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Comment by scandox | Hacker News Reader

So what you're saying is: VOIP will never match PSTN? Because that's the mood music I get from the sysadmins at work too.

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Never is a long time. What if some day the technology exists to interpolate dropped packets (e.g., using ML) fast and cheaply enough to fill in the distortion? That doesn't seem too outlandish in the next 20, or even 10 years.

That sounds fine, except for the timeline. What I'd like is an affordable, ubiquitous solution next week :)

Uh, the modern codecs are already using linear prediction and correction. (essentially a rephrased more advanced perceptron) This is why SILK (Skype), Speex and Opus (SILK-derived part) can work even mildly well on bad links. They predict syllables and also long term speech characteristics. Going to full sentences or words is much harder and not likely to bring gain other than free subtitles. And would require really good voice recognition comparable to Siri at least. Good luck running that on an embedded device reliably.

Remember, the codec has to handle less than 20 ms latency. Running a sizable ANN (fast) or HMM in that time is not easy or even possible. A small one will be no good for various speakers.

Instead the codecs use various specialised error hiding tricks and adaptive strategies.

Heck, Codec2, one of the strongest voice codecs around could be used if all else fails. It is beginning to be used in SDR and amateur radio - optimised for availability and intelligibility not quality. It would be nice to have a variant integrated with CELT for higher quality.

VOIP is considerably better than PSTN if configured correctly. There's no comparison. Your sysadmins don't understand VOIP.

Their ignorance is always just slightly less than mine. Which is a sign of their intellectual "efficiency".

just want to point out: vast knowledge of telephony/VOIP is not something every sysadmin should be expected to know. If you don't have telephony expertise on staff I hesitate to suggest someone build out a VOIP infrastructure on their own. It can be extremely time consuming to learn and it is definitely painful to troubleshoot.

Shouldn't it be possible for me to get some IP phones in my office and tunnel through my router to a competent online voice provider? If so, can you recommend such a provider?

Yes, that is exactly what we do at work. We have ~20 IP phones in the office plus a few in remote locations. A few of us also use softphones (X-Lite or Zoiper) & USB headsets.

Our pbx is hosted by http://sipworxx.com/ and it works great.

http://voip.ms is pretty decent and offers a few nice features that can avoid building an on-site PBX.

I use this with good Polycom "HD voice" devices. I'm also rather picky about voice quality.

Eh, I'd argue this varies a lot. In some cases you are more likely to run into issues setting up multiple SIP clients behind a NAT trying to connect to a server out on the public internet than you are to install your own PBX on-premises.

Grandstream, Yealink, etc. all provide pretty easy to administer Asterisk-based PBX appliances that can be paired with a SIP trunk of your choice (shout out to Flowroute here, they're great).

If you're feeling more adventurous sipXecs from sipfoundry is pretty easy to configure relative to the additional complexity it brings over the more simplified alternatives. I went from having 0 real world experience with SIP to managing a small installation for my aunts shop as well as one at home in a couple days.

For proprietary offerings there's of course Skype for Business, etc. and I would argue they aren't terribly hard to configure or maintain either.

Most of the work is just in network configuration once you have multiple sites and you need to ensure QOS going over your MPLS or whatever to ensure your voice packets take priority - that can be more difficult but that's something a competent network admin can help you with.

I think that statement needs some qualification. If you're sending the VoIP traffic across networks you don't control all bets are off. VoIP across an enterprise's private network or through a leased network where SLA's can be mandated and QoS can be controlled is one thing. "Wild west" VoIP across the Internet, especially through sketchy networks, is another.

Edit: "canbe"

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