Do an RTP analysis with Wireshark of a sample call. That could probably narrow down the source of the problem. I would suspect you will either see some jitter or packets out of order.
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 > -- Andres Technical Support http://www.telesip.net _______________________________________________ --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