All of a sudden (as of this morning), I’ve been getting random choppy audio on all of my calls.
I have FreePBX running on a Hyper-V VM and have been for years without problems. Now all of a sudden the audio on calls is very choppy.
I have both a T1 and Centrex lines running through a Vega 100G and a Vega 60G respectively. Calls through both are problematic, both inbound and outbound. I can’t seem to duplicate this on internal calls (extension to extension) so I’m thinking it’s network based.
Nothing on my network has changed and I can’t see any traffic issues on any of my switches. An RTP debug shows (what I think is) good data. Packets are alternating between to and from without any series of consistent to’s or from’s.
I have rebooted my 100G and 60G and my VM all to no avail. Calls seem to start fine and almost immediately deteriorate.
I’m at a loss for other things to try.
Any thoughts/ideas/suggestions would be greatly appreciated.
Firstly, as RTP uses UDP, our of order arrival is permitted.
It’s causes are likely to be for reasons related to implementation details of the network, but typically they would be the result of the network trying to ensure they get delivered even when data was being lost, or the network configuration was changing to cope with the failure, or overloading of individual links.
The same sorts of things can result in duplicate packets, and large jitters, which you should also look for.
Unless you have a wide choice of suppler, there may be little you can do about it. The other tradeoff factors are likely to be resilience and cost.
IP doesn’t provide a reliable service, and UDP does not add reliability. Generally for VoIP, you want minimum delays, but the TCP/IP services that provide reliability, such as TCP, achieve that at the cost of potential delays of many seconds.
Thanks for the reply. I’m not convinced it’s an RTP problem as nothing on the network has changed and the packets seem to be fine during a debug, though I’m not entirely sure how to rule that out, honestly.
If you are using public networks, the changes in internet usage brought on by Covid could well be a factor.
The other thing to look for is CPU overloads, or thrashing. If you are using a virtual machine, these could be an issue if you are not able to dedicate enough host CPU and memory to the VoIP guest, and, in the case of CPU, guarantee that it is available at very short notice (preferably << 20ms.
I guess I wouldn’t expect it to be COVID-related when it only started yesterday morning. It’s also affecting my Centrex lines and the only network related to it is the internal connection between FreePBX and my Vega60.
My VM settings haven’t changed and it’s the only VM on the host with full access to the CPU and 8GB RAM.
I’m sort of leaning toward database performance issues (see my previous post about ibdata1), but I don’t seem to be having problems with internal calls.
Called from my cell into both my PRI and Centrex lines and heard choppy audio, though my operator said I sounded fine.
Called from my home phone which is a remote extension (meaning it connects back to the PBX over the internet) and had no audio issues.
So my thoughts are that it’s not a network issue as the remote extension call works just fine. Possibly an issue with both my Vega100 and my Vega60 but both at exactly the same time? That’s far too coincidental.
Just to be precise, T1/PRI lines are not exactly POTS (plain old telephone system) which historically refer to Analog pairs to the phone on your wall,
They are however considered PSTN (public switched telephone network) devices which divide the 1.544Mb signal from the telco into 24 64k ‘channels’ in the Time Domain (multiplex) (ISDN efficiently reserves one for signalling, T1’s used what we call ‘robbed bit signalling’ but recede to 56K, to notify hangup, off hook, etc events)
As such TDM circuits are not vulnerable to ‘choppy audio’ it either works or it doesn’t. but the CPE will indicate an alarm as to what is broke.
Took my two Vegas and the PBX and put them on their own switch with an uplink to the PoE switch that powers my phones and things are better but not perfect.
There’s a voice prompt that plays when you call in on the T1 and it’s usually choppy but the call is fine once it reaches an operator. It’s like it’s starting poorly but ultimately working fine.
Just more fuel to add to the fire, I guess. Frustrating that I can’t figure this out.
Just as an addition: the recorded calls sound just fine. There’s no choppy audio in any of the calls I’ve listened to, despite them being choppy while on the phone.
Yeah I saw that thread immediately after I made my post. Disabling DPMA and restarting had no effect and I’m still seeing the little chat bubbles on my phones.
I’m not too concerned with it. It’s just one more thing acting goofy with my server that I can’t fix.
