'iax2 show channels'....maybe.... I have a feeling this is going to be one of those ugly ones where it's going to be a pain to troubleshoot...
Offhand - have you tested 'trunk=yes' vs 'trunk=no'? PaulH On Wed, 2007-12-12 at 17:00 +1100, Daniel Cole wrote: > Hi Paul, > > Where abouts exactly is the best place to get these figures from? > > I have been checking iax2 show netstats, which does give some figures. > These appear not to be accurate though, as when there are multiple > inter-site calls, the result for one channel of audio can show no > jitter or latency, but another will have some jitter and latency. Or > is this a weird way for the problem to show its head? > > Thanks, > > Daniel Cole (CCNA) > > > P Please consider the environment before you print this e-mail or any > attachments. > > > > > > ______________________________________________________________________ > From: Paul Hales [mailto:[EMAIL PROTECTED] > Sent: Wednesday, 12 December 2007 4:40 PM > To: Daniel Cole > Subject: RE: [asterisk-users] Call Quality Issues With 2 Trixbox's - > Router Issue? > > > > > Hmmm......wierd.... > > Are you getting an weird jitter/latency figures in the CLI? > > PaulH > > > On Wed, 2007-12-12 at 16:37 +1100, Daniel Cole wrote: > > G729 All Around. > > Daniel Cole (CCNA) > > > > P Please consider the environment before you print this e-mail or > > any attachments. > > > > > > > > > > ____________________________________________________________________ > > > > From: Paul Hales [mailto:[EMAIL PROTECTED] > > Sent: Wednesday, 12 December 2007 4:10 PM > > To: Daniel Cole > > Subject: Re: [asterisk-users] Call Quality Issues With 2 Trixbox's - > > Router Issue? > > > > > > > > > > What codec are you using? > > > > PaulH > > > > > > On Wed, 2007-12-12 at 13:00 +1100, Daniel Cole wrote: > > > Hello Everyone, > > > > > > We have recently installed a pair of Trixbox servers in for a > > > client of our. They have two locations, with one server each. The > > > servers terminate 3 standard POTS lines into a Sangoma A200D card. > > > The servers are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, > > > Dual Core Xenon Processors). We are using Trixbox 2.2, and G729 > > > all around. > > > > > > Each site has two (2) 512k/512k ADSL connections terminating into > > > a Cisco 877W router (using an additional 'dumb' modem in a > > > separate VLAN for the extra dsl connection). Using policy based > > > routing, all Voice Data goes over one DSL connection (the one that > > > terminates directly into the router), and all other traffic (e.g. > > > Web and VPN) goes out the second connection (the bridged dumb dsl > > > modem). > > > > > > We are also the ISP for this client, and as thus we have full > > > monitoring of our Layer 2 and Layer 3 networks. From our analysis, > > > it doesn't appear that there is any issue in these networks. We > > > have other customers using the VoIP service, who have not > > > complained of these issues. > > > > > > Now for the Fun part! > > > The client is complaining of issues with inter-site calls. They > > > are reporting issues with crackly and broken speech, and horrible > > > jitter (or packet loss). This presents a huge issues, because they > > > have one receptionist answering all calls for both sites. So if a > > > call comes in from the other site, it automatically an inter-site > > > call, and the quality falls out of it. If the call is then > > > transfered back to the originating site, the audio 'bounces' > > > between the two sites, which add to the call quality degradation. > > > > > > We have been monitoring the router while these incidents have been > > > reported, and it does not appear to be a bandwidth issue. The DSL > > > tail used for Voice gets to no more then 120k in each direction > > > (we have tested the links, and can pull data at 53k/s between > > > sites). CPU usage floats at around 20-25% under load. The router > > > has only shows major packet loss (that we can tell) when REALLY > > > pushing it in testing (e.g. 10+ calls between sites). > > > We have enabled the SIP jitter buffer, as well as the IAX jitter > > > buffer, which appeared to make a huge difference, but the issue is > > > still ongoing. > > > > > > These issues have also been reported with some outbound VoIP > > > calls. Internal calls, and calls directly in or out of the Sangoma > > > card are clear, with no issues reported. > > > > > > Does anyone have any thoughts on what could be causing these > > > issues? We have been racking our brains here, and have tried > > > everything that we can think of. These system is a million times > > > better then what is what when it was first installed, but it is > > > still not where it should be in terms of quality. > > > > > > Any thoughts/ideas are most welcome. > > > > > > Thank you > > > > > > > > > Daniel Cole (CCNA) > > > > > > > > > > > > > > > P Please consider the environment before you print this e-mail or > > > any attachments. > > > > > > > > > _______________________________________________ > > > --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 > _______________________________________________ > --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 _______________________________________________ --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
