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I miss landline telephones. For whatever reason, the audio clarity of video conferencing still has not touched the quality we used to get out of old fashioned

I miss landline telephones. For whatever reason, the audio clarity of video conferencing still has not touched the quality we used to get out of old fashioned analogue over copper.

gregmac 4y ago
Clearly false. Analog voice service is limited to 300–3,300 Hz, and being analog, subject to line noise/static/etc. About the only constant with it was that people would hold handsets up to their heads that were purpose-built, and reliably had a decent minimal level of quality. With IP-based conferencing, you get as good as the worst link. A lot of people use built-in microphones on their laptops in a noisy room, and so the audio sounds like garbage. The people with consistently high quality have gaming-style headsets (or really, any >$100 headset). There's a couple people I talk with that have (inexpensive) studio-style microphones on boom arms, and they sound phenomenal. Of course it could be you. If you're listening using garbage headphones or the built-in speakers in your laptop, it doesn't really matter how great other people's microphones are. You're clearly misremembering the quality of analog phone lines; at their worst they're better than a bad IP-based service, but at their best they are blown out of the water by people that put even a modicum of effort into their setup.
- Karrot_Kream 4y ago
  Jitter and buffer bloat are huge issues with VoIP. People that deploy audio VoIP solutions often terminate VoIP at the nearest router to the WAN they can and use DECT to serve wireless VoIP phones because of how bad WiFi can be for jitter. It's not as simple as your setup and codecs.
Karrot_Kream 4y ago
Some of it might be nostalgia, but some of it is definitely latency and jitter. Even into the 90s there was only so much latency on a phone call and most phone lines tried their hardest to _act_ circuit-switched. The average video caller is on WiFi, so there's packet buffering at the AP (which can be quite large since most people are running meh quality consumer routers at home), again at the router, then however many hops until the RTP endpoint. Jitter can be quite large. That doesn't even get into the scourge that is modern NAT and the possibility of TURN relays along the way. One quick "hack" is to do your conferencing on a box connected to your router via Ethernet (or through switches). It can help bad connection paths but it's not a panacea. Another is if you have a slow internet connection, try to get a router with QoS and apply that to prioritize your VoIP packets. Unfortunately other than cellular standards, there's no good way way to easily get a low latency, low jitter connection as most packet switching is optimized around throughput. Personally we use Zoom at work and I know the Zoom edge has low jitter to me so if anyone needs to VC me at work, they should get "as good as it gets". For my personal VC needs I ended up using Matrix and hosting a TURN server in a DC that peers directly with my ISP. Folks in my area can call me on it through WebRTC and get latency on the 3-6ms range.

I miss landline telephones. For whatever reason, the audio clarity of video conferencing still has not touched the quality we used to get out of old fashioned analogue over copper.

gregmac 4y ago

Clearly false. Analog voice service is limited to 300–3,300 Hz, and being analog, subject to line noise/static/etc. About the only constant with it was that people would hold handsets up to their heads that were purpose-built, and reliably had a decent minimal level of quality. With IP-based conferencing, you get as good as the worst link. A lot of people use built-in microphones on their laptops in a noisy room, and so the audio sounds like garbage. The people with consistently high quality have gaming-style headsets (or really, any >$100 headset). There's a couple people I talk with that have (inexpensive) studio-style microphones on boom arms, and they sound phenomenal. Of course it could be you. If you're listening using garbage headphones or the built-in speakers in your laptop, it doesn't really matter how great other people's microphones are. You're clearly misremembering the quality of analog phone lines; at their worst they're better than a bad IP-based service, but at their best they are blown out of the water by people that put even a modicum of effort into their setup.

Karrot_Kream 4y ago
Jitter and buffer bloat are huge issues with VoIP. People that deploy audio VoIP solutions often terminate VoIP at the nearest router to the WAN they can and use DECT to serve wireless VoIP phones because of how bad WiFi can be for jitter. It's not as simple as your setup and codecs.

Karrot_Kream 4y ago

Some of it might be nostalgia, but some of it is definitely latency and jitter. Even into the 90s there was only so much latency on a phone call and most phone lines tried their hardest to _act_ circuit-switched. The average video caller is on WiFi, so there's packet buffering at the AP (which can be quite large since most people are running meh quality consumer routers at home), again at the router, then however many hops until the RTP endpoint. Jitter can be quite large. That doesn't even get into the scourge that is modern NAT and the possibility of TURN relays along the way. One quick "hack" is to do your conferencing on a box connected to your router via Ethernet (or through switches). It can help bad connection paths but it's not a panacea. Another is if you have a slow internet connection, try to get a router with QoS and apply that to prioritize your VoIP packets. Unfortunately other than cellular standards, there's no good way way to easily get a low latency, low jitter connection as most packet switching is optimized around throughput. Personally we use Zoom at work and I know the Zoom edge has low jitter to me so if anyone needs to VC me at work, they should get "as good as it gets". For my personal VC needs I ended up using Matrix and hosting a TURN server in a DC that peers directly with my ISP. Folks in my area can call me on it through WebRTC and get latency on the 3-6ms range.