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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."
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."
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?
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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.
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.
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.
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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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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