Ok this is my first post and I will try to keep it short. I have searched everywhere and haven't found an answer to my question I have two Trixbox servers that are connected over the Internet via an IAX2 connection. We are experiencing very poor sound quality. I have tried many different codecs gsm, ilbc, g729, g711 and all seem to have the same problem. (All though g729 seems to work the best but still isn't reliable) The problems are intermittent sometimes the sound will cut out for 3-4 seconds and other times the sound will just be loosing every other word, and other times it sounds just fine. Also, we have been using Skype over this same Internet connection and have very good sound quality with very few lost words. So here are my questions. First, is it a correct assumption to say that because Skype works well over this connection then I should be able to get asterisk to work over this connect. I am hoping that Skype isn't "better" then asterisk in this area. If I should be able to get the same sound quality could you point me in the right direction on how to achieve this. (I have tried messing with the jitterbuffer but haven't been able to find very good docs on how to utilize this functionality so about all I have done is set jitterbuffer=yes) Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070428/9e26dec1/attachment.htm
Yossi Ben Hagai
2007-Apr-28 01:41 UTC
[asterisk-users] Two Connected Servers Sound Quailty
Hi Matt, you didn't mention what type/bw of each site Internet connection, i suggest that you try to split the scenario into smaller pieces: - run long term pings between the server while you make a call and check for packet loss. - make internal calls between extensions on the same branch and verify that both servers work okay (eliminating Internet connectivity) - register a UA from one site to a server on the other site, make a call and viceversa (eliminating a problem on one of the servers). - check for speed/duplex setting on NIC and switch port. - check if the sound quality issues are symmetric (does both sides experience the sound cut or it only happens on a specific site). - make sure you don't use G.711 as it consumes bw and from the codec list you've mentioned has the lowest tolerance to packet loss. Since the problems are intermittent my bet is that someone in the office is have the p2p client work overtime or sending lots emails with funny attachments On 4/28/07, Matt Gardner <garddawg@gmail.com> wrote:> > Ok this is my first post and I will try to keep it short. > > I have searched everywhere and haven't found an answer to my question > > I have two Trixbox servers that are connected over the Internet via an > IAX2 connection. We are experiencing very poor sound quality. I have tried > many different codecs gsm, ilbc, g729, g711 and all seem to have the same > problem. (All though g729 seems to work the best but still isn't reliable) > The problems are intermittent sometimes the sound will cut out for 3-4 > seconds and other times the sound will just be loosing every other word, and > other times it sounds just fine. > > Also, we have been using Skype over this same Internet connection and have > very good sound quality with very few lost words. > > So here are my questions. > > First, is it a correct assumption to say that because Skype works well > over this connection then I should be able to get asterisk to work over this > connect. I am hoping that Skype isn't "better" then asterisk in this area. > > If I should be able to get the same sound quality could you point me in > the right direction on how to achieve this. (I have tried messing with the > jitterbuffer but haven't been able to find very good docs on how to utilize > this functionality so about all I have done is set jitterbuffer=yes) > > Thanks in advance. > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com <http://easynews.com/>-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070428/9e4e53d9/attachment.htm
Try SIP if at all possible. I have had mixed results with IAX that SIP made go away. If you try SIP, you can at least rule out IAX as the cause. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Yossi Ben Hagai Sent: Saturday, April 28, 2007 4:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Two Connected Servers Sound Quailty Hi Matt, you didn't mention what type/bw of each site Internet connection, i suggest that you try to split the scenario into smaller pieces: - run long term pings between the server while you make a call and check for packet loss. - make internal calls between extensions on the same branch and verify that both servers work okay (eliminating Internet connectivity) - register a UA from one site to a server on the other site, make a call and viceversa (eliminating a problem on one of the servers). - check for speed/duplex setting on NIC and switch port. - check if the sound quality issues are symmetric (does both sides experience the sound cut or it only happens on a specific site). - make sure you don't use G.711 as it consumes bw and from the codec list you've mentioned has the lowest tolerance to packet loss. Since the problems are intermittent my bet is that someone in the office is have the p2p client work overtime or sending lots emails with funny attachments On 4/28/07, Matt Gardner <garddawg@gmail.com> wrote: Ok this is my first post and I will try to keep it short. I have searched everywhere and haven't found an answer to my question I have two Trixbox servers that are connected over the Internet via an IAX2 connection. We are experiencing very poor sound quality. I have tried many different codecs gsm, ilbc, g729, g711 and all seem to have the same problem. (All though g729 seems to work the best but still isn't reliable) The problems are intermittent sometimes the sound will cut out for 3-4 seconds and other times the sound will just be loosing every other word, and other times it sounds just fine. Also, we have been using Skype over this same Internet connection and have very good sound quality with very few lost words. So here are my questions. First, is it a correct assumption to say that because Skype works well over this connection then I should be able to get asterisk to work over this connect. I am hoping that Skype isn't "better" then asterisk in this area. If I should be able to get the same sound quality could you point me in the right direction on how to achieve this. (I have tried messing with the jitterbuffer but haven't been able to find very good docs on how to utilize this functionality so about all I have done is set jitterbuffer=yes) Thanks in advance. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com <http://easynews.com/> -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070428/cf4d5b24/attachment.htm
Hi Matt -> I have two Trixbox servers that are connected over the Internet via an IAX2 > connection. We are experiencing very poor sound quality. I have tried many > different codecs gsm, ilbc, g729, g711 and all seem to have the same > problem. (All though g729 seems to work the best but still isn't reliable) > The problems are intermittent sometimes the sound will cut out for 3-4 > seconds and other times the sound will just be loosing every other word, and > other times it sounds just fine. > > First, is it a correct assumption to say that because Skype works well over > this connection then I should be able to get asterisk to work over this > connect. I am hoping that Skype isn't "better" then asterisk in this area.Yes, it's certainly possible to get good quality with asterisk. Skype is not better, they just build in more default latency. I've never measured exactly, but it seems that Skype calls typically have a built in buffer between 250ms and 1000ms. Asterisk, will try to use as little latency as possible. You've set jitterbuffer=yes, but you'll also need to set maxjitterbuffer (probably to 1000), resyncthreshold (probably to 1000), and maxjitterinterps (10 is a good safe value). You can try adjusting these values to see how it affects your calls. You'll also want to do something about QoS. If you don't, the next time you try to FTP a file, it will try to grab all your available bandwidth, whether or not you are on a call. This will surely screw up your call quality. If your routers have QoS options, you can ensure that your voice traffic will get first dibs on bandwidth. You'll need to configure QoS on both ends. - Noah
One thing I would suggest trying, just from experience, Is the load on the boxes. Unless you have REALLY poor latency, calls do not cut out for just 3-4, but they very well could if the box load is getting very high. Keep a look at top (though not reliable) and the call count when the breakups start happening. Give it a shot :] -bkruse ----- Original Message ----- From: "Steve Totaro" <stotaro@asteriskhelpdesk.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Saturday, April 28, 2007 4:51:52 AM (GMT-0800) America/Tijuana Subject: RE: [asterisk-users] Two Connected Servers Sound Quailty _______________________________________________ --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
On 28 Apr 2007, at 09:05, Matt Gardner wrote:> Ok this is my first post and I will try to keep it short. > > I have searched everywhere and haven't found an answer to my question > > I have two Trixbox servers that are connected over the Internet via > an IAX2 connection. We are experiencing very poor sound quality. > I have tried many different codecs gsm, ilbc, g729, g711 and all > seem to have the same problem. (All though g729 seems to work the > best but still isn't reliable) The problems are intermittent > sometimes the sound will cut out for 3-4 seconds and other times > the sound will just be loosing every other word, and other times it > sounds just fine. > > Also, we have been using Skype over this same Internet connection > and have very good sound quality with very few lost words. > > So here are my questions. > > First, is it a correct assumption to say that because Skype works > well over this connection then I should be able to get asterisk to > work over this connect. I am hoping that Skype isn't "better" then > asterisk in this area. > > If I should be able to get the same sound quality could you point > me in the right direction on how to achieve this. (I have tried > messing with the jitterbuffer but haven't been able to find very > good docs on how to utilize this functionality so about all I have > done is set jitterbuffer=yes)Try making a call, and then use iax2 show netstats (I think that is the syntax in 1.4, I'm off-line at the moment and my memory is going :-( ) This will give you some statistics collected by the jitterbuffer code. Post them here and we will take a look. As a reference point, we get _perfect_ g711 calls over IAX from the UK to new york, so it is possible. Theoretically Skype _can_ produce better audio quality than asterisk as it supports a wideband codec but in practice asterisk/iax should be just as good and somewhat more predictable as Skype's routing varies from call to call. Tim Panton www.mexuar.net www.westhawk.co.uk/
On Sun, Apr 29, 2007 at 11:55:43AM +0100, Tim Panton wrote: TP> Theoretically Skype _can_ produce better audio quality than asterisk TP> as it supports a wideband codec Asterisk 1.4 support G.722 transit, and trunk support G.722 transcoding. -- JID: ds@im.seiros.ru ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/
Hi all, I have the same problem using SIP with G729 and it's just on one direction. But ... there is bandwidth management on the FW equipment (sonicwall) and others clients (we are a IP centrex) works find using the same server. A idea ? Thomas ________________________________ De : asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] De la part de Matt Gardner Envoy? : samedi, 28. avril 2007 10:06 ? : asterisk-users@lists.digium.com Objet : [asterisk-users] Two Connected Servers Sound Quailty Ok this is my first post and I will try to keep it short. I have searched everywhere and haven't found an answer to my question I have two Trixbox servers that are connected over the Internet via an IAX2 connection. We are experiencing very poor sound quality. I have tried many different codecs gsm, ilbc, g729, g711 and all seem to have the same problem. (All though g729 seems to work the best but still isn't reliable) The problems are intermittent sometimes the sound will cut out for 3-4 seconds and other times the sound will just be loosing every other word, and other times it sounds just fine. Also, we have been using Skype over this same Internet connection and have very good sound quality with very few lost words. So here are my questions. First, is it a correct assumption to say that because Skype works well over this connection then I should be able to get asterisk to work over this connect. I am hoping that Skype isn't "better" then asterisk in this area. If I should be able to get the same sound quality could you point me in the right direction on how to achieve this. (I have tried messing with the jitterbuffer but haven't been able to find very good docs on how to utilize this functionality so about all I have done is set jitterbuffer=yes) Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070501/cc4f0fa2/attachment.htm