ap_6200 Posted January 17, 2008 Report Share Posted January 17, 2008 I have a Feature Server set up with 19 domains and a total of 390 Accounts across those domains. I am seeing a consistent day-time call volume of 15-25 concurrent calls using 2.1.4 until 2.1.5 can be further tested: Calls: 11323/2587 (CDR: 4124) 20/38 Calls SIP packet statistics: Tx: 2426674 Rx: 2326830 Emails: Successful sent: 438 Unsuccessful attempts: 122 (Warning: Last email could not be sent!) Uptime: 3 03:13:36 (77MB/2047MB 20% 43010880-0) WAV cache: 3 The media CPU usage during the day is between 25% and 50%... Ususally coinciding with the Call Objects.. Meaning at this time - the utilization would be at 38% (based on the graph) Have I reached the theorhetical maximum? Are there any other customers or forum-posters with this many domains/accounts - and concurrent calls? Complaints are coming in referring to static, audio cut outs, etc... Is 2.1.5 any better, or do I need to start moving customers off of this feature server? Please advise! Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 17, 2008 Report Share Posted January 17, 2008 Is 2.1.5 any better, or do I need to start moving customers off of this feature server? No, 2.1.5 is no difference regarding the performance. My opinion is that hardware has become so cheap today that it is not worth having customers complaining about dropouts. So I would definitevely power up another server. The new domain backup feature comes in handy. It should be relatively simple to move a few domains off to a new server! Quote Link to comment Share on other sites More sharing options...
ap_6200 Posted January 17, 2008 Author Report Share Posted January 17, 2008 No, 2.1.5 is no difference regarding the performance. My opinion is that hardware has become so cheap today that it is not worth having customers complaining about dropouts. So I would definitevely power up another server. The new domain backup feature comes in handy. It should be relatively simple to move a few domains off to a new server! So a thought could also be that the PBXnSIP software can't handle this load, as opposed to the hardware? We didn't buy cheap hardware - and have invested heavily. Other suggestions? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 18, 2008 Report Share Posted January 18, 2008 Well, there are other limitations like the number of sockets that you can have open. And the absolute maximum number of calls in 2.1 are 256 calls (2-legged). All other structures are pretty much dynamic. But that all does not explain choppy audio. The CPU load meter is what we should pay attention to. Maybe the blue line at 75 % is still too optimistic and we should lower it to like 50 %. Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted January 18, 2008 Report Share Posted January 18, 2008 So a thought could also be that the PBXnSIP software can't handle this load, as opposed to the hardware? We didn't buy cheap hardware - and have invested heavily. Other suggestions? Linux or Win? What Version? disk controllers might be heaviliy used causing interupts? Consider RAM drive from Gigabit for swap on linux or or Win. In the Win system are you familiar with perfmon or system admin, WMI etc. I can refer you to a Voip SIP call recorder, and you can with client permissions record all calls and confirm the drop outs, cracks etc exist on the NIC ports the Servers. Perhaps it's the QOS,DIF stuff on the circuit? when you said, "Invested heavily" Just what is this running on? I know 23 concurrent calls for 5 or 6 hours straight across 45 extensions for 4 weeks in a row never raised a AMD3100 over 9% utilization on a WinXP Pro box with a NIC to a PRI gateway, a NIC to the LOCAL Lan, and a NIC to the public intenret. 2.1.5.2357 (Win32) License Status: Office 75 License Duration: Permanent Additional license information: Extensions: 58/75 Accounts: 78/100 Working Directory: C:\Program Files\pbxnsip\PBX IP Addresses: 127.0.0.1 192.168.1.250 192.168.100.1 216.xxx.yyy.20 MAC Addresses: 1D04ACC519C4 1D15F29BC703 1D902713C720 Calls: 2917/486 (CDR: 6322) 0/0 Calls SIP packet statistics: Tx: 3844156 Rx: 3844269 Emails: Successful sent: 31 Unsuccessful attempts: 5591 (Warning: Last email could not be sent!) Uptime: 8 02:06:28 (51MB/511MB 54% 13842388--6078828) WAV cache: 0 Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted January 19, 2008 Report Share Posted January 19, 2008 Complaints are coming in referring to static, audio cut outs, etc... Is 2.1.5 any better, or do I need to start moving customers off of this feature server? Please advise! Can you share whats become of this situation? You can't sweep something like this under the rug with your clients. Quote Link to comment Share on other sites More sharing options...
ap_6200 Posted January 19, 2008 Author Report Share Posted January 19, 2008 We upgraded to 2.1.5 and moved some domains off of this particular FS to load balance across the server farm. This will give me a clear indication on Monday or Tuesday if indeed I have reached the upper echelon of active calls without affecting voice quality. By no means am I happy with this solution, because I should be able to do more than what i'm able to do - but the limitations are there, and nobody from PBXnSIP has been able to offer me any other suggestions. I have 6 FS's for my customer base, all built identically.. The only one I'm having trouble with - is the one I described earlier. I am not seeing any negative indication from the server itself - it is only the Media CPU Usage graphed in PBXnSIP. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 20, 2008 Report Share Posted January 20, 2008 I have 6 FS's for my customer base, all built identically.. The only one I'm having trouble with - is the one I described earlier. Hmm. We had a very strange case recently where the MTU was set to a short value which caused a lot of strange effects... It took a lot of time to find that problem, but in the end it was good to see why the "mystery" was not a mystery. Maybe it also makes sense to find out what is different on that specific server and once we find it out put it on the check list. Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted January 20, 2008 Report Share Posted January 20, 2008 By no means am I happy with this solution, I am not seeing any negative indication from the server itself - it is only the Media CPU Usage graphed in PBXnSIP. Would you expand on the questions regarding your platform. If you'd like to do that off-line I think my profile will drop me a line. Diagnosing this problem should be easy since it appears as if you can predict failure based upon load. So therefore the system should be working fine with a smaller load and we can track performance and isolate the failure to the extranet or the intranet and the place to start is to packet capture all data to the failing server during a low use (working OK) and into the (Not working OK) high use period. Using a good quality Switch that supports port mirroring capture all traffic... I also know of a low cost software package that will record and rebuild in realtime all SIP/RTP sessions and record all calls. This will give you something to go on, an until you can starting asking questions that can be answered you will be chasing ghosts... Happy to help. Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted January 23, 2008 Report Share Posted January 23, 2008 What's become of this situation? Quote Link to comment Share on other sites More sharing options...
clarity Posted January 17, 2009 Report Share Posted January 17, 2009 Don't rely on just those customer reports of audio dropping off to determine you are nearing your max CPU for PBXNSIP. Get an IP phone plugged directly into the lan or WAN of the server.. try both. During the busiest time of the day and during those types of loads test with a locally connected telephone and make calls into/out of voicemail. If it's not choppy and sounds perfect in both directions (recorded messages) I'd bet you have plenty of CPU/RAM/IO to spare. Also try PSTN calls if you have locally connected PSTN. (not distant voip). Just my immediate thoughts.. Maybe you have already done these tests. -Steve Quote Link to comment Share on other sites More sharing options...
hosted Posted January 18, 2009 Report Share Posted January 18, 2009 In our tests we were transcoding to increase CPU load and we started getting choppy audio at 50% CPU load. So this our maximum per PC. It will be nice when pbxnsip will pass RTP across processors. Quote Link to comment Share on other sites More sharing options...
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