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
Hi> Also - are there any useful stats/logs that I can examine to "see" the > quality of calls? >You didn't mention that you have any QOS on your router, so we can basically guarantee that your problem is the internet connection. Remember that all the research on networking has been how to saturate a single connection and download as fast as possible, so when some spod hits a website and reads a web page then he grabs basically the whole connection for a short space of time. During that time your voip packets tend to loose out and get delayed - the jitter buffer does some stuff to try and compensate, but ultimately it will loose Add some kind of priorisation to the T1 line and your quality should go up dramatically Check first using something like testmyvoip.com to get an idea of your situation (stress the internet by opening up lots of simultaneous downloads during the test) Cheap fix is to get a separate DSL line and run the voice over that... Ed W
> Our 'net link is a dedicated 2Mb fibre connection (of which we have ever > used 50% max bandwidth).Remember in computer terms this means that you used 100% of the connection, 50% of the time.... Your voice will loose out against the big data packets and spoil the voice quality big time Ed W
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-usersTim Panton www.mexuar.net www.westhawk.co.uk/
On Friday 20 April 2007 20:01, 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. >Iam using Gradwell in UK with SIP with several lines and I do not have these problems. Try SIP and if the problem is gone check your IAX configuration.
> Check first using something like testmyvoip.com to get an idea of your > situation (stress the internet by opening up lots of simultaneous > downloads during the test)Repeat: Try the above before you do anything else... Ed W
Jan Gerryt du Toit
2007-Apr-26 06:11 UTC
[asterisk-users] Re: How can I improve call quality?
We had similar QoS problems as described but only within meetme rooms. As you know the meetme app needs zaptel to be installed, this is because the meetme app needs a timing device. Now, if one does not have any cards installed on the box, then one needs to load the ztdummy for the meetme app to work. As its name suggests this is a zaptel dummy and it provides a software timing device. It turned out that this software timer is not that great and caused all the QoS problems. Our problems were solved by plugging in a physical card into the box. This way the meetme app uses the hardware timing device on the card itself. Our setup is still 100% SIP - the card is only used for timing purposes. Hope this helps.