On Wed, Mar 5, 2008 at 6:53 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > On Wed, Mar 5, 2008 at 6:09 PM, Alex Balashov <[EMAIL PROTECTED]> wrote: > > bilal ghayyad wrote: > > > > > Hi all; > > > > > > I have two asterisk boxes installed in two separated > > > sites, the internet bandwidth between them is very > > > good and I am using G729 codec to communicate with > > > them and IAX. > > > > Try playing around with the adaptive / fixed jitter buffer settings for > > IAX2. Also, if it's consistently a problem found on one side, consider > > any resource consumption and/or processing issues that may be occuring > > on the sending side that is creating the problem. > > > > Is this "very good bandwidth" just good bandwidth, or also good latency, > > by which we really mean _consistent_ latency? Or is it highly bursty > > and variable? This will create jitter and break voice, even if the > > amount of potential throughput is very high. Consider satellite links > > if you need a good example. > > > > -- > > Alex Balashov > > Evariste Systems > > Web : http://www.evaristesys.com/ > > Tel : (+1) (678) 954-0670 > > Direct : (+1) (678) 954-0671 > > Mobile : (+1) (706) 338-8599 > > > > Try using SIP. Post back with results. > > Thanks, > Steve Totaro >
Sorry to reply to my own post but if NAT is an issue, consider OpenVPN and SIP. I bet your calls become crystal clear, if not, try GSM instead of G729. Actually, that might be where you want to start if changing to SIP is non-trivial in your setup. Thanks, Steve Totaro _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
