skype

Yes I think it has great potential in terms of PC to PC and PC to PSTN because of its use of P2P. Regrettably the company behind it seem immature coming from the Kazaa stable and have recently started a furore on their bulletin board by increasing their gateway to PSTN charges without notice but conveniently introducing new Terms and Conditions again without notice to their customers.

Being P2P the PC to PC quality is excellent indeed better then PSTN but give it a go and try it for yourself. There are other similar products out there on the horizon e.g. Peerio444 and iTalk2U but Skype has stolen a head start for the meantime.

No, its an propriatary protocoll and they are the only choise of provider!

/Mats

Other limitations of Skype are:

- Skype is only available in 'softphone' format; dedicated interface hardware is not available

- To operate Skype, the host computer needs to be powered up constantly

- Interfacing/interworking with non-Skype VoIP services is not possible

- Single voice codec (iLBC) supported; no high quality (e.g. G.711) codec

- PSTN break out only ('Skype Out'); no PSTN break-in ('Skype-In')

- Proprietary code, no third party development allowed; no 'Open Source' Skype applications

- No support!

If you have a requirement for any of these, go and find yourself a SIP VoIP solution

Actually, they have Skype for Pocket PC and have announced plans to sell it into dedicated devices and other handhelds.

Actually, they clearly have a SIP gateway, it's how they do their PSTN termination. They don't let it call SIP addresses because they (sadly, but correctly) ask, "How many people can you call with a SIP URL?" At most a few tens of thousands, compared to 400,000 on Skype and a billion on PSTN. No wonder they put SIP gatewaying low on their list.

You've never used Skype, have you? The one thing people notice about it first is how it uses GIPS high-frequency codecs, the voice quality blows G711 and other low frequency codecs out of the water. I think it only uses iLBC if you have a very poor connection.

This is what's interesting. This is true, and we feel it's bad, yet they are whupping SIP's ass in terms of people knowingly using it.

Good - we agree then - Skype have SOFTPHONE ONLY, if you want a VoIP solution that interfaces to a 'proper' telephone or some other existing telephony hardware, Skype really is not the solution to choose.

Skype has a SIP gateway? I think not!

And please check your figures - Free World Dialup is over 250,000 and Vonage passed 155,000 on 17 May 2004 - and that is only two of the many SIP ITSPs available worldwide!

I have been using Skype since it was first released. Skype claim over

19 million downloads of their client - I note that over 20 of these downloads have been by me!

FYI - GIPS (Global IP Sound) produce one implementation of the iLBC codec ( See

)

Please also note that iLBC uses a sampling rate of 8 kHz, resulting in an aggregated bit rate after overheads of around 13-16 kbps. Whilst iLBC is indeed a fine codec (it is used extensively by the Open Source SIP Developers community), it can not be compared with a 'lossless' codec such as G.711a/u. Also note that SIP allows even higher quality 'hi-fi' codecs to be employed if required - with SIP you are not tied to one single codec!

'Whupping' is not a term I would have used here - one thing that Skype are VERY good at is HYPE - and I guess that the entire VoIP community are grateful to Niklas Zennstrom for raising public awareness on VoIP, yet it does not make Skype a better product! And there are limits to the size of community that the Skype architecture can support - Skype creaks a bit in its current form - what will it be like if it doubles or even triples in size?

In the interests of maintaining a balanced debate here, may I direct readers to a well written (and unbiased) article at The Register

Ultimately I see voice telephony going the same way as e-mail; with services becoming so cheap that they are virtually free. Whilst today, it is easy to sign up for a 'free' Hotmail account, who is business would want to run a respectable business through such an account? Similarly, whilst there will be a range of 'free' VoIP services (both SIP (e.g. FWD) and Skype), I guess that most users will prefer to pay a little more for a supported service with more functionality....

Is this codec available as either open source or open specs so that an open source compatible version could be written?

As an aside, anyone know why ogg can't be used for the voip data? Is it a delay issue?

-wolfgang

The GIPS codecs are proprietary. The ILBC codec is an open standard and GIPS sells a royalty free version of it, I don't know if there is a totally free one.

Ogg is not a voice compression algorithm, it's a data file format. You probably mean Vorbis or Speex. Speex is designed for the compression of voice, and indeed there are Speex based codecs but it's still not very far along. At least not so far enough along that the Skype guys must have felt they had reasons to pay GIPS for their codecs rather than use the free Speex ones.

I've been encouraging people to try the Speex ones.

There are some open standards high frequence codecs (G.722) and the Grandstream firmware now supports them but I have yet to have a call using them yet.

I believe that GIPS high frequency codecs also include various tricks for dealing with lost packets and congestion. One would need to design a good Speex based codec to do that.

Have at it!

I am not sure what a "proper telephone" is and why a handheld device would not be usuable as one.

But actually, this is where a lot of SIP folks have it dead wrong. They think, "Let's make VoIP as much like the regular phone as we can! Let's demand it work on a 'proper' telephone."

I understand perfectly why they say this. It is the classic mistake in dealing with a potentially disruptive technology, try to make it evolutionary instead of revolutionary.

"Almost as good as a regular phone but at least it's harder to configure" is a failure-destined strategy for VoIP.

Yup, or an H.323 one. It's just not available directly. In order to use iBasis to terminate in the PSTN, which is what I hear they are using, you would need to use SIP or H.323. So when you use SkypeOut, at some point the call is being converted to SIP or 323.

I have checked my figures. FWD has that many registered users, but only a few thousand are on at any given time. Vonage has reached many customers, but you can't dial them with an open SIP URL. (FWD has a gateway into Vonage to make SIP calls, so one hopes at some point things will oepn up a bit more.)

Skype has many millions of downloaders, and seems to have 400,000 actively on at any given time. It's orders of magnitude bigger than all the pure SIP internetworks, and it go there in under a year, compared to years for the SIP groups.

Ok, so why do you think Skype has poor quality voice codecs? Have you only used it over a dialup link? Tried g.711 over dialup?

Again, you seem to have not really explored Skype in spite of the times you have downloaded it.. All the Skype calls I have done have not used ILBC, they use the GIPS high-frequency codecs, which 16khz of frequency, and thus near-FM quality.

My understanding from talking to their folks is the reverse, that it should keep scaling fine.

I have done a lot of SIP development, I made big bets on SIP, and I am a big fan of it. But I know when somebody comes in and does it better, and I admit it.

Skype is not hype. It is better, a lot better. It's a wakeup call for the SIP and H.323 communities. Somebody doing it entirely proprietary can do it better than you and pass you while you are sleeping.

This is not just me. The people who designed a lot of the SIP protocols, whom I have spoken to most of personaly, will privately admit that the SIP world has dropped balls here. That all these things are in the SIP protocol but so rarely correctly implemented that you can't use them. Seamless NAT traversal, encryption, high frequency codecs, presence, easy configuration. It's all there, but it never gets used. Just boring G.711 calls to a small number of people. That's the reality of SIP deployment after years of effort.

Skype wakes you up because it delivers easy install, encryption every time, high quality codecs, near-flawless NAT traversal, presence, chat. All the time (except on dial-up where good luck doing a SIP call) and with no effort. To a much larger body of people.

One would think, but then why has Skype got more people in under a year using it than SIP or H.323. (I mean knowingly using it. There are large numbers of people using SIP or 323 based VOIP without knowing it, using it as PSTNoIP.) If SIP is to be more than PoIP, it's already lost to Skype, and the ball is in SIP's court.

I've been using PC to PC voice services for years. My favorite was firetalk, never equaled since. These days i use one called

. It works fine and has millions of users.

When PC to PC voice first came along i had high hopes of persuading friends and family to adopt it with no success. I use the service to join group discussions of various topics.

The VOIP services are a completely different animal. They connect to the traditional phone system and they work on standalone boxes that connect to traditional phone equipment like cordless phones. The PC to PC services do none of this. Skype is a chat service, not a phone service.

Take a look

They have hardware design for Skype.

Skype's gateway Skype's phone

"John" schrieb im Newsbeitrag news: snipped-for-privacy@posting.google.com...

They don't. They have an interface hardware for your PC - and I guess, that you need to run the skype-software on it to use this hardware...

Tobias

Not true, i have seen articles from at least one hardware vendor that is including it.

Thats their business model today, it doesn;t mean that they won't change that in future.

G.711 is not "high quality" at all, its a narrow band, uncompressed and a basic codec. All implementations i have come across use this to provide 'low quality' audio i.e. 3.4 KHz which is the same as you'll find in mobile phones. Skype, SIP and H.323 are all capable of and do use better codecs. G.711 will go the way of the dianosors, even professional VoIP systems (such as Cisco) that use this quality use compressed versions such as G.729. Alot of H.323 systems in the video sphere use G.722 as a minimum and more over G.722.1 or better.

In fact the most sucessful VoIP system in use today is based on H.323 and is used by carriers to route millions of billed minutes of voice traffic across international links for reasons of efficiency. [1] [2]

The voice over ip market is anything but settled at present, in the medium term i believe that sucessful solutions will be those that interoperate with all major protocols. I think the same thing is going to happen with IM too. Skype (imho) will be sucessful in the short to medium term because it is 'free' in terms of setup and basic use, just like email and IM.

-=-peas-=-

[1] GIANT Service Providers Carried over 1 Billion H.323 minutes to date.

a.. China Netcom Corporation (CNC) b.. China Unicom c.. Genuity d.. iBasis e.. IDT f.. ITXC g.. JiTong Communications h.. Ntera i.. PhoneOpia j.. Telephone Organization of Thailand (TOT) [2]

This one?

And it doesn't mean they won't change it again later. Your balls are in their hands.

There are other free solutions. I think Skype distinguishes itself by providing easily-used (initially, at least) software and outstanding marketing. It is the Microsoft of VoIP.

--kyler

Working great (although I only use it in the other direction, controlling my DECT bases from my desktop without running 20m of USB cabling across the house.

So what? It's not like leasing a Siemens System with outomatic contrat extension by 5 years (or German Telecom with their 10-years-are-sooo-cheap contracts you can only get out by bancruptcy) as soon as you need another phone on someones desktop. You're free to leave if you don't like it.

Not their marketing - it's really awful. But it was the first solution I could run out of the box which was working even without doing anything to our layered firewall (of course - this means the firewall isn't really good enough to keep something dangerous contained inside either... *grmbl*!) to someone with a similar firewall. It survives asymmetric routing across three links with dynamic IP addresses, moving through a load balancing HTTPS proxy and a number of further indignities. Whatever you might think of the rest of their technology, its mechanisms for setting up and keeping connections is the most advanced stuff on the market today.

Noses.

Have you encountered any system that chokes because of HTTP(S) proxies?

Are you saying that it's something other than basic NAT hole-punching?

--kyler