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Has anyone else noticed how pristine the audio quality of a Slack call sounds? I wonder how they optimized it. My company's voip, and Lync, and Skype all fail t

• kosinus 10y ago
  I'm not sure what codec those products use, but I wouldn't be surprised if Slack went for Opus. Seems like it's the most popular choice these days, with little competition. (Discord also uses Opus.)
  • zerocrates 10y ago
    Slack uses Opus.
• mypalmike 10y ago
  I'm pretty sure they use Pied Piper compression. *Wow, tough crowd.
• nkw 10y ago
  Probably a wide band or at least mid-band codec. One thing I learned working with VOIP is people perceive quality based on what they have been using recently, which for most people is the absolutely horribly stepped on audio of skype, cheap voip, or the cellphone network. Carriers have pretty much let their baseline audio quality go to \\\\ because no one cares anymore. (Remember the Sprint pin-drop campaign?) Now, when you give somebody a wide band high quality audio connection they are blown away.
  • heywire 10y ago
    Yeah, every once in a while when my wife calls me, we get what I'm assuming is an LTE voice call and the audio is great (AT&T HD voice, I believe). Outside of that, it is amazing how bad the call quality can be. You really notice it when listening to hold music. It is like the audio version of one of those old jpegs that looks like it has been recompressed each time the email was forwarded.
  • taurath 10y ago
    Their dropped packet handling is far far better than Skypes as well - in a dodgy wifi situation linq and skype are all but unusable while slack is crisp.
    • xeromal 10y ago
      No hate towards you at all, but describing technical things as crisp drives me crazy. Can anyone enlighten me as to why that would bother my brain? Does a certain region use this or its usage widespread?
      • hrayr 10y ago
        It's probably because you have not heard it often enough to just not think about it. If I were to describe a sound as being rectangle, it just doesn't make sense and I have to think about it and try to interpret it. The extra effort in trying to decode the meaning is just annoying. In contrast, you've probably heard the phrase sharp sound. It's equally as technically meaningless as crisp sound, but you probably get it and aren't annoyed by it.
        • xeromal 10y ago
          I think you are spot on. If they described the call as clear or sharp, I'd be fine. Crisp seems out of place even though it does convey a similar meaning.
      • taurath 10y ago
        I'm not entirely certain - I use it to describe the opposite of "dull/distorted" sound, kind of a synonym of "fresh". I really have no idea where I picked it up, other than I grew up on the west coast.
      • Spooky23 10y ago
        Not sure why it bothers you. Is English your first language? This kind of thing is often difficult for non-native speakers. Crisp is a word with a few meanings. In this context, it's a metaphor assigning physical qualities to sound. (A crisp dollar bill is smooth and without creases)
• dbg31415 10y ago
  No to mention the pristine UX of the "call failed" screens. Ugh. I love Slack, don't get me wrong, but the calls fail to connect about 50% of the time for me when I'm connected to my work VPN. Work VPN seems fast for everything else, just Slack calls crap out while using it... I don't have the same problems with Skype or Google Hangouts -- opened tickets with Slack and was told, "Don't use a VPN with our service..." Uh... cool, right, will get on changing my work policy around this. =P
  • JonoBB 10y ago
    Hah, I thought it was only me. I can never connect to slack call if I'm connected via VPN. I have to drop the VPN connection first.
    • dbg31415 10y ago
      Speak of the devil... couldn't get a single Slack voice or video call to work all day due to VPN issues... we all have to use a VPN... and it just flat out doesn't work when you connect to one. Not one call all day. So... we just used Google Hangouts.
• lobster_johnson 10y ago
  Slack's audio quality is on par with GoToMeeting, I think. Of all the different solutions I've visited (Google Hangouts, Skype, Slack, Zoom, Join.me etc.), G2M has been the most reliable and best-sounding one that I've found. It's got things like native Mac app, screen sharing (just one presenter, unfortunately), audio/video recording, telephone dial-in, calendar scheduling, and persistence (i.e. you can reuse the same meeting every day). It uses little bandwidth and behaves really well on a low-bandwidth connection. And unlike GH, we've never experienced any issues connecting or staying connected. The only thing I miss is the ability for it to pipe back a little bit of my own audio, the way I believe phones do, so that you can talk while wearing earbuds or a closed headset. (Without this, you won't hear your own voice, which I find too unnerving.)
  • Sebguer 10y ago
    You can generally do this in your computer's sound settings by specifying that you want to listen to your recording device.
    • lobster_johnson 10y ago
      Not on macOS, I believe. According to the other commenter, this is called sidetone, and requires external hardware [1]. [1] https://www.citrix.com/blogs/2016/03/22/gotomeeting-audio-hardware-for-a-busy-office/ https://www.citrix.com/blogs/2016/03/22/gotomeeting-audio-ha...
      • rolodato 10y ago
        I believe you can use Loopback [1] for this (paid). If not, JACK [2] would certainly work but it's a bit of a hassle to get everything set up. [1] https://www.rogueamoeba.com/loopback/ https://www.rogueamoeba.com/loopback/ [2] http://www.jackosx.com/ http://www.jackosx.com/
        • j_s 10y ago
          I also have not tried it myself, but it appears the LineIn tool on their freebies page is exactly what is needed: https://www.rogueamoeba.com/freebies/ https://www.rogueamoeba.com/freebies/ hear the sound coming in through a microphone Edit: It is specifically called out in the Citrix blog referenced above as not working well enough. Not sure if the Loopback software would be any different, but it does offer a free trial.
• calchris42 10y ago
  Slack audio calling is using webRTC tech. On Chrome, you can check out statistics and configuration for how they set up the call by going to chrome://webrtc-internals I checked, and it looks like they are using pretty standard webRTC settings. The audio is compressed with Opus codec. They look to be limiting the bitrate to 40 kbps. (I'm not sure, but I thought Chrome default limited the AVERAGE bitrate to 40 kbps, not the peak, so Slack might actually be imposing an even tighter limit here). Opus does a pretty nice job and is full band. A lot of older VOIP tech used compression that severely limited the frequency response. We've been working on a more conversational video conferencing service called Locus (inthelocus.com). There we do some processing on the audio stream to spatialize audio sources, and found audio quality improved up to bitrate of ~120 kbps. The compression difference was more noticeable when processing the stream vs. directly playing back.
  • abalashov 10y ago
    > A lot of older VOIP tech used compression that severely limited the frequency response. I don't think that's accurate. Most conventional VoIP (in North America) uses G.711µ, whose purpose is to provide the IP correlate to logarithmically companded PCM encoding used in a normal DS0 (digital PSTN loop carrier). DS0s are not compressed. Logarithmic steps result in some approximation, but that has more to do with the mechanics of digital sampling and quantisation than bandwidth savings. In fact, G.711µ is an uncompressed 64 Kbps codec, exactly equivalent to the synchronous data rate of a DS0 in the circuit-switched world. This is in sharp contrast to codecs like G.729, which, to simplify things, use waveform translation tables to achieve significant compression (down to about 8 Kbps in the case of G.729 specifically). Some less patent-encumbered variants do similar things. The frequency response is limited by the standard PSTN bearer channel range of ~3.1 KHz, which is for historical reasons. That was the most frequency response one could reliably squeeze out of copper analog lines, for both physical and economic reasons tied up in the history of the telephone system. Since the primary concern of the VoIP industry was—and still is—interoperation with the traditional telephone network (PSTN, or Publicly Switched Telephone Network), adopting codecs which translate readily into that world with a minimum of CPU-hungry transcoding and reframing makes sense. WebRTC, as a peer-to-peer multimedia calling mechanism intended to resemble something like Skype, is more purely a product of Internet-orientated thinking and has a rather post-PSTN mindset. WebRTC endpoints, like IP phones supporting wide-band codecs (e.g. G.722), are not constrained by the requirement to talk to the old-school telephone network. So, they can use codecs like G.722 and Opus.
• Spooky23 10y ago
  Lync is awful, it does some adaptive optimization that always seems to kill audio quality. VoIP at work (At least with Cisco) is usually some grumpy telecom guy disabling HD audio or the company skimping on telephones.
  • trome 10y ago
    HD Audio (G.722 & Opus) is essentially not supported outside internal networks sadly.
  • abalashov 10y ago
    It's not that he's grumpy. It's that almost all phone calls terminate to or originate from the traditional Publicly Switched Telephone Network, which does not support HD audio and—for long-established historical reasons unlikely to change—uses a 3.1 KHz bearer frequency response range. If they disable HD audio on pure-IP extension-to-extension/intra-company calling, that's stupid, though, and indeed could be done only by Mr. Grumpy. There is unfortunately a trend to skimp on desk phones, but I think it's out of the perception that most users don't really care or use them that much.
• petarb 10y ago
  As weird as this is, every time I've done a slack audio call, I only he's audio from my right speaker on my MacBook Pro. It's like the eliminated the whole left channel. Sane behavior on multiple computers as well.
• vortico 10y ago
  Since high quality, near-lossless 48kHz mono audio can be encoded at about 96kbps, which is much lower than most users' upload/download bandwidth, why do providers even think about going under this bitrate? For example, Skype is ~30kbps and frequently sounds overly-bandlimited and garbled, so why doesn't their algorithm just use only three times this amount? Is it an issue with latency, or just keeping down their server costs? I legitimately want to know why, I'm not ranting.