Skype under Fedora-10
Rick Sewill
rsewill at cableone.net
Mon Jun 8 23:49:40 UTC 2009
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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