In the early days of VoIP we all were affected by things like packet loss, buffering and jitter. Most of that has gone away with the arrival of faster last mile access, faster and more powerful processors on our PCs, in our desktop and mobile devices and most of all better compression with codecs that make VoIP possible. Yet, the decline in call quality, once you go off the traditional mobile or PSTN calling approach seems to be increasing not decreasing.
If you ever wondered why more people call each other on the new apps, this is one more reason that alternative calling is occurring and why services like Slack have integrated voice and video.
For example, my calls from T-Mobile or AT&T to Verizon always have a less robust tonal quality than on-net calls to customers. Of course Verizon to Verizon sounds fine, but throw in a GoogleVoice number in the middle and the call quality really goes bad.
Calls on HD Voice from Dialpad over LTE and WiFi sound pristine, the same with Telzio, Facebook's Messenger, WhatsApp, Viber or FaceTime, as the LTE or Wi-Fi connections I run on are rock solid. But as soon as the calls move to the mobile network or call someone on the other mobile networks it all goes to crap.
VoIP to 800 numbers are even more of a crap shoot it seem. As calls are passed from one switch to another, and from one call center to another the signaling paths and the media goes from bad to worse. And, 800 numbers to international call centers seem to end up not sounding very good when called from any VoIP provider, and only marginally better from a mobile operator.
Most of this is caused by transcoding and then retranscoding. While origination and termination are usually fairly standard, as calls move through the ins and outs, of each network, and the various hops a call takes we end up with so many legs, that the audio gets compressed, expanded, delayed and delivered.
In the old days, all telephony was the same. Not today.. Standards may exist but the idea of a quality standard is far from being a reality.