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
