Any idea what is it about SIP over IAX2 that made such an improvement? -M
On Thu, Dec 2, 2010 at 6:01 AM, Steve Totaro <stot...@asteriskhelpdesk.com> wrote: > > If getting a second circuit is out of the question. > > 1. Switch to SIP > 2. Install and Learn Vyatta for QoS (Squid may help you quite a bit > as well) as your router (or whatever you prefer) I use the paid > versions of Vyatta but the free edition should be sufficient. > > I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping > times. I used GSM and some tricks on the Vyatta box. > > Originally, before I deployed the above, it was a wild west situation > like what you have now. Going from G729 to GSM made a big improvement > in conjunction with QoS. > > My theory on that is that G729 is already a very lossy codec, so any > more loss, garbled audio. GSM is less lossy. > > Switch from IAX to SIP was another huge improvement, and then finally > putting Vyatta and QoS as my router made calls almost crystal clear. > > There was the obvious lag time but users get used to that and wait a > second or two before speaking so they don't talk over each other and > the quality was five by five, except for solar flares, sandstorms, > rain. Things beyond my control. > > Thanks, > Steve T > > -- > _____________________________________________________________________ > -- 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