G729 All Around. Daniel Cole (CCNA) Technical Support [http://www.hugonet.com.au/clients/hugonet.gif] Ph: 1800 424 683 Fax: 03 5221 7659 e: [EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]> w: hugonet.com.au<http://www.hugonet.com.au/>
----------------------------------------------------------------------------------- The information transmitted is the property of HugoNet and is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the company. Any review, retransmission, dissemination and other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. 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
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