Jitter Buffer being enabled on the ConfBridge user profile was a big issue. Once we turned that off things were noticeably better. That allowed us to reach 300 ConfBridges 600 calls without a problem.
We reached 400 ConfBridges with 800 calls and only periodic hiccups in the audio for a second or two then it is happy. We also have been able to increase our capacity some by increasing the AWS instance CPUs and that was a big help. Looking further into the resources, I noticed something that I think indicates where my audio problems are coming from. On the Asterisk machine performing the 400 ConfBridges (combining audio), htop is showing periodic spikes (90%) of a single CPU core happening. It's not the same CPU core each time, but it is always in the upper half of the CPU cores. All the other cpu cores seem to stay around 10%, maybe 20%. Far less frequently, we may see a second CPU core jump to 80%. Anyone have an idea why a single CPU core would spike while the others remain low? Have a great day! Dan -----Original Message----- From: asterisk-users <[email protected]> On Behalf Of Dan Cropp Sent: Wednesday, August 21, 2019 12:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Subject: Re: [asterisk-users] Amazon AWS question Thanks Doug. We are running Asterisk 16.3.0 so I think we're on a pretty good version for the timing. We have Asterisk running on ESXi here and it's running at several customer sites in various VM environments. Ironically, none of them have the latency sensitivity set to high. This is something I didn't realize may be needed until encountering the issue during load testing this week. I would guess our largest at present has around 100 simultaneous calls and we haven't run into problems. At 200 calls, the Amazon AWS VM has 0 issues with audio. I believe it even was good at 250. It was when the tester went to 300 calls that I am told audio is often choppy. I tried performing the noload on res_timing_timerfd.so and the person testing indicated it did not help. I forgot to mention, this is on an Ubuntu 16 OS with the latest packages. Our second Asterisk box receives the call and bridges them with another endpoint (allowing us to hear it on various phones) The debugging on this box isn't indicating any issues (at least from what I can tell). Obviously a lot is going on (basically 600 calls). Neither box is indicating jitter issues. Dan -----Original Message----- From: asterisk-users <[email protected]> On Behalf Of Doug Lytle Sent: Wednesday, August 21, 2019 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Subject: Re: [asterisk-users] Amazon AWS question Dan, I don't run Asterisk on AWS, but I do on ESXi. Are you running a version of Asterisk before 13? Newer versions Asterisk handle timing better that don't require a hardware timing source. I'm running Asterisk 13 on a small 60 phone system without issues under ESXi 6.0 Doug -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
