Why do you use Ulaw as codec ? Try another codec ( g729 is by far the best but isn't free ). The overhead + the 64 kbps in each direction big if you try a conference call. With 3 members, the bandwith is near 300 Kpbs / second
The QOS is handled by which kind of router ? Cisco have a netflow feature that can detect sip traffic and make a priority but don't forget that the QOS is only tuned for outbound traffic, never inbound.... Alain -----Message d'origine----- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Adrian Marsh Envoyé : Saturday, April 21, 2007 7:06 PM À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [asterisk-users] How can I improve call quality? Thanks Tim, I'd turned it on when I was at a site that had bad internet access... I'll try turning it off for a while, but I thought it was supposed to help.. Thanks, Adrian -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: 20 April 2007 19:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How can I improve call quality? Try turning the jitterbuffer off, I found that often the endpoints can do better on their own. On 20 Apr 2007, at 19:01, Adrian Marsh wrote: > Hi All, > > I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used > for PSTN calls via IAX2. > Our 'net link is a dedicated 2Mb fibre connection (of which we have > ever > used 50% max bandwidth). We've no E1/T1 links, everything is IP > based. > > My boss complains that many of the calls he holds with others has a > bad > quality. He also says that its not just him. > > My iax.conf file has: > > disallow=all > allow=ulaw > allow=alaw > bandwidth=high > jitterbuffer=yes > dropcount=2 > maxjitterbuffer=1000 > maxjitterinterps=10 > resyncthreshold=1000 > maxexcessbuffer=80 > minexcessbuffer=10 > jittershrinkrate=1 > tos=lowdelay > autokill=yes > > He complains of broken audio, muffled audio, and says compared to > Skype > its very poor, particularly during conference calls (zaptel meetme). > Most of these would be SIP based within our server though, rather than > IAX/PSTN based (X-lite/SJphone). > > > Obviously I can't do much about the far end IP connections/Mobiles > etc, > but what can I do to tweak/improve the call quality on the A*k box > itself? > > The CPU stays at a constant 10% usage, mainly due to a few other > monitoring apps on there (with these turned off, its < 2%, but > still the > same issues). > > > Also - are there any useful stats/logs that I can examine to "see" the > quality of calls? > > Thanks, > > Adrian > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton www.mexuar.net www.westhawk.co.uk/ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users