Fwd: RE: [Asterisk-Users] SIP / IAX over satellite

[Uriel Carrasquilla]

Ku is very sensitive to rain fade or big cities with a lot of polution.  I
have always used C band.
8 MHz (by 2 to make it duplex)would be taken very fast by SIP unless a well
balance CODEC is used (such as g729).  Also make sure that there is no
double compression going on.
Regards,
Uriel

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Olaf Menzel
Sent: Sunday, October 12, 2003 6:02 AM
To: asterisk-users at lists.digium.com; esa.momosat at horz.de;
esa.momosat at horz.de; karl.jonas at ieee.org
Subject: Re: Fwd: RE: [Asterisk-Users] SIP / IAX over satellite


Hi all,
--------
thank U all for your very fast response.  I want to clarify that my
question was not regarding about the fasibility of
Voip over satellite in general but especial the behavior of the Asterisk
PBX on a long delay path. We just successfully tested
H323 Voip with a Innovaphone IP 2000 over satellite to a Innovaphone
PSTN Gayteway. It works fine if you be aware about
the > 500 ms RTT and you are ready to train youself a little bit in
conversation discipline.

BTW. We are operating a DVB hub station over SESAT 36 E on transponder
G6, Ku band. SESAT "hangs " directly over Sibirea and
covers with his wide beam all of Europe and most of Asia and the north
part of Africa. We have 8 MHz space segment and using 64 - 384 kBits
FDMA DAMA in the return channel.

The reason for the registration timeouts can probably in a very short
programmed "registration timer".  Maybe somebody of you is familiar
with the Asterisk source code and knows which of the "wheels" I have to
turn. It would help me very much. (:-)  I would prefer to use SIP
rather than IAX over satellite because I am more familar with it.

To diagnose this behaviour I will test the satellite link with the SIP
test tool SipSak (http://sipsak.berlios.de) and later with a network
link simulator to increase the propagation delay step by step until the
SIP registration times out. First of all I will test a Snom-2-Snom
direct connection without any SIP registrator or proxy. I this works the
debugging can go on.


BTW. If somebody of you knows how to configure the setup below in
iax.conf and extensions.conf I would be very happy,
because it would spare a lot of time for me with trials and errors. (Sip
configuration is clear)

Snom-Office ------SIP----- *-Office ------ IAX --- SAT ----IAX
----*-Field--- SIP--- Snom-Filed

many regards

Olaf



<http://dict.leo.org/?p=lURE.&search=feasibility>

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