I setuped a FreePBX 14 instance running asterisk 13.32.I also configured a few Conference Rooms so we can have conferences since we are all confined and remote working. When one join the conference from external line, audio is fine. When they join from their Mitel 6867i SIP phone (pjsip) directly through the room’s 4 digits number, it gets really choppy. All the other calls from these remote workers are perfectly fine so I don’t think it’s a bandwith problem on their side. My system is running in ESXi 6.7 and I assigned 2 vCPUs and 4gb of RAM. We never really get more than 8 participants in the rooms altogether. It really seems like there are some kind of transcoding when using the conf. room with the SIP Phones that plague the audio quality. I read stuff from years ago that conferencing was too intensive for virtualization but it was like 5 or 6 years ago…
While I doubt that this is the problem, your version of Asterisk and FreePBX are at least this old, so …
I’d start by looking at the SIP DEBUG for the Mitel phone and see what codec it’s using. If you are running a full conference on a virtual server and through transcoding in there as well, it could be overwhelming the conference.
On the other hand, since it’s only that one extension and (I infer) that the extension is local, so it should be something related to that single extension.
If all else fails, you might want to update your system. Conferencing as suddenly gotten a lot of attention and there are several “corona” fixes to the module in the past month. Upgrading may not solve the problem; in fact, it might cause other problems, but your installation is pretty old, so when you get some time, you should consider updating the system in place.
It is with all remote workers extensions, not a single one. I installed that system like a year ago it is not 5 or 6 years old but I understand I should still upgrade it. I set all extensions to use G.711 but I don’t know about the conf. room itself and/or the phones. I’ll have to through endpoint manager profiles to know more. The thing is that it only happens in the conf. rooms, all other calls from remote workers or CSR working at the office are perfectly fine and we spend days on the phone these times. I’ll do a SIP debug from the Mitel phones and another test with softphones. It happens even with only 2 participants if they are local extensions. I noted CPU load can go quite high (a tad under 2 spikes) but I don’t know if adding cores would do it as the code needs to be multithreaded to take advantage of core addition.
Just so I understand, are these local extensions, or are they remote extensions?
All extensions are extensions, whether they are local or remote, so if it’s only happening with remote extensions (extensions outside your local LAN), the problem description is different than if they are local (inside the LAN) extensions.
I’m teaching a class in this right now. Your assertion is pedantically incomplete. Each process in the system is made up of one or more threads, so the code doesn’t need to be multi-threaded to take advantage of multiple cores - it need to be multi-process, which means that each extension (which is a distinct subordinate process/thread) can run on its own core. Adding core to solve this problems, especially if you are seeing processor spikes, could alleviate the problem. Also, remember that adding conference participants increases the bandwidth used in each conference, as well as the memory utilization of adding all of those subordinate processes.
Given the new (and kind of conflicting) information, I’d say your instinct on adding more resources would be a good place to start.
They are local to the PBX but remote from the office. Local, to me, means they are not coming from our SIP Provider. Weirdly enough a conference made of one local extension but as many external users is going just fine. All calls but conferences are perfectly fine from remote workers extensions or office’s extensions. I’ll upgrade and add two cores and go from there!
Local extensions, for purposes of SIP and FreePBX, means that they’re coming from the Local Area Network. If you have your phones in the LAN and the server in the LAN, they are Local. If you have the phones anywhere outside your LAN and the server inside the LAN, they are remote. If your phones are anywhere, and you have the PBX in the cloud, they are remote.
If you want to use the terms differently, you’re going to have to figure it out.
Adding more resources to the server may or may not make any difference. Give it a try and let us know.
This is for the purposes of NAT and NAT Configuration. With any other definition, you cannot answer the pertinent questions about whether the NAT configuration is correct.