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."

What a bunch of crap...

[rthille]

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'.

Re:Isn't that the point?

[Greyfox]

Yah. Something kind of similar with sip (or IAX) is fwdout. [fwdout.net] You share your landline for local calls in return for use of the network.

Of course these days it's pretty rare that the person you're calling does not also have a computer, so you could make a free point to point call with sip just as easily. You just need to know the IP address of the person you're calling, or they can get a free number for internet calls from Free World Dialup. [freeworlddialup.com]

Re:Doesn't play well with others

[svanstrom]

Yeah, but Skype isn't advertized as decentralized P2P, so most people won't know that that's how it works when they install it; and most people using Skype doesn't even understand what decentralized P2P is.

IM-programs have for years _NOT_ been decentralized P2P-based, and Skype is largely competing with other IM-solutions; so it's not like people expect it to be P2P in the sense that their computers are used for other peoples filetransfers etc.

You guys are kidding, right?

[glomph]

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/ [gizmoproject.com] ) which has a very economical and functional interface to the regular phone system (POTS). Even cheaper, try http://www.voipstunt.com/ [voipstunt.com] or http://www.freecall.com/ [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/ [asterisk.org] ) which is best of all!

Skype belongs in the shitbin of history. Closed systems suck.

Thanks for the differences; there are even more

[postbigbang]

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.

Re:So, now the shoe is on the other foot?

[arose]

Sure I could open it up in my firewall but why would I?

The problem isn't opening your firewall for Skype, the problem is closing it.

Re:What a bunch of crap...

[spinkham]

Yes, skype can traverse NAT, but cannot be a supernode for others when it is behind NAT. That is what your parent post was saying.

Use Asterisk plus SIP endpoint of your choice!

[Chembal]

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 [trixbox.org] 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 [sureteq.com]. 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 [voipjet.com] for long distance over VoIP, and one for calls in from and out to the Free World Dialup [freeworlddialup.com] 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

Re:Isn't that the point?

[STFS]

Did you ask to yourself why there is no such service? Actually there is: http://openwengo.com/ [openwengo.com]

Wengos application is GPL and it uses SIP as its protocol. Behind it, there's a french phone company I believe which seems to think that this may somehow pay off for them (I don't know much more about their business model though).

Re:There's lots of bogus anti-Skype FUD

[GreyPoopon]

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.)

Uh, if the supernode mode is enough to cause performance problems with a single user over a 128kbps upstream connection, it's likely to cause some noticeable problems at a business with thousands of users running Skype -- even if the connection to the internet that they share is a very large one.

Re:Isn't that the point?

[phatmonkey]

Yes there is. [gizmoproject.com]

Re:Isn't that the point?

[Rodness]

I've done a lot of research on Skype lately. If you're running as a supernode, it only uses about 6k of bandwidth up and down, it doesn't monopolize your network connection.

And if you want to make sure you'll never be a supernode, just make sure you're behind a firewall/NAT. The criteria for being a supernode are fast processor, available memory, and fast UNFIREWALLED connection. It's very easy to make sure you're not a supernode by using a firewall. It's really not that big a deal, and certainly not worth being up in arms against.

If you want to be pissed off about Skype, there's bigger things to worry about. Among those:

Closed protocol
RC4 obfuscation layer so that we can't see what's being transmitted
Binary with anti-debugging techniques so we can't see what it's doing
And so forth...

The trouble is that SIP is a crap protocol.

[kholburn]

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.