Replace Skype with Discord for Better Audio Quality and Web-Based Interviews
macobserver.com
macobserver.com
It's crypto-audited[2], open-sourced[3], has native mobile clients[4], Swiss based[5], and pretty well known[6].
[1] https://wire.com
[2] https://techcrunch.com/2017/02/10/messaging-app-wire-now-has...
[3] https://medium.com/@wireapp/wire-server-code-now-100-open-so...
[4] https://github.com/wireapp
Compared to Signal / WhatsApp gives me e2e encryption and I can run it also as a Desktop app on Linux (so I can still message my family/kids without having a phone switched on)
[1] https://support.wire.com/hc/en-us/articles/202856854-What-is...
They're straightforward to do manually. All parties participate in a recorded "1-2-3-clap" at the beginning and end of the call, and the claps are later used for sync.
There are also many "assisted" solutions, including Cast[2], Cleanfeed[3], Ringr[4], and Zencastr[5]. Some folks find them useful, but the additional dependency also means more things that can go wrong. Anybody using them has learned (or will learn) that they must record the live/VoIP portion of the call as well, as backup.
[1] Wikipedia calls this a "phone-sync" (https://en.wikipedia.org/wiki/Phone-sync), which is not a term I've ever heard people actually use.
[2] https://tryca.st/ [3] http://cleanfeed.net/ [4] https://www.ringr.com/ [5] https://zencastr.com/
As this would replace Skype (or whatever conferencing tool you're using), it's not really an additional dependency, is it?
It is in the sense that those solutions are not only replacing the realtime part of the call, but additionally must do separate source-quality call recording and post-call file transfer sessions for each participant.
I've lost calls (or would have, if I hadn't recorded the realtime part) with more than one of those solutions because of unrecoverable errors on participants' sides. I'm reluctant to name them since (1) they in turn depend on browsers and their WebRTC implementations, and (2) in my experience different podcasters will swear by and swear at the same solution.
Here's how I'd order the reliability of common remote recording scenarios, from most to least reliable:
(1) Landline-to-landline
(2) Landline-to-mobile or mobile-to-mobile
(3) SkypeOut-to-landline/mobile
(4) VoIP-to-VoIP — Skype, Discord, whatever.
(5) Manual multi-ender — Can be done very reliably if remote participants are reasonably comfortable with recording audio. Still requires a POTS, mobile, or VoIP call for the realtime part.
(6) Assisted multi-ender solutions — Generally way less reliable than manual multi-enders. In my experience and from talking with other podcasters, expect 1/20 calls to fail.
What doesn't, and what I'd pay nontrivial (but not vMix Call) money for, is a tool that separated out individual audio and video streams into separate NDI streams for video mixing and composition.
Occasionally I deal with clients who insist on Skype and it will consistently give me headaches... I can't wait till it dies!
[2] https://riot.im/
Additionally, the desktop web app for Riot is far and away better than WhatsApp Web IMO, which is horribly non-performant, bringing my web browser to a crawl and steadily throttling up my CPU over time. I consider it almost unusable.
As far as battery usage on Android, I do not share your experience. I am seeing 8% of total usage on a phone primarily used for texting and reading emails and news. It's less than the 9% from Google Play services, and slightly more than a newsreader app at 6%. Unfortunately I can't offer a comparison to WhatsApp.
I’m actively looking for a replacement for the company.
I haven’t used it, but came across it in https://news.ycombinator.com/item?id=15890816. I haven’t used Discord either.
I’m interested in actually informed opinions on the matter!
The main problem with them is that they put more of an onus on the interviewee, which often isn't an option with high-value guests.
There are limits to what external tools can do for video; at some point you either need to be a video mixer (and this you are not doing in a web browser, unless you want people like me to ask "hey, how do I configure interpolation on my HDMI intakes?" and you go "uhhhhhh" because you have no APIs for that), or you need to plug into a video mixer.
I mentioned upthread, but the best option for these cases that I can see is ripping each video and audio feed out of a VoIP solution and sending them to a video mixer via NDI. I'd pay decent money for that.
Cleanfeed isn't actually a multi-ender, it was originally developed for live radio interviews. So it aims to give the best quality live and in that regard it's different to, eg. Zencastr which does the recording each side and syncs up the better quality audio later. Each method comes with its own pros and cons; my own experience is also that anything after the interview is not always practical.
Does anyone know something that does that
I and many gamers have been quite happy with it.
SIP is successful in the telecom space because it still requires specialized knowledge and dedicated teams of people to run it, and preserving that might be the reason it’s still such a pain.
It has options for multiple stream transports, a proper SIP gateway can even allow two totally incompatible clients to talk to each other by relaying and converting the calls. There's no reason half of your call can't be webrtc to someone's browser (asterisk SIP gateway supports this out of the box) or even POTS in an emergency.
Your asterisk solution could include a webpage and webrtc, which would actually be an easier client setup than Skype. No app, just hit this URL and start talking.
I refuse to use Skype, discord, or any other proprietary black box system for something so important.
Either use something that asks nothing of your guest, such as appear.in, or host your own service where at least you control the data. Jitsi or Lync might be palatable to your IT department.
From what I understand before they rolled it out to everyone the system worked perfectly so it's just an issue of them scaling their backend.
No idea why they don't just use WebRTC for video&audio though.
What kind of codec are you using? Are you using Amazon's streaming service or did you roll your own?