on server side no special configuration is needed.<br>To have qos on the sat link, we contact sat link operator, and I think this is the only way to do it.<br>The codec is g729. Iīm not sure about the bandwidth, I think we have about 64Kbps allocated, because we almost donīt have concurrent calls.<br>
The quality is very good, you listen everything the other part says, but delayed. From landline to sat link, delay is about 2 seconds. With 2way sat link, it goes to 4, 5 seconds. <br><br><div class="gmail_quote">On Wed, Jan 18, 2012 at 9:03 AM, Arthur Stanfield <span dir="ltr"><<a href="mailto:aj@dmcip.com">aj@dmcip.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Arlen,<br>
<br>
I'm interested in seeing what setup you settled on to get decent voice quality over the Sat link? Which codec are you using, and what is the bandwidth usage?. Are you doing just one concurrent call, Or multiple?.<br>
<br>
-<br>
Regards,<br>
AJ Stanfield<br>
<div class="im HOEnZb"><br>
<br>
----- Original Message -----<br>
From: "Arlen Nascimento" <<a href="mailto:arlen.nascimento@gmail.com">arlen.nascimento@gmail.com</a>><br>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Sent: Wednesday, 18 January, 2012 12:29:23 PM<br>
Subject: Re: [asterisk-users] Peer doesn't answer<br>
<br>
</div><div class="HOEnZb"><div class="h5">Hi guys,<br>
<br>
the problem was too many NATs on the way.<br>
Although the server had a valid ip, it was behind a nat, as soon as I<br>
set ip directly on the server, things worked fine.<br>
Also, despite the huge delay, if the link has qos, the quality is very<br>
good.<br>
<br>
<br>
<br>
On Mon, Jan 16, 2012 at 9:06 AM, Sammy Govind < <a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a> ><br>
wrote:<br>
<br>
<br>
I'm only expecting NAT issues if not the latency issues. SIP traces of<br>
any such calls will make more sense.<br>
<br>
<br>
<br>
<br>
On Mon, Jan 16, 2012 at 6:05 PM, Arlen Nascimento <<br>
<a href="mailto:arlen.nascimento@gmail.com">arlen.nascimento@gmail.com</a> > wrote:<br>
<br>
<br>
the client is aware of the adverse environment and this is the only<br>
solution for him<br>
<br>
<br>
<br>
<br>
On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda <<br>
<a href="mailto:flaviormiranda@hotmail.com">flaviormiranda@hotmail.com</a> > wrote:<br>
<br>
<br>
<br>
<br>
Unless you are doing test with SIP under adverse environmet, that is not<br>
the point, but, if you intend to have Communication, you should worry<br>
about this detail.<br>
Basic infra-estructure is the first thing to think in any new project.<br>
<br>
Good luck!<br>
<br>
Att,<br>
<br>
Flavio Roberto Miranda<br>
<a href="mailto:MSN%3Aflaviormiranda@hotmail.com">MSN:flaviormiranda@hotmail.com</a><br>
Skype: flaviormiranda<br>
<br>
<br>
<br>
<br>
</div></div><div class="im HOEnZb">Date: Mon, 16 Jan 2012 07:58:34 -0400<br>
From: <a href="mailto:arlen.nascimento@gmail.com">arlen.nascimento@gmail.com</a><br>
To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] Peer doesn't answer<br>
<br>
<br>
<br>
It is a satellite connection, so ping is about 500ms. I know it is not<br>
ok to keep a normal conversation, that is not the point.<br>
<br>
<br>
<br>
On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda <<br>
<a href="mailto:flaviormiranda@hotmail.com">flaviormiranda@hotmail.com</a> > wrote:<br>
<br>
<br>
<br>
<br>
Hi Arlen,<br>
<br>
A reasonable time to Voip calls is about 250 ms. What about the Ping<br>
test end-to-end ?<br>
<br>
Att,<br>
<br>
Flavio Roberto Miranda<br>
<a href="mailto:MSN%3Aflaviormiranda@hotmail.com">MSN:flaviormiranda@hotmail.com</a><br>
Skype: flaviormiranda<br>
<br>
<br>
<br>
<br>
</div><div class="HOEnZb"><div class="h5">Date: Sun, 15 Jan 2012 21:53:46 -0400<br>
From: <a href="mailto:arlen.nascimento@gmail.com">arlen.nascimento@gmail.com</a><br>
To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
Subject: [asterisk-users] Peer doesn't answer<br>
<br>
<br>
<br>
Hi all,<br>
<br>
i'm implementing an asterisk server that will have several peers<br>
connected by satellite links.<br>
When qualify=yes or some value (from 3000 to 50000), 'sip show peers'<br>
shows the peer as unreachable. In this case i can place calls from the<br>
phone in the satellite link, but can't call to it.<br>
When i turn off qualify, the status changes to unmonitored. In this<br>
case, I can make calls in both directions but the call is never<br>
established. The phone keeps ringing until 'ring time' expires even when<br>
I answer the call on the phone/softphone.<br>
<br>
Any thoughts?<br>
<br>
Regards,<br>
<br>
-- Arlen Nascimento<br>
<br>
<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Arlen Nascimento<br><br>