significant system resources would be used up for other people's calls. When you make your calls, it happens to other people. It's a give-pull situation where everyone has to share resources in order for the system to scale with the number of subscribers. Would you rather have nothing?
- It's not the Macs I hate. It's Digg users. -
If you don't want to run as a supernode, don't make voip. Scumbags like you don't deserve to use any community service. Good luck downloading from a filesharing service, you'd be banned by everyone you tried.
LEECHER
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Sure, I prefer open solutions, but to say that Skype will drain your system's resources is just crap. A simple consumer firewall between your skype-running PC and the internet will prevent Skype from using your PC as a 'supernode'.
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So - it's cool if others donate their time (and other resources) to give you something for free (SIP - both the protocol, but also a LOT of the programs for it), but if YOU have to give something of your resources (like bandwidth) for free - well, that's something COMPLETELY different.
Somehow I don't see how that works.
Sure, Skype is proprietary - but I've never paid anything for it (except bandwith), and it works just fine for me, so - to me at least - it's free (monetary, not libre). SIP - well, never tried anything that worked just as well as Skype, so it's libre, but it's not free to me (costs me and others resources like their and my time to get it working).
I don't really see the difference (but I'm not a fanatic proponent of Libre software).
We do not live in the 21st century. We live in the 20 second century.
"If you were to make the choice between SIP and Skype for Linux, which one would you go for?"
IAX I guess...
if all of that stuff was legal down here. That is not so clear.
all the best,
drew
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but it's fine if you tie up someone else's resources for your own phone call?
Article translation:
SKYPE: OMG! A supernode! you gotta be kidding me! You mean if I turn it on, it might use more bandwidth than I imagined? And if you use it to make phone calls, and lose your password, you probably won't get your money back.
Gizmo: Well, at least it uses SIP.
Full Open Source SIP stuff: Now this is the way to go. Too bad there's not much out there anybody else uses.
Okay, it's Mad Penguin, but who exactly are we preaching to?
Supernodes. Yeah, skype does that, and it can be a pita. If skype is running more than 4 contacts, you've been elected. If you don't like it, shut it down. If you can't monitor your network activity, and are running Linux, what kinda geek are you?
Terrible news if you lose your skype password, you might lose up to 25 bucks! If you were using an open-source alternative, you wouldn't have this problem, because you wouldn't be making or receiving PSTN calls.
The #1 reason why I use Skype over SIP: It's encrypted. At least that's what they tell me. Give me a solution that's F/OSS and uses point-to-point encryption, and I will switch to the superior product. #1 reason why others use Skype: it just works: those supernodes do their job and it blows through most obstacles those idiots in IT try to put in the way. Turn it on, it connects and it works.
Another interesting Skype weakness: A second client can be connected to skype under the same account, and will receive a copy of all correspondence without the other client knowing about it.
Its worth considering Google Talk (Jingle protocol).
It uses XMPP (Jabber) then kicks up to Jingle for voice.
Nice.
"With Skype, I bog down my PC, and still have to look myself in the mirror as many of my friends shake their heads with disgust." - surely running as a super node for 5 minutes a month isn't that bad.
Until the competition let me call a landline through their product for free, as Skype does at the moment, I won't be switching.
the very idea of my PC having its resources tied up for someone else's phone call is frankly maddening to me
This strikes me as an attitude of someone who can't stand the idea that he's not in control of everything (which you never are). The real question is, does it use any significant resources that effect what you're trying to do at the time? Frankly I don't really care about 20-50 megabytes of memory, or 5% of my processor usage, or even 100% of my idle processor usage. Those numbers are all low enough that you'd more than likely never notice or miss those resources. I would be concerned if the app started taking up hundreds of megabytes of memory, or 30-40% of my processor time, or locked up system resources that interfere with other apps I'm running. So which is it? The author didn't provide us with any of that information, only the extremist position that ANY useage of his computers resources that wasn't for him was unacceptable. What a useless article.
AccountKiller
I recently discoffered jabbin.
http://www.jabbin.com/int/ it's free as in speech, and has voip support.
perhaps he should give it a try. there are windows, linux and mac releases.
Of course, this wonderfully balanced, well-researched and well-written article did overlook one rather important point: The whole supernode concept lets Skype get through nearly ANY router configuration imagineable. I beat my head against a brick wall for three months trying to convince my father's ADSL modem to let through voice comms for a SIP client. No go. The only way to get a VoIP solution through was to install Skype.
The guys's right about SIP being better, but his reason is wrong.
SIP is better, because it allows more interoperability (open standard, duh!).
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I love Skype, I really do. The program is slick and works well. But they have SERIOUS problems handing purchases for Skype-Out (their PC-to-landline service). The complaints are numerous ... so numerous in fact that Skype has had to set up a separate forum to handle them all. Here's a rundown of my personal nightmare with them. You really, and I mean REALLY, need to look at the problems that Skype is having before you go with them.
Back in July, Bob Cringely at PBS had a column in which he talked about Skype and its use of super-nodes.
Frankly, I'm surprised to hear that people, especially Slashdotters, still talk to each other. I'm guessing this "voice" thing is some kind of emergency backup in case your email and instant messaging clients go down.
One word: laziness
Longer reason:
I really didn't want to have to learn the intricacies of a protocol in order to get everything up and running. I'd been seeing various things that at least *implied* that I'd have to start mucking about with firewall rules in order to get SIP running. This was something I had zero interest in doing at the time. I also wanted something that would be cross-platform in an easy manner (I was looking at this so my manager & I could keep in touch while we both worked from home, without using cell minutes or LD charges....him Windows, me Linux).
Then I needed to be able to call out to POTS lines. Enter SkypeOut. No monthly charges, just a relatively low per-minute charge (which was OK, I still, after 16 months, haven't used the initial 10 Euros I put in). Then I needed people from work to be able to contact me, and I didn't want to give them my cell or house lines. Enter SkypeIn. $38 for a full year, with voicemail. Usable anywhere. That was a big draw. So long as I had a network connection, I could head off to a family member's place for a long weekend out-of-state, and still be reachable. No problemo spending 8 hours with sun and surf in the background and me being several hundred miles away...
And then things soured... I tried to renew my SkypeIn number. Failed. Again and again and again. Skype's purchase process is rather....opaque. They use a variety of 3rd party payment processors, and all Skype can tell you is "success" or "failure" until you start bitching about being unable to pay. Though don't expect an immediate response, as it will take up to 4 days. And then, if you're like me, you'll be told that your NATed laptop running Linux on a static IP with no proxies is "an anonymous proxy", and be told to check you IE settings to ensure you're not using a proxy (yes, IE settings in Linux...). You'll be told to, get this, try a different ISP. And even though you'll have already tried multiple credit cards, and multiple browsers with them, you'll be told to try another credit card, or another browser. Or worse (IMNSHO), another payment method that until just recently announced, had practically NO consumer protections (way to pay, pal!).
Ultimately, I ended up with access to work's terminal server, and after one too many complaints from the muckety-mucks who'd already been given my cell number (remember I didn't want to do that?) because I couldn't renew my SkypeIn, I decided that I could make a business case for using company resources for attempting to renew. And....it worked. NFC why, but it did.
So, now I have about another year to come up with another solution that'll work for me on random networks, doesn't require special hardware (other than a headset or speakers+microphone), and doesn't have recurring monthly fees, as I don't actually make calls every day, or even once a week. SIP still gives me a headache just trying to wrap my head around. Trying to figure out WHICH providers offer WHAT parts of what Skype offers as an all-in-one package is something I tend to just grow bored trying to research. Some of the more promising-looking clients seem to be geared towards specific providers, while others leave you trying to guess who to go with. Ick. I really don't want to stick with Skype having experienced the bad side of things, but I'm afraid momentum an just how unfocused SIP solutions are for what I want will force me to stay.
(Let's not even get into the whole Skype's Linux client lagging way behind their Windows client, with the Mac client having leapfrogged Linux at some point. There HAVE been a few betas that have finally brought support for ALSA, and some UI improvements. Still miles behind Windows & Mac, which is frustrating, but there's been at least some progress now almost a year after their last major Linux release)
"The urge to save humanity is almost always a false front for the urge to rule." --H.L. Mencken
Skype's software works surprisingly well in these environments compares to software from other providers. I have not tried any of Skypes HW based options.
An engineer from one provider whose hardware solution I was testing and who was actually trying to be helpful said that my problem was with SIP's Jitter Buffer not being able to handle the ghastly latencies that I've been experiencing during my journies. If I hadn't already had problems with 2-3 other providers, I'd have thought that he was blowing smoke...
I'm not sure but I don't think GnomeMeeting/Ekiga is clever enough to automatically punch through firewalls like Skype/Gizmo. Therefore users (ie grandma!) will be expected to alter their broadband router settings to open a port. Frankly, that's asking too much. It's also insecure.
don't like skype because of closed source, p2p and not open?
keep your eyes on wengophone then: free, open source, supports msn, jabber, yahoo and other protocols, uses sip, can send sms, video calls, can handle conference (just up to 3 users for now), and it even costs a bit less than skype on skypeout. it only lacks the skype-in function and crypted traffic, but both will be integrated in near future. for windows, linux & mac for now, they're developing even a firefox extension and one for pda with wince.
Wow, that has GOT the be the LAMEST excuse ever for choosing sip over skype.
He gets crybaby points from me...
Move sig!
It's SIP, It's GPLed, multiplataform and soon multiprovider. Has anybody given it a try ?
Skype is a closed-source resource-eating protocol. SIP is an open, standardized protocol which projects like Asterisk use. I think telephony should be a PPP (Point-to-Point Protocol) and not a Kazaa-like P2P (Peer-to-Peer protocol). First of all: what information passes the supernodes? Can anyone see or analyze who you're calling, when and where? Second: what is the bandwidth usage? I can call SIP through 56k, can I call Skype as SuperNode on 56k?
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I was wondering how I reached it so quickly, while downloading nothing [but the occasional update]. Yes, we in .za have 10GB caps!
Skype has video, which is surprisingly usable. SIP does not.
Avoid becoming a supernode by blocking the inbound port, or by closing Skype.
Skype is the only proprietary bit of software that I use, but I am reluctant to use another simply because the rest of the known world (known to me, anyway) are using Skype. So if I want to communicate with them, I have no choice BUT to use it.
I would desperately love to use an OSS version, but I can't do so without severely limiting my contact with my friends/relatives.
I actually use Skype solely for it's paid-for services (SkypeOut, SMS, etc) as I live abroad, and don't want to pay extortionate rates for international calling.
This isn't covered in any SIP clients that I'm aware of, thus making this debate irrelevant for me at least. My only gripe with Skype for Linux is that they haven't ported over the SMS functionality yet - it, for me is a reason on it's own to use Skype.
throw new NoSignatureException();
If someone would port Ekiga (http://www.ekiga.org/ formerly GnomeMeeting) to Windows, there might be some hope for SIP!
:)
The key to Skype and SightSpeed (www.sightspeed.com) is that they are cross-platform and have solution to the NAT problem. But they are both non-standard. And they both work really well
Ekiga has a solution to the NAT problem using STUN servers, but until someone writes a Windows port for it, there will not be massive usage of Ekiga.
Skype is a closed secretive disaster. The community RANTED when MSN/AIM/Yahoo messenger played games to cut out client choice (Gaim, Trillian, et al.). If you want simple, use Gizmo ( http://www.gizmoproject.com/ ) which has a very economical and functional interface to the regular phone system (POTS). Even cheaper, try http://www.voipstunt.com/ or http://www.freecall.com/
All of these are street-legal SIP, and you can use any SIP-capable device you like, or use your computer if you want to.
And of course you can use Asterisk ( http://www.asterisk.org/ ) which is best of all!
Skype belongs in the shitbin of history. Closed systems suck.
"the very idea of my PC having its resources tied up for someone else's phone call is frankly maddening to me."
But having somebody else provide the same service for you is no more than your due, right, Your Majesty?
Why yes, I AM a rocket scientist!
My business of approx. 40 people recent moved from a fancy, very expensive centrex phone system to a Asterisk based IP phone system. Overall, our results have been very positive. Part of our plans, however, were to have approx. half of our users use Softphones. We have tried probably about 2 dozen IAX, SIP, and proprietary softphones, and to this day we had not found a single one that worked very well, exactly how we wanted it to. Our requirements were fairly simple:
1. Had to be able to connect to outside phone numbers, not just other clients on the network.
2. Had to work with our IP-PBX (Asterisk). This ruled out alot of the good, proprietary clients like Gizmo, Skype.
3. Since alot of our users travel ALOT, the phone had to work flawlessly through firewalls. Although we can configure the firewall on our end, when our users are at hotels, etc, they couldn't connect. This pretty much ruled out SIP altogether in favor of firewall-friendly IAX..
4. Voice quality had to be excellent. Too many users would be using it as a primary phone.
5. Cost wasn't an issue, if the phone was good.
6. The phone would be used by ID10T's. It had to be simple and intuitive.
All I want to say is that although there are a ton of them out there, finding a good softphone is harder than you think. There's lots that fit indivdual users needs very well (I use skype with skype-out and it works great). The best we had found overall was idefisk (asteriskguru.com). Although, it leaved alot to be desired as far as voice quality goes. Still, we're searching for a good softphone for our mobile users.
Granted, this is not always the case; however, the very idea of my PC having its resources tied up for someone else's phone call is frankly maddening to me.
This same guy probably runs BitTorrent and the Blizzard Downloader for WOW with no complaints.
What it comes down to is that Skype plain has better sound quality. It sounds better and with less latency than all the competitors I've tried. Period.
Comment of the year
The Windows XP version of Gizmo silently installs some crapware called Bonjour. This pos will spawn yet another background process which then tries to communicate with external servers. Gizmo project hacks: "Are you listening?". This is a very, very, very bad practice! The crapware can be disabled, and you can find out how by searching the Gizmo project forum.
I think that comparing Skype vs SIP is like comparing apples vs oranges.
SIP is just a session protocol, so it has nothing to do with audio quality (compare codecs then: ilbc,speex,g723,g729), free calls (compare services skype,vonage,people call,..) or applications cpu usage.
Thanks for pointing out the obvious thing that people forget. Let me go further and say:
1) Skype is closed and a single metamodel that's been implemented nicely and virally (not that it matters)
2) SIP (and ENUM) are perilously prone, not because they're protocols, but how the protocols are implemented, to shenanigans. SIP is natively text, and ENUM is a DNS method that's prone to spoofing and other problems. For now, Skype wins only because few people know how it works at its deepest levels.
3) Skype isn't as extensible as the SIP/ENUM combination, and it makes one dependent on a single (if diverse and highly peered) network.
4) SIP and ENUM don't care about the service and are largely service neutral (some coming problems, here, though, as it doesn't do nice things like embue codec choices, encryption/authentication means, and other security niceties).
5) Skype is one closed vendor, very few business partners, while SIP is a technological infrastructure that invites whomever to do whatever.
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Upset about it tying up your system's resources?... ehh. I don't have hard evidence to support this claim, but I'm pretty sure the average desktop CPU doesn't operate at 100% most of the time. Neither does their "comfortable" network bandwidth (i.e. more could be being done without noticeably adversely affecting anything). If you can skim a little off the top to help someone else out without actually interfering with what you need to do with your computer, I don't see that as too much of a problem.
Think of it as going into a building and holding the door open for someone following a little bit behind you.
If you're really that concerned that your computer should never be used to help anyone but yourself, I'm sure you could figure out some way around it. If you're like most other people, I bet you could get away with simply lowering the priority of the process...
I caught the Mountain Wumpus! He gave me his treasure chest ($100) to let him go free again.
A few downsides with Skype are that it may act as an intermediate for other callers and it is relatively CPU-hungry.
A better protocol than SIP is actually the IAX2 protocol (Inter-Asterisk protocol v2). At least it's better from the firewall point of view, but has the drawback of not being very widely supported outside of the Asterisk domain.
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Maybe there is some other reason to choose SIP. It's openness (as in OPEN).
You can choose you favouite service provider (even more than one) and your favourite software (even more than one).
Yet, if you dare enough, you can run your own service or write your own SIP client.
Nothing of this can be done with Skype (and similar initiatives).
Some more interesting considerations about Skype can be found at here, written some time ago by Bob Cringely.
And if you find these latter things interesting, you'd give a look to this project.
Maybe Computers will never be as intelligent as Humans.
For sure they won't ever become so stupid. [VR-1988]
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Well, the problem with SIP and Skype is NAT. The phone/computer doesn't know it's WAN IP to establish connections with other phones, and there aren't necessarily inbound UDP ports open to route UDP traffic to the phone.
;) ) both ways once you've estabilished an outbound connection. For VoIP with both users behind NAT, however, this is unlikely to work.
r aversal3.html
:)
NAT routers temporarily accept inbound UDP packets on a port when there has been an outbound UDP packet on that port (aka UDP pinholes). So you get a working UDP "connection" (well, stateless
Skype gets around this by using computers that aren't behind NAT to route traffic between two phones that are behind NAT. So if everyone was to block this behaviour, Skype just wouldn't work for NAT users. It requires some community spirit (even if this is unintentional on the part of the user).
SIP systems often employ STUN servers that allow a phone/computer to query the server to find out what its WAN IP and NAT type are (and use the query itself to open up temporary UDP inbound ports on the router - something that works with all NAT types except symmetric).
There's a description and some pretty pictures of how STUN works here: http://www.newport-networks.com/whitepapers/nat-t
In addition, SIP is also an open protocol, so there is nice free open-source software available (Asterix) to allow you to set up your own home switchboard (calls from different outside lines can be routed to different phones - IE, whenever your daughter's boyfriend calls, it can be routed to ring her VoIP phone). Skype is proprietry so you won't get any customisable features like this.
So really, SIP is the way to go if you're a supporter of open standards, and Skype if you want to follow the headless masses.
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My only real gripe with Skype is that the protocol is secret. This means that no alternative implementations can easily (or even legally, depending on jurisdiction) be created. Want to create a client that doesn't need a GUI, implement one in hardware, make one for OpenVMS, or create one that is verifiably free of malware? You can't without the blessing of Skype, Inc.
I refuse to use secret protocols or file formats wherever feasible, and it's definitely feasible in this case.
Please correct me if I got my facts wrong.
I tried several different VOIP programs some of which were based on SIP and set up fwd accounts and the like but in the end we chose skype for one reason only: the call quality was superior. Most of our calls are placed over 802.11 networks in remote areas that have no cell phone coverage and use satellite for the internet connection.
Some calls have to make it through seven access points and then out over 800ms of latency and skype calls actually still are very usable and even sound good. I don't know how they pull that off. I'm willing to put up with being a supernode for that.
Because ID10T's can install it. Its cross platform and easy to use. We use it to talk to my Cousin in Thailand or Korea. My Aunt and Uncle have it in Oregon. It is also able to punch through firewalls and NAT problems extremely well BECAUSE THEY USE SUPERNODES to connect the call.
Also here is a good "Whitepaper" on the skype protocall
http://arxiv.org/ftp/cs/papers/0412/0412017.pdf
About the only advantage Skype has over the competition is that it Just Works in a variety of firewall and NAT scenarios. About the only trick it uses that no open protocol system uses (to my knowledge) is a peer-to-peer proxying network that is used if direct communication between peers is not possible. It seems to me that such a network can easily be implemented in a generalized (i.e. protocol independent) fashion.
I've been thinking in the direction of a TUN device connected to a proxy node with a globally accessible IP address, where any port being opened on the TUN device is also opened on the proxy node, and all traffic sent to the proxy node is forwarded to the TUN device (effectively creating the illusion that the TUN device is a network device of the proxy node). This system can be largely transparent to applications, except that they have to use the globally reachable IP address instead of whatever other IP addresses the machine may have (e.g. the same problems that always occur when a machine has multiple network interfaces).
This scheme could be extended with little effort to include hiding of the real source of data and encryption, thus providing a degree of anonymity and confidentiality, all transparent to applications.
So far, it's a nice dream. I started mucking around with TUN/TAP, but I couldn't see any of the packets I sent to the virtual device, and I couldn't find good documentation to tell me what I was doing wrong and what I should have been doing instead.
Please correct me if I got my facts wrong.
Skype doesn't use a lot of bandwidth, so even if you're running in supernode mode it's not going to make a big difference, except maybe if you're on a home DSL with 128kbps upstream bottleneck, and the FUD's targeted at businesses and universities that have much larger connections (i.e. places that make decent supernodes.) It's good to hear that Skype's doing some consulting and proxy-server work to help manage that kind of business.
Skype does have the problem that they're a rabidly closed-source company with rabidly closed-source protocols, so it's partly their own fault. In contrast, BitTorrent is wide open - so if you want to do things to reduce its bandwidth consumption, you can, though it's also designed to get around firewalls whenever it needs to.
Bill Stewart
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Just so you know, I started with Skype, and it works just dandy. If it drains away my resources, I haven't noticed it. But, I've recently had the pleasure of installing asterisk at home, and it is the way to go! If you install and use Asterisk, you have the widest choice of phones (including soft phones) and VoIP endpoint providers, and you have the flexibility of changing your mind about any one of those choices at any time without disturbing the rest of the system. In my recent conversion, I converted the whole house to use Asterisk without changing any of our phones. My wife doesn't notice the difference - she just calls normally and it works. But, behind the scenes, I can selectively route different calls to different networks - hard wired or VoIP - to take advantage of whatever route I decide is the best. If any one of those routes starts to irritate me, I can change it without affecting the rest of the system. Try that with Skype.
Now, a common argument you might get against this approach is that it's unneccesarily complicated and requires a dedicated machine. Well, it may be partially true, in that it's more complicated than installing a single SIP or Skype phone or softphone, and the best (IMHO) approach for an install takes a surplus box; however, the TrixBox distribution gets you up and running awfully fast, and can be installed onto a crap machine (I'm using a celeron 500). Follow the How-To here. The flexibility is worth it. And, if you have a decent net connection and VoIP provider, the call quality even for VoIP is outstanding.
Other advantages are flexibility in call routing. I currently have a digium TDM400P card hooked up in my install, with one module hooked up to the phone line, and the other module hooked up to all my analog phones in the house. (I'll eventually replace some of the analog phones with some nice IP phones when I have the cash.) I could just as easily add SIP softphones connected to Asterisk, if I wanted, but normal phones seem more natural to me, and it's cheap to do with the TDM400P card. I have three inbound and outbound trunks set up, one using the land line, one using VoipJet for long distance over VoIP, and one for calls in from and out to the Free World Dialup SIP network. I have my dial plan set up as follows:
Any calls coming in from either my old PSTN landline or my Free World Dialup account are routed to my dialplan, which during the day (6AM to 11PM) rings the analog phones. If the caller is blocking caller id, it forces them to enter their phone number first before ringing the phones. At night, (currently defined as 11PM to 6AM) callers are sent to a VRU, which asks them to hang up if they're a phone solicitation, press 1 to actually call us, or 2 to send the call straight to voicemail without waking us up. In either case when it rings the phones, it will go to voicemail if we don't answer. That voicemail can be retrieved either by the phone, by secure web interface, or currently I also have it email me the wav file of the message.
For outgoing calls, I have it set like this: If you dial a seven digit number, a toll free number, 911, or use a 9 prefix before a long distance call (in case my network connection is down), it dials out through the land line. If you dial a long distance number normally (using just 1 + area code + number, or 011 + country code + international number), it routes it through the IAX2 trunk to VoipJet and saves us tons of money. If you dial a 8 or 393 prefix before the number, it assumes you want to call a FWD number, and routes it out the IAX2 trunk to FWD, which would be a direct SIP to SIP call for free.
In summary, it works awesome, and I had the whole thing working in a basic way (PSTN + analog phone + VoipJet trunk) in one Saturday morning. I had rerouted the whole house's phone system and revam
Life is but a mist upon the horizon.
Maybe there is some other reason to choose SIP. It's openness (as in OPEN).
You can choose you favouite service provider (even more than one) and your favourite software (even more than one).
Yet, if you dare enough, you can run your own service or write your own SIP client.
Nothing of this can be done with Skype (and similar initiatives).
Some more interesting considerations about Skype can be found at here, written some time ago by Bob Cringely.
And if you find these latter things interesting, you'd give a look to this project.
Maybe Computers will never be as intelligent as Humans.
For sure they won't ever become so stupid. [VR-1988]
I don't think this has been brought up yet - SIP has NO standard authenticity to its caller-ID unlike Skype. I can just go online, and pretend to be your Mom, best friend or your employer, just by asserting the FROM address, just as one can with e-mail. E-mail has an excuse - it was invented in a fairly closed internet where there was a lot of mutual trust. There's no such forgiveness for SIP.
I have called a couple of people today with SkypeOut for international calls. I have chatted with individuals and on the company grpup chat. I am behind a firewall in a private address and so are all my buddies. All this is encrypted, except the POTS part of calls. I can immediately call 7 million of online Skype users. I have SkypeOut, SkypeIn (my GSM phone is in another country currently, I have transferred my calls to SkypeIn), I have Skype voice mail, I can send voice messages to any Skype users, I can send/receive SMS, and whatnot...
I also have Gizmo and X-lite and some other SIP clients installed, and Gizmo even allows free calls to many POTS phones. But for me these are worth testing only. I have trouble finding people's SIP addresses, if they have any, and then trouble making the clients call each other. I have trouble with firewalls and private addresses. There is no encryption. There is only a limited basic service. SIP sucks big time, and needs to be forgotten, before a million engineers waste their career on it. Come up with something better instead, if you want to have an open source competitor for Skype. Jabber, Jing, whatever, maybe they can have a try.
Anssi Porttikivi / app@iki.fi
Why go with SIP or Skype when Jingle integrates more sanely into existing infrastructure?
Help us build a better map!
You might get a few people who don't know that Bonjour is simply multicast DNS used to provide chat on your local network.
GCHQ Quantum Insert installed. If only our tongues were made of glass, how much more careful we would be when we speak
The one reason why Skype sucks is this:
Poor user control over Internet usage.
I have two internet connections and I would like to use Skype on
just one of them while keeping the other one as my default route.
Even when I configure Skype to use a proxy, which should direct it's
traffic to just one ISP, I still get Skype traffic all over my other
ISP link (the default) which is really bad because that link is
reserved for other traffic.
The proxy setting simply works as if the setting was "a recommendation".
To my query to Skype their support has not
even bothered to reply for months and months now.
This is so arrogant of Skype I'm taking now every opportunity
to tell people what they are really installing when they get Skype.
But hey.. what do you want.. the same people who wrote Kazaa, write
skype - I guess it's normal practice for them to "know better than
the user"...
SIP is not a well designed protocol - it's a head-on collision of techniques from SMTP and HTTP, often with several equivalent ways of expressing the same thing. And the protocol authors put very few size limits on things. SIP has "buffer overflow attack" written on it in large red letters.
I use several Asterisk systems in my VOIP net, I use IAX, not SIP, to hook 'em together and punch through firewalls and NATs.
As for voice quality - SIP is a call setup protocol. The voice or video is carried on an entirely different protocol, RTP (and, its often unimplemented partner, RTCP, which has nothing to do with TCP)
No matter what protocol is used to carry the voice, delay (and jitter) will cause there to be a delay. Skype has no magic bean to evade the march of time. The software in the receiving phone has to put things back together into an audible stream. Some receiving software is not even worthy of the word "crap". Some receiving software is very good, receiving engines even have good algorithms for paching over the gaps left by missing packets.
To the utter disdain of companies like Comcast and many telecom people in China, no one has truly figured out how to discern that there is actually skype traffic on their network much less block it -- this goes for everyone from a nosey IT admin, the govt, or an unfriendly ISP. On the otherhand, Comcast has put the kabosh on SIP from Vonage because it is easy to identify. Skype should continue to improve on the efficiency and stealth of their protocol and improve the features of their client. Sadly, I think with eBay now owning Skype they may have lost that innovative spark.
Install Skype but delete if from the startup list. I use AdAware's TeaTimer app which forces the apps to ask permission to alter system settings, so I just had to say no. Skype installs, but will only be grabbing resources after I actually use it, and then only if I don't kill the app after I'm done. Actually not using Skype at the moment, using Google Talk and Gizmo. Learning to realy love my Nokia-770
Letter To Iran
Happens with entirely too many things. Google is about the only one that I like. iPod, MySpace, PowerPoint, Tivo...
So yes, people sell "Skype headsets" or "gTalk headsets", and we don't notice when there start being "Xbox Live headsets" which actually use a different plug. "PowerPoint" remotes. "iPod" cables. MySpace as a noun -- not "My MySpace page", but "MySpace: <url>".
Language these days is really getting abused. I don't know how recent it is, but it certainly feels Orwellian. Keep oldspeak alive!
Don't thank God, thank a doctor!
XMPP and GoogleTalk
Supernodes are simply used to establish a connection between two computers so they can open a conversation. At a guess, this involves the IP addresses, Skype ports, and possibly usernames. None of that is particularly private info, though if you REALLY don't want people to randomly develop the ability to know what users 'you' (your username) called, you may be SOL. (If this really bothers you, why are you on the Internet at all? It's not exactly private...) The encryption is end-to-end; FWIK supernodes cannot compromise the encryption and listen in. Since supernodes change all the time, it's highly doubtful that any given SN can obtain enough info on you and those you call to make very much of it. I don't know about you, but if I Skype 10x in a week, that's a lot.
Bandwidth usage is extremely low. I've read that Skype's codec will work with as little as 8kbps/channel, though the quality loss is noticable then. I have personally used it on a shared (multiple computers all in use) 56k line with no problems, quality comparable to a decent cell phone. You might not be able to run as a supernode with 56k - I don't know what the bandwidth needs for that are - but it desn't matter, because there are literally millions of other Skype users, and plenty of them are supernodes. The client works FINE if you are never a supernode, which is good... people behind firewall, perhaps anybody behind NAT, cannot be supernodes.
There's no place I could be, since I've found Serenity...
I hit their forum with some well-crafted questions and one kind soul finally had mercy (and the decency) to tell me that the North American Gateway was having difficulties and was giving Skype Out (the paid PC to phone service) a bad case of hiccups. I had but to wait and my service would work. (Had the company been honest with me I would have been spared hours of pointless geeking around.) I waited a couple of weeks, meanwhile paying the Norwegian telephone monopoly their pound of flesh for the few calls I needed to make, and tried again. Still the same old interference. I was not happy with the Norwegian telephone monopoly's long-distance rate card, so I downloaded Gizmo Project. It installed very easily and configured itself. The sound quality is a hair lower than Skype's at its best. But really they are on a par. But I find that Gizmo is extremely reliable. I would get dropped calls with Skype. Not so with gizmo. Note: I tried Skype out a few months later and still had the same screeching interference. It was worse for my recipients. But I could hear it.
In all of my reading about Skype --- I was all over their documentation trying to solve my problem --- I never tumbled to the fact that they were using my system in a distributed way. Not that I particularly mind that sort of thing if I am made aware of it. I am happy to share bandwidth as part of a peer-to-peer community. I mean, I'm as happy to seed a Linux distribution as anyone. I am sure that somewhere Skype lets it be known that it uses your resources. And perhaps they are right up front about it and I just missed it.
Okay, this has been noted, but it's worth saying again. Even torrent client software lets you choose your level of participation. If you want to be a dirty leech (and you can live with yourself) then you can configure the software accordingly and be a dirty leech. But Skype not only doesn't tell you, they also don't give you control.
But here's the kicker. Recently I downloaded a new version of Skype at the behest of an old friend who had installed it on his Mac in California and ordered a headset. We hadn't talked in years, although we had been in e-mail communication, but I was able to have a palaver with him Skype to Skype. There were no technical problems and the sound quality was excellent. Now, as I said, I learn on Slashdot that Skype is sucking bandwidth and performance. I was considering trying again with Skype Out since their service worked Skype to Skype so well. But now I think I'll stick with Gizmo Project.
Skype has been running as a TSR since I installed it. And I have noticed some performance degradation recently. I suspected it might have to do with Skype -- especially since I get a "Range error" message that is apparently due to my dual monitor setup. I generally don't like TSRs. Now I know that Skype is Skyping me as well as the telcos and it's buggy. It would have been polite to tell me. I HATE rude software. Time to hit MS config and remove the booger from the starting lineup. You're on the bench Skype. I will call you up when I need you. (FYI I went around the block with the "range error" message. Some guy in Japan with a kludgey patch. His site was down.
"No fear. No envy. No meanness." Liam Clancy
"Q: Does Zfone work with Skype?
A: No. Skype uses a closed proprietary protocol, which they do not publish. That makes it hard to make Zfone work with it. Skype does not interoperate with the rest of the VoIP industry, which is built on open standards. I decided to follow the industry standards."
Given the likelihood of government eavesdropping (without a warrant of course) on VOIP calls, this is worth considering.
I would prefer an open standards solution but the trouble is that SIP and H323 are really crap protocols.
1) You have to open hundreds of special ports. Which makes forwarding this stuff a nightmare and means you have to open all this crap in your firwall.
2) Each company uses different ports. Ports don't seem to be a part of the standard.
3) The source and destination address are in the data not in the headers (how hard is it to use standard IP source and destination addresses?) Which makes any system with NAT a real pain when it should just be transparent, it should just work.
At least skype works with almost any firewall and if you have a firewall, almost any firewall skype will never make your system a supernode. That wasn't that hard was it?
Some group should create an open standards base protocol for sound and video that works properly.
Some of the issues about skype payments are going to be a problem with any company say gizom or wengo.
Reading this thread, I thought I'd post some very pointed numbers, which are real, relevant and probably unknown to some of you.
Skype TCP
Where our client are using Skype and are "clients" of conversations - 2.9k of bandwidth is consumed.
1.1 incoming/2.8 outgoing, total of 32 connections
Where our client are "servers" of conversations
10 active users, 39.7k incoming/63.9k outgoing. Total of 705 connections.
Without going into boring "gory" details, we see an average of 750 active connections per user using Skype-p2p udp. This is less than 160k overall in/out from a traffic perspective. We can sit there and watch the active connections zip to several thousand connections per user (not as a Skype hub-to-hub, or supernode).
Skype is JUST like is predecessor, viral in nature, aggressive in its connections. It would be preferable if Skype would limit the number of active connections to tone down its aggressiveness lest they go the way of Kazaa. As an ISP, we are considering banning hub-to-hub (we can do this centrally) and limiting the bandwidth and connections of users for Skype, as we get tired of answering their support calls only to see their problem is their PC can't handle some awful number of connections and surf the Internet too. Before we start out on this, we need to think it through from a procedural standpoint, because we know some folks "pay" for the service. In my mind it's OK to limit to "viral" effect to a reasonable limit, as it does wreak havoc on individual connections, and does hamper the ISP's ability to deliver a functional connection to the end user.
fun read:6 /bh-eu-06-biondi/bh-eu-06-biondi-up.pdf
http://www.blackhat.com/presentations/bh-europe-0
If you need text styles to communicate then you don't have a message.
SIP, in my opinion, is already better than Skype. First it is free in the sense that any company can start offering SIP. Large companies can even
offer it two its own empolyees. Second, because of Skype, prices dropped a lot during the last year. Now I can have a free landline number and free registration. The phone calls are either free for a monthly flat rate fee, or as cheap as skype. Since I moved to SIP from (www.sipate.de), I saved a lot of money. Enough to pay for the new equipment and more. I now think about buying a wifi phone, because we get free wifi in town.
This is great, because there will be no difference between mobile and landline phones anymore. And calls are basically free already today.
So why should I use Skype? With Skype I pay for the number and calls are only free for US citizens (or skype to skype). With SIP I have free calls
in 15 european countries and of course sip to sip. And I can use multiple SIP providers on a single phone, so I'm not locked in by one particular
SIP company, although I like sipgate so far (they still seem like a startup company to me, not like the big telecoms).
So, if you have internet over cable or can have nacked DSL, go for SIP!