Stefan at WPF
2012-Jul-21 17:46 UTC
[asterisk-users] Less good call quality using Asterisk
Hello, I am currently using the following setup: Snom 300 - Asterisk 1.8.13.0 running on Raspberry Pi - Sipgate SIP Provider When I am using this setup, the call quality isn't as good as when using a direct connection like Snom 300 - Sipgate SIP Provider to my SIP Provider (Sipgate). Sipgate supports> > - G.729 - ~ 12 kbit/s > - G.711 - ~ 100 kbit/s > - iLBC - ~ 15 kbit/s > - GSM - 13 - 20 kbit/s > - G.726 - 16 - 40 kbit/s > > According to my Asterisk server and sip show channels the used codec is0x08 (G.711), so the best one possible. Still the quality isn't as good as on a direct connection between my Snom phone and Asterisk. Using Asterisk it feels a little bit like there is some more background noise or so. Any hints why the quality using Asterisk is less good than on a direct connection? Does Asterisk by default change anything on the signal or is it just passed through? Thanks :-) Best regards Stefan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120721/6229c752/attachment.htm>
Stefan at WPF wrote:> Snom 300 - Asterisk 1.8.13.0 running on Raspberry PiNot that I can help, but I'm sorta shocked that you have Asterisk running on a Raspberry Pi! Very cool! Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
Stefan at WPF
2012-Jul-25 11:42 UTC
[asterisk-users] Less good call quality using Asterisk
Hmm is it possible, that the monitor command changes the quality? If not I guess I also once have to try compiling it from source, though I wanted to avoid that. 2012/7/23 Bakko <asannucci at gmail.com>> Hello, > > I tried Asterisk Confbridge with raspberry pi without audio issue. > > Asterisk was compiled from sources. > > http://www.voztovoice.org/?q=**node/553<http://www.voztovoice.org/?q=node/553> > > Regards > > > -- > ______________________________**______________________________**_________ > -- 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<http://lists.digium.com/mailman/listinfo/asterisk-users> >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120725/09c1fbc1/attachment-0001.htm>