[Asterisk-Dev] A crazy idea... Skype channel in Asterisk

Benjamin on Asterisk Mailing Lists benjk.on.asterisk.ml at gmail.com
Tue Oct 19 09:06:59 MST 2004


On Tue, 19 Oct 2004 15:38:05 +0100, Chris Lee
<cslee-list at cybericom.co.uk> wrote:

> Ok, all fair comments, though Hype may be a substantial part of why
> skype is doing well it is also not true to say it is not ease of use.

I didn't say it was not ease of use. What I said was that there are
others which are also ease of use, so my point was that Skype is *not
the only one* that has ease of use.

But it may come down to the following ...

Of all the PC-VoIP stuff that is hyped a lot, Skype is the easiest to
use. Other ease of use solutions are either not hyped (ie Firefly) or
they are not available on Windoze (ie iChat).

... which basically supports my call for making more effort lobbying for IAX.

> I first was told about Skype by a friend of mine in South Africa, the
> two of us gave up on other products because between S.A. and the UK most
> VoIP products (IMs and the like) just did not cut it.

Similar story with Apple's iChat. It traverses NAT very well (unless
you are behind a second level of NAT), it's very intuitive and easy to
use and the sound quality is very good. I am having a hard time
getting some Mac heads to give Asterisk a try because they are happy
with iChat as it is. It's a difficult sale. There is no point arguing
that iChat uses a non-standard derivative of SIP, that won't work with
anything else, because unless this limitation is felt as a limitation
by the end user, they simply won't care.

> IAX is also very good but it does drop the call far more often than skype

I cannot agree with that. The only dropped calls I have are a result
of Zaptel devices detecting false hangups, which has nothing to do
with IAX. However, I remember to have experienced dropped calls with
IAX softphones, so I guess the softphones aren't all that stable yet.
Maybe some of the Firefly and iaxComm users can comment on this.

However, I can assure you that I have won a few "IAX versus Skype --
who gets to connect to his peers?" contest in internet cafes around
the globe. I run Asterisk on my Powerbook and use it as a SIP/IAX
gateway, either for X-Lite on the same Powerbook or for a SIP phone I
may carry with me. There are very very few places where I have been
unable to hook up, but fellow travellers with their Skyped Windoze
notebooks have not as often been able to connect as I have with
Asterisk and IAX. Likewise, in my experience IAX calls are stable
almost to the point of what I would describe as defying gravity.
 

> 2. I set up an IAX client, I now have to find out how to receive calls
> through the NAT that is hiding me away, I can not get the NAT
> re-configured (If I even know what NAT is), so I am stuck

Are you sure you are talking about IAX? It sounds more like you are
talking about SIP. IAX is a NAT friendly protocol that doesn't have
those kind of issues.

> so what I was suggesting was a peer-to-peer fabric that can help with
> the membership, naming and call receipt issues so that IAX could pick up
> some of the ease of use provided by the supernodes.

But now you are mixing protocol and directory service. Yes, Skype
provides a directory. IAX is only a protocol. However, there are IAX
directories and they are free. All you need to do is sign up with such
a directory, for example IAXTEL (IAX only directory run by Digium) or
FWD (both SIP and IAX run by Jeff Pulver), also Freshtel (run by
Virbiage, the folks who brought you Firefly).

Unlike Skype, those directories are peering with many other
directories and services. For example, if you are hooked up to FWD,
then users of many VoIP services in the US such as Vonage can call you
directly over the internet on your FWD number. There are public access
numbers in some countries where people can call you from the PSTN.
Further, you can call toll-free numbers in a number of countries free
of charge via FWD.

I'd definitely say that FWD offers far more than Skype and if you
connect to it via IAX you don't have to deal with any NAT traversal
issues.

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

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