[Asterisk-Users] http://www.skype.com/
John Todd
jtodd at loligo.com
Wed Nov 5 03:34:36 MST 2003
>Philipp von Klitzing wrote:
>
>>>http://www.skype.com/
>>>seesm to be the latest craze... anyone have any knowledge of their
>>>technolgy use etc ??
>>
>>
>>- closed source
>>- WinXP and 2k only
>>- peer-2-peer, i.e. they route foreign calls through your client
>>(and bandwidth) if that helps the calling parties
>In one of our Swedish daily newspapers, like the national "Financial
>Times", one
>of the owners said that they're going to sell a commercial version with PSTN
>connectivity early next year.
>
>As I understand it they must not be fully peer-to-peer even if they use your
>bandwidth, there has to be media servers in their network, handling calls.
>Or?
>
>/O
My limited understanding:
If you have a public IP address (non-NAT) then you will see more
traffic going through your session than most, since NAT'ed hosts need
a "relay" on the outside of their NATs.
Skype uses the Global IP Sound codecs, which are tremendously
efficient. Voice quality is reportedly excellent, even under the
extreme examples of multi-application use dialup connections.
Skype encrypts all sessions at the management and media layers, which
is a feature that I _love_ and wish Asterisk would develop more
robustly.
Skype is indeed proprietary, and is a for-profit company, so don't
expect a chan_skype to happen soon unless they decide that they want
to play nice with others (doubtful.)
Skype will certainly be introducing PSTN connectivity, but I am very
interested in what their numbering plan will look like for inbound
calls, if such a plan is contemplated at all. These guys have to
make money, so look for any new features costing $$$ - don't get too
hooked yet (Anyone remember the problems .mp3 and .gif formats?
Helloooo?)
Skype has the ease of use and features to which we, as the rest of
the VoIP community, should aspire. Extremely easy setup, excellent
call quality, robust and distributed routing, and secure
transmissions. They are certainly lacking many of the features that
makes something "good", such as compliance with standards, but as a
private company they can ignore those issues because they're not
doing this for the betterment of anyone but themselves. If we can
implement Skype-like features in our software but still develop in
the open source, standards-compliant world, then that is a noble
goal. Skype will certainly lead the way in showing us what features
the customers want, and their system will push us towards making
"real" VoIP networks of a much larger and robust (P2P) scale, but
ultimately I think they'll fail due to their closed source methods.
JT
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