D&C GLug - Home Page

[ Date Index ] [ Thread Index ] [ <= Previous by date / thread ] [ Next by date / thread => ]

Re: [LUG] OT - SIP providers?

 

On Wed, 7 Oct 2009, tom wrote:

Rob Beard wrote:
Grant Sewell wrote:
I'm quite happy with giving them all software-phones.  Many people are
already using Skype for Skype-Skype communications, so the concept of
talking to someone through the computer isn't an alien one.

Grant.

Skype is an abuse of protocol. Its like the IE4 of the VOIP world - it stops people taking advantage of the real FREE thing, while costing you money and screwing up the internet.

While I'm not a fan of Skype for reasons mentioned - I'm not sure it's abuse... And how does it cost you money?

I guess one advantage of Skype is that it's also available for mobiles. For instance my Nokia e63 on Three has a native Symbian S60 Skype client (voice calling and text chatting only), plus I also have Fring which supports Skype (again voice calling and text only), MSN (text only), Yahoo (text only) and a couple of others. Not sure if it supports Jabber, I believe it does.

On the other hand, my phone also supports SIP (at least via wifi), I haven't tried it over the 3G connection due to not having a SIP account anywhere and well, the 3G in my area is poor.

You don't need a (separate) sip account. - you already have one if you have a SIP phone. SIP is like html or ftp - its a protocol - the web does the rest.

The web has nothing to do with Skype - The Internet on the other hand... :)

See my other post about NAT which is why SIP to SIP for phone to phone rarely works over the public Internet.

Skype degrades your service, both deliberately (to make you pay for upgrade) and by design - all Skype calls go through Skype so there's always extra hops - if I talk to the bloke next door on Skype it takes me all the way to their servers and back.

I really don't get that - How does it degrade anything? All Skype calls need to use a central server of some sort - how else can it know the IP address of the other party? Now Skype is quite clever in that rather than having one central "PBX" it has many - the so-called supernodes, and yes, some of these will relay audio data - just as sometimes it's neccessary for an external SIP proxy to relay audio data - all due to NAT issues.

If I go SIP:blokenext door I go the a router in the exchange and back again, and the service is as good as it can get.

Only it isn't because NAT will stop you. To make 2 phones work peer to peer from you to your neighbour, you'll need to have routers with working SIP ALGs (and I don't know any), and do port-forwarding. And with port-forwarding, you can only 'dial' one phone per IP address unless you change the default ports you use.

Ever tried to dial an IP address from a phone?

If you start with Skype you will probably be stuck with them forever as it will never be 'convenient ' to change - not a good move for you but more money in the bank for Skype -which was built on free software to start with!

Skype made it easy - they worked hard to overcome NAT and firewall issues (OK, it was the p2p forerunner that did that!), but...

Gordon

--
The Mailing List for the Devon & Cornwall LUG
http://mailman.dclug.org.uk/listinfo/list
FAQ: http://www.dcglug.org.uk/linux_adm/list-faq.html