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Re: Skype under Fedora-10



On Mon, 2009-06-08 at 16:16 -0500, Jud Craft wrote:
> Besides the fact that Skype is closed source, I was under the
> impression that the idea of distributed voice transmission was an
> advantage.
> 
> Shouldn't a ideal communication system have no single point of failure
> (or a country club of them, ala a group of ISP servers?)


I am curious.  What are the points of failure when using SIP or Skype?

For SIP, if I wish to have a voice call between two parties,
each party must be on the Internet and reachable by the other party.
I would assume each party must also be able to reach the SIP server.

If there is a breakdown in the path between the two people, or between a
person and the SIP server, won't the Internet attempt to route IP
traffic through other paths?  

If either party or the SIP server can't reach the Internet, there would
be a breakdown.

I have not looked closely at the SIP protocol.

There are things I do not understand.

What exactly does the SIP server do?

I thought the SIP server acted as a telephone directory where one could
register one's presence and look-up the presence of others.  

I thought the SIP server was used to do call setup and call tear-down
between the two parties through a control channel.  I thought this
control channel could use either TCP or UDP. 

I thought the actual voice traffic went over a separate data channel,
and, if we ignore NAT and conferencing, could be between the two
parties.

I thought the control channel, with the SIP server, needs to be
maintained while the data channel exists.

I may never know much about the protocol Skype uses because it is a
proprietary protocol.  I am guessing Skype must have a server where a
party registers.  I am guessing the Skype server must do call setup and
call tear-down.  I am guessing the actual voice data is on a separate,
data channel.

What exactly do these super-nodes, when using Skype, do?  Do these
super-nodes only route the voice data -or- do these super-nodes also do
directory registration and directory look-up and call setup and call
tear-down?

Can someone more knowledgeable tell us what is correct?


On a personal note, I couldn't get friends and people in a company I
used to work for, to switch from Skype to SIP.  I tried.  They said,
"Skype just worked".  I had to use Skype, on a Windows PC, provided by
the people I used to work for.

On my personal PC, which runs Linux, I don't have Skype installed.  

I tried Ekiga on my personal PC, but had problems.  My registration with
the SIP server would time-out after one hour.  I tried changing UDP
session time-out values, in iptables, with no success.  I suspect, but
am not certain, my iptables firewall rules are acting as a NAT/Firewall.
I believe iptables keeps a UDP assocation between me and the SIP server
when I register.  I believe iptables is timing out my UDP association
with the SIP servers which, in turn, is causing my registration, with
the SIP server, to fail after one hour.  

I tried Twinkle on my personal PC, with greater success.  Twinkle sends
periodic keep-alive packets.  I am guessing these keep-alive packets
keep my iptables UDP association alive.  Whatever Twinkle is doing, my
registration with the SIP server does not timeout after one hour.

I don't run Twinkle often on my personal PC.  It does no good running a
VoIP program if you have nobody to talk with when you run that program.

I have a final question.  My cable company is pushing VoIP.  Other
companies, like Vonage, are pushing VoIP.  I thought their VoIP was SIP.
Doesn't that mean SIP can be made to work?

Sorry for my long-winded messages.  I need to learn brevity.

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