----- "Ben Schorr" <b...@rolandschorr.com> wrote: > Oh, dear. So my users with "less-than-ideal" bandwidth are stuck > with > drop-outs and poor sound quality because they can't use the reduced > bandwidth codec for those calls? :-( > > They've been complaining that they often end up on a call where one > or > both parties are "cutting in and out". Unfortunately it's only this > one > remote site, with about 8 users, who connect across a VPN to the site > where the server is. We've tried increasing their bandwidth and > tweaking the QOS settings on their firewalls but so far we haven't > been > able to solve it. I was hoping that switching to a lower bandwidth > CODEC would give them the call reliability they need. > > If not then I guess I'm back to the drawing board, with increasingly > impatient users, trying to troubleshoot their call quality issues. >
You need to install the G.729a codec on your system so that it will transcode your calls from ulaw (on your PRI side) to g729 (on your SIP side). Keep in mind that G.729 is a patented codec which requires licensing. The two companies offering G.729 for Asterisk(that I know of, please correct me if there are others :-) ) are here: http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC http://www.howlertech.com/products/howlets/ I've always used the Digium G.729 and it has worked flawlessly. I've also heard good things about Howler G.729 but never used it personally. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 _______________________________________________ -- 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
