[asterisk-users] Peer doesn't answer

[Arlen Nascimento]

on server side no special configuration is needed.
To have qos on the sat link, we contact sat link operator, and I think this
is the only way to do it.
The codec is g729. I´m not sure about the bandwidth, I think we have about
64Kbps allocated, because we almost don´t have concurrent calls.
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.

On Wed, Jan 18, 2012 at 9:03 AM, Arthur Stanfield <aj at dmcip.com> wrote:

> Hi Arlen,
>
> 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?.
>
> -
> Regards,
> AJ Stanfield
>
>
> ----- Original Message -----
> From: "Arlen Nascimento" <arlen.nascimento at gmail.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users at lists.digium.com>
> Sent: Wednesday, 18 January, 2012 12:29:23 PM
> Subject: Re: [asterisk-users] Peer doesn't answer
>
> Hi guys,
>
> the problem was too many NATs on the way.
> Although the server had a valid ip, it was behind a nat, as soon as I
> set ip directly on the server, things worked fine.
> Also, despite the huge delay, if the link has qos, the quality is very
> good.
>
>
>
> On Mon, Jan 16, 2012 at 9:06 AM, Sammy Govind < govoiper at gmail.com >
> wrote:
>
>
> I'm only expecting NAT issues if not the latency issues. SIP traces of
> any such calls will make more sense.
>
>
>
>
> On Mon, Jan 16, 2012 at 6:05 PM, Arlen Nascimento <
> arlen.nascimento at gmail.com > wrote:
>
>
> the client is aware of the adverse environment and this is the only
> solution for him
>
>
>
>
> On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda <
> flaviormiranda at hotmail.com > wrote:
>
>
>
>
> Unless you are doing test with SIP under adverse environmet, that is not
> the point, but, if you intend to have Communication, you should worry
> about this detail.
> Basic infra-estructure is the first thing to think in any new project.
>
> Good luck!
>
> Att,
>
> Flavio Roberto Miranda
> MSN:flaviormiranda at hotmail.com
> Skype: flaviormiranda
>
>
>
>
> Date: Mon, 16 Jan 2012 07:58:34 -0400
> From: arlen.nascimento at gmail.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Peer doesn't answer
>
>
>
> It is a satellite connection, so ping is about 500ms. I know it is not
> ok to keep a normal conversation, that is not the point.
>
>
>
> On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda <
> flaviormiranda at hotmail.com > wrote:
>
>
>
>
> Hi Arlen,
>
> A reasonable time to Voip calls is about 250 ms. What about the Ping
> test end-to-end ?
>
> Att,
>
> Flavio Roberto Miranda
> MSN:flaviormiranda at hotmail.com
> Skype: flaviormiranda
>
>
>
>
> Date: Sun, 15 Jan 2012 21:53:46 -0400
> From: arlen.nascimento at gmail.com
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] Peer doesn't answer
>
>
>
> Hi all,
>
> i'm implementing an asterisk server that will have several peers
> connected by satellite links.
> When qualify=yes or some value (from 3000 to 50000), 'sip show peers'
> shows the peer as unreachable. In this case i can place calls from the
> phone in the satellite link, but can't call to it.
> When i turn off qualify, the status changes to unmonitored. In this
> case, I can make calls in both directions but the call is never
> established. The phone keeps ringing until 'ring time' expires even when
> I answer the call on the phone/softphone.
>
> Any thoughts?
>
> Regards,
>
> -- Arlen Nascimento
>
>
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> -- Arlen Nascimento
>
>
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> -- Arlen Nascimento
>
>
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>
>
> -- Arlen Nascimento
>
>
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-- 
Arlen Nascimento
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