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 <[email protected]> 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" <[email protected]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" < > [email protected]> > 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 < [email protected] > > 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 < > [email protected] > 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 < > [email protected] > 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:[email protected] > Skype: flaviormiranda > > > > > Date: Mon, 16 Jan 2012 07:58:34 -0400 > From: [email protected] > To: [email protected] > 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 < > [email protected] > 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:[email protected] > Skype: flaviormiranda > > > > > Date: Sun, 15 Jan 2012 21:53:46 -0400 > From: [email protected] > To: [email protected] > 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 > > > -- _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE > or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- Arlen Nascimento > > > -- _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE > or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- Arlen Nascimento > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- Arlen Nascimento > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Arlen Nascimento
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
