7 ms·

I will definitely try this because I have trouble with all other voice chat services including Skype, WhatsApp and Hangouts. Often one of those three will work

I will definitely try this because I have trouble with all other voice chat services including Skype, WhatsApp and Hangouts. Often one of those three will work but not always. But the most annoying problem with any internet voice chat is not so much the quality but the latency. Landline phones have noticeably lower voice quality but one can still enjoy a conversation perfectly well. High latency, on the other hand, absolutely ruins the experience. I always see these services talking about "crystal clear quality", but never latency, which is a shame. Maybe there is simply nothing they can do about it. I've noticed latency get worse and worse on the Internet since I started using it in the 90s and nobody seems to talk about it. There are so many more sources of latency on every layer on the modern internet and it's a damn shame. I remember when interleaving got enabled on my ADSL and latency to everything doubled overnight. Then there are NATs, and all kinds of filtering shit that ISPs insist on. Sigh...

boterock 10y ago
Can relate, I remember enjoying more talking with friends in TeamSpeak than with my girlfriend via skype/hangouts/facebook call... now with a discord server, talking with friends is so much easier, but quality is not as good as TS... but still convenience beats quality, and the loss is not that bad.
- vram22 10y ago
  I've been using TS recently on a consulting project, after my client suggested we use it, due to issues sometimes with Skype. Seems good so far, though not used it for many days yet. Needs a server though. The software is free for a basic version. I think it supports multiple people on the call, though we haven't tried that yet. And seems to have many features overall, not tried them out yet, but will try some over time.
pixl97 10y ago
>I always see these services talking about "crystal clear quality", but never latency, which is a shame. Maybe there is simply nothing they can do about it. I've noticed latency get worse and worse on the Internet since I started using it in the 90s and nobody seems to talk about it. Well, the first problem at home is likely your wireless. Live in an apartment and you can pick up 30 other routers in your wifi search. Chances are you're going to have latency and re-transmit issues to your AP. Second problem, your router. Most of them suck. If you're lucky and yours doesn't setup QoS/shaping rules. I use fd_codel on mine to attempt to minimize the amount of buffer bloat that occurs. In the race to make your internet 'fast' huge buffers act like buckets for data, which is really bad if you want your data instantly. And if you search buffer bloat you'll see a lot of people talk about it, very few ISPs want to do anything about it though. Lastly, the one you can't do anything about is the path that your data takes to the remote user. If it is overloaded or has capacity induced latency there isn't much you can do, other than change ISPs maybe.
- kough 10y ago
  For anyone interested, MIT's systems class has students read two different papers on buffer bloat: http://web.mit.edu/6.033/www/papers/gettys.pdf http://web.mit.edu/6.033/www/papers/gettys.pdf http://web.mit.edu/6.033/www/papers/allman.pdf http://web.mit.edu/6.033/www/papers/allman.pdf
- adamheins 10y ago
  I don't know much about the different router brands and models. Which would you consider to be some of the good ones?
  - jumpCastle 10y ago
    I'm not an expert on routers but if I was in the market I would look at the Wirecutter. I find their recommendations pretty good. thewirecutter.com/reviews/best-wi-fi-router/
  - amq 10y ago
    A starting point for me: LEDE support
    - e12e 10y ago
      That's: https://lede-project.org https://lede-project.org or:“Linux Embedded Development Environment” - the new project born from OpenWRT. I've used it with some old d-link routers, and I'm quite happy. The documentation is still not stellar, but it has gotten better, and the configuration more consistent. It's quite easy to set up ssh-access with a key, and simply get a copy of configuration files with scp - ready to set up identical configurations on new router installs.
  - corford 10y ago
    Draytek are a good choice at the consumer level.
    - Nexxxeh 10y ago
driverdan 10y ago
> I've noticed latency get worse and worse on the Internet since I started using it in the 90s and nobody seems to talk about it. You're remembering incorrectly then. Dialup latency was in the range of hundreds of ms. Early high speed internet access also had higher latency than modern connections. If you have latency problems it's likely a problem with your hardware.
- torrent-of-ions 10y ago
  I don't remember dialup latency. I only remember it having low bandwidth. I'm talking about ADSL latency in the 90s.
  - moomin 10y ago
    Latency for a 56k modem was 180ms. Latency for a voice call on the same line was _much_ smaller. AFAICT, latency from my games console television to my TV is much higher...
    - NoGravitas 10y ago
      300ms was not unusual on a 56k modem, if my memory serves.
      - moomin 10y ago
        180ms is what I got on WirePlay. The number is burnt into my brain. If the machine wasn't co-located, I can envisage it being a lot higher.
    - alphapapa 10y ago
      When I accessed the Internet over a "56k" modem connection, my network latency was less than 100 ms, so how could that be possible?
    - torrent-of-ions 10y ago
      Yes, I believe you, I meant I don't remember what the latency was.
ElijahLynn 10y ago
If you are on shitty wifi, no app will ever fix that.
vram22 10y ago
>interleaving got enabled on my ADSL What is interleaving in this context?
- detaro 10y ago
  Multiple data frames are interleaved and transmitted together. Good for error tolerance, since bursty noise doesn't wipe out a complete frame, but small parts of multiple ones, which then can individually be fixed by their Reed-Solomon error correction codes. But it adds latency, since the sender has to collect data from multiple frames to combine, and the receiver has to receive the entire block before it can be decoded, instead of transmitting each frame as it arrives.
  - vram22 10y ago
    Thanks.
peraquins 10y ago
If you're ok with the Chinese gov snooping, I find WeChat to be amazingly good at voice chat, even if your connection is bad.
jdfreefly 10y ago
Have you tried Amazon Chime? We are ridiculously aware of the latency issue in high quality voip. We do not at the moment support user generated keys the way Telegram does but we do use TLS/DTLS encryption end to end for our VOIP streams.
- CharlesW 10y ago
  > We are ridiculously aware of the latency issue in high quality voip. We do not at the moment support user generated keys the way Telegram does but we do use TLS/DTLS encryption end to end for our VOIP streams. Any plans to make this available to developers as a service via AWS?
  - jdfreefly 10y ago
    Sorry, I can't really comment on our roadmap and our priorities within it.
Animats 10y ago
The best phone voice quality is still ISDN. Digital all the way to the handset, and rigidly clocked with fixed delay. Switzerland has ISDN phones, and the voice quality is impressive.
- DaiPlusPlus 10y ago
  ISDN was 64kbps pet channel though - and to my knowledge did not use a wideband voice codec by default (or at least didn't mandate all ISDN receivers support wideband audio). Lots of people who had ISDN kit plugged their old POTS phones into their TAs anyway.
  - doctorshady 10y ago
    ISDN is actually used quite a bit in niche markets in the states. For studios, typically you'll see a couple B-channels bonded to give someone 128K MP3 or AAC. At a bare minimum, 64 kbps/16khz g.722 is used. I worked in a newsroom a few years ago where I frequently had to record a weather guy from a distant area who used this.
    - nickpsecurity 10y ago
      Used in STE units for secure communication as well: https://en.wikipedia.org/wiki/Secure_Terminal_Equipment https://en.wikipedia.org/wiki/Secure_Terminal_Equipment
  - dboreham 10y ago
    I think it sounded good because it was clean 64K end-to-end. No trickery with transcoding a bunch of times along the way so someone could save a nanopenny.
    - Animats 10y ago
      Right. The bits you send are the bits that are received, clocked at a fixed rate. No compression or timing artifacts.
- 10y ago
Florin_Andrei 10y ago
> But the most annoying problem with any internet voice chat is not so much the quality but the latency. Landline phones have noticeably lower voice quality but one can still enjoy a conversation perfectly well. High latency, on the other hand, absolutely ruins the experience. Maybe this is because traditional communications were built by people working exclusively in that field - and they eventually became true experts, with a deep understanding of the real issues. Nowadays Internet chat services are cobbled together by happy-go-lucky engineers who, prior to this, maybe did an ads firehose on Twitter, and after this project will perhaps move on to work on Facebook games or whatever. Smart people to be sure, but their understanding of the field is necessarily superficial. The fast rate of change in this industry explains many of the things that are ailing it currently.
- dmix 10y ago
  > Nowadays Internet chat services are cobbled together by happy-go-lucky engineers who, prior to this, maybe did an ads firehose on Twitter, Skype has been around for 13 years. It isn't a young startup full of ex-node.js programmers. WhatsApp also has some excellent infrastructure built on top of Erlang which has decades of experience and refinement of heavy-duty telecom usage baked into the framework. If anything the experience of previous development trickles down and younger engineers benefit from that knowledge. But I'm not sure I buy the idea that you can only build great systems if you've only been doing that one category for your whole career. The rate of jobs changes faster and in front-end development the pace of toolset changes pretty fast. But even in front-end development it's still the same languages. usually just the tools are the trendy part. And if it's switching languages every 5-7 yrs that knowledge of languages is transferable. I've only benefited as a programmer from exposure to many other languages (and projects). Regardless, the hardcore backend guys you get to build this type of infrastructure isn't the same hipster front-end devs that Twitter hired in the early years to build out a consumer product. Consumer is very different than building out telecom infrastructure in Erlang/C++.
l1ambda 10y ago
The Opus codec ( http://opus-codec.org/comparison/ http://opus-codec.org/comparison/ ) can enable very low latency audio. 26.5 ms algorithmic delay (the time it takes to go through the algorithm) by default, but can be set to a very impressive 5ms. Digital audio can't achieve the near no-latency analog POTS has, but it can get pretty darn close e.g. if Opus is used in the very low latency mode. I believe Mumble and maybe TeamSpeak let you choose the Opus latency. I know some other apps like WhatsApp, Wire, and Signal use Opus as well. I don't know what settings they use though. Another problem is cell phones add ~150ms latency (pre VoLTE) and people that route calls through Google Voice adds an additional 150ms+ latency on top. If you ever talk to someone on a cell phone on google voice, it sounds like they're calling from the moon. 150ms is about the human limit before turn-taking and talking over each other becomes an annoyance. That's why cell-to-landline conversations (150ms one way) were tolerable, but cell-to-cell (300ms one way) conversations are abysmal. And that's all in the same area code. Once you add long distance, it gets worse.
- doctorshady 10y ago
  Just fyi, POTS is digital up until the very last mile. While the network is digital, it uses time multiplexed serial PCM streams under the hood. Aside from the delay imposed by fiber transmission at the trunk level (~40 milliseconds from one side of the US to the next. If you're using a long distance network that packetizes the connection or sends a call on some weird route, that can certainly increase), the g.711 encoding delay should be fixed at 0.125 milliseconds.
  - l1ambda 10y ago
    Interesting. I didn't know g.711 was so low. Typical VOIP latency must then stem from D/A and A/D conversion and internet latency.
    - makomk 10y ago
      g.711 has such low latency because it doesn't take advantage of inter-sample redundancy in any way whatsoever - it's just raw 8-bit samples at 8 ksps with a compander on either end (traditionally on the analog side of the DAC and ADC).
doctorshady 10y ago
Unless you're hitting an IP node that transcodes to g.729 or something, landline phones should sound pretty good; the mu-law codec uses companded 8-bit PCM that roughly equates to about 14-bit linear in terms of precision. If you're sending a lot of very low frequencies (most phones and line cards have a pass filter in them to remove stuff below ~60-100 hertz, so while the network is very capable of sending it, it doesn't always in practice), you'll start to hear companding noise, but it's rarely the case that it gets loud enough to cause an issue. As for the sample rate, 8 khz is a bit low, but you can make it sound significantly better with a bit of equalization; usually turning down frequencies near 1 khz a little goes a long way. Sometimes too, it doesn't hurt to gently (and I stress, gently) reduce frequencies near ~300 hertz if it sounds too muffled.
77pt77 10y ago
Try using Skype over a 3G connection. The latency is almost palpable.

boterock 10y ago

Can relate, I remember enjoying more talking with friends in TeamSpeak than with my girlfriend via skype/hangouts/facebook call... now with a discord server, talking with friends is so much easier, but quality is not as good as TS... but still convenience beats quality, and the loss is not that bad.

vram22 10y ago
I've been using TS recently on a consulting project, after my client suggested we use it, due to issues sometimes with Skype. Seems good so far, though not used it for many days yet. Needs a server though. The software is free for a basic version. I think it supports multiple people on the call, though we haven't tried that yet. And seems to have many features overall, not tried them out yet, but will try some over time.

pixl97 10y ago

>I always see these services talking about "crystal clear quality", but never latency, which is a shame. Maybe there is simply nothing they can do about it. I've noticed latency get worse and worse on the Internet since I started using it in the 90s and nobody seems to talk about it. Well, the first problem at home is likely your wireless. Live in an apartment and you can pick up 30 other routers in your wifi search. Chances are you're going to have latency and re-transmit issues to your AP. Second problem, your router. Most of them suck. If you're lucky and yours doesn't setup QoS/shaping rules. I use fd_codel on mine to attempt to minimize the amount of buffer bloat that occurs. In the race to make your internet 'fast' huge buffers act like buckets for data, which is really bad if you want your data instantly. And if you search buffer bloat you'll see a lot of people talk about it, very few ISPs want to do anything about it though. Lastly, the one you can't do anything about is the path that your data takes to the remote user. If it is overloaded or has capacity induced latency there isn't much you can do, other than change ISPs maybe.

kough 10y ago
For anyone interested, MIT's systems class has students read two different papers on buffer bloat: http://web.mit.edu/6.033/www/papers/gettys.pdf http://web.mit.edu/6.033/www/papers/gettys.pdf http://web.mit.edu/6.033/www/papers/allman.pdf http://web.mit.edu/6.033/www/papers/allman.pdf
adamheins 10y ago
I don't know much about the different router brands and models. Which would you consider to be some of the good ones?
- jumpCastle 10y ago
  I'm not an expert on routers but if I was in the market I would look at the Wirecutter. I find their recommendations pretty good. thewirecutter.com/reviews/best-wi-fi-router/
- amq 10y ago
  A starting point for me: LEDE support
  - e12e 10y ago
    That's: https://lede-project.org https://lede-project.org or:“Linux Embedded Development Environment” - the new project born from OpenWRT. I've used it with some old d-link routers, and I'm quite happy. The documentation is still not stellar, but it has gotten better, and the configuration more consistent. It's quite easy to set up ssh-access with a key, and simply get a copy of configuration files with scp - ready to set up identical configurations on new router installs.
- corford 10y ago
  Draytek are a good choice at the consumer level.
  - Nexxxeh 10y ago

driverdan 10y ago

> I've noticed latency get worse and worse on the Internet since I started using it in the 90s and nobody seems to talk about it. You're remembering incorrectly then. Dialup latency was in the range of hundreds of ms. Early high speed internet access also had higher latency than modern connections. If you have latency problems it's likely a problem with your hardware.

torrent-of-ions 10y ago
I don't remember dialup latency. I only remember it having low bandwidth. I'm talking about ADSL latency in the 90s.
- moomin 10y ago
  Latency for a 56k modem was 180ms. Latency for a voice call on the same line was _much_ smaller. AFAICT, latency from my games console television to my TV is much higher...
  - NoGravitas 10y ago
    300ms was not unusual on a 56k modem, if my memory serves.
    - moomin 10y ago
      180ms is what I got on WirePlay. The number is burnt into my brain. If the machine wasn't co-located, I can envisage it being a lot higher.
  - alphapapa 10y ago
    When I accessed the Internet over a "56k" modem connection, my network latency was less than 100 ms, so how could that be possible?
  - torrent-of-ions 10y ago
    Yes, I believe you, I meant I don't remember what the latency was.

ElijahLynn 10y ago

If you are on shitty wifi, no app will ever fix that.

vram22 10y ago

>interleaving got enabled on my ADSL What is interleaving in this context?

detaro 10y ago
Multiple data frames are interleaved and transmitted together. Good for error tolerance, since bursty noise doesn't wipe out a complete frame, but small parts of multiple ones, which then can individually be fixed by their Reed-Solomon error correction codes. But it adds latency, since the sender has to collect data from multiple frames to combine, and the receiver has to receive the entire block before it can be decoded, instead of transmitting each frame as it arrives.
- vram22 10y ago
  Thanks.

peraquins 10y ago

If you're ok with the Chinese gov snooping, I find WeChat to be amazingly good at voice chat, even if your connection is bad.

jdfreefly 10y ago

Have you tried Amazon Chime? We are ridiculously aware of the latency issue in high quality voip. We do not at the moment support user generated keys the way Telegram does but we do use TLS/DTLS encryption end to end for our VOIP streams.

CharlesW 10y ago
> We are ridiculously aware of the latency issue in high quality voip. We do not at the moment support user generated keys the way Telegram does but we do use TLS/DTLS encryption end to end for our VOIP streams. Any plans to make this available to developers as a service via AWS?
- jdfreefly 10y ago
  Sorry, I can't really comment on our roadmap and our priorities within it.

Animats 10y ago

The best phone voice quality is still ISDN. Digital all the way to the handset, and rigidly clocked with fixed delay. Switzerland has ISDN phones, and the voice quality is impressive.

DaiPlusPlus 10y ago
ISDN was 64kbps pet channel though - and to my knowledge did not use a wideband voice codec by default (or at least didn't mandate all ISDN receivers support wideband audio). Lots of people who had ISDN kit plugged their old POTS phones into their TAs anyway.
- doctorshady 10y ago
  ISDN is actually used quite a bit in niche markets in the states. For studios, typically you'll see a couple B-channels bonded to give someone 128K MP3 or AAC. At a bare minimum, 64 kbps/16khz g.722 is used. I worked in a newsroom a few years ago where I frequently had to record a weather guy from a distant area who used this.
  - nickpsecurity 10y ago
    Used in STE units for secure communication as well: https://en.wikipedia.org/wiki/Secure_Terminal_Equipment https://en.wikipedia.org/wiki/Secure_Terminal_Equipment
- dboreham 10y ago
  I think it sounded good because it was clean 64K end-to-end. No trickery with transcoding a bunch of times along the way so someone could save a nanopenny.
  - Animats 10y ago
    Right. The bits you send are the bits that are received, clocked at a fixed rate. No compression or timing artifacts.
10y ago

Florin_Andrei 10y ago

> But the most annoying problem with any internet voice chat is not so much the quality but the latency. Landline phones have noticeably lower voice quality but one can still enjoy a conversation perfectly well. High latency, on the other hand, absolutely ruins the experience. Maybe this is because traditional communications were built by people working exclusively in that field - and they eventually became true experts, with a deep understanding of the real issues. Nowadays Internet chat services are cobbled together by happy-go-lucky engineers who, prior to this, maybe did an ads firehose on Twitter, and after this project will perhaps move on to work on Facebook games or whatever. Smart people to be sure, but their understanding of the field is necessarily superficial. The fast rate of change in this industry explains many of the things that are ailing it currently.

dmix 10y ago
> Nowadays Internet chat services are cobbled together by happy-go-lucky engineers who, prior to this, maybe did an ads firehose on Twitter, Skype has been around for 13 years. It isn't a young startup full of ex-node.js programmers. WhatsApp also has some excellent infrastructure built on top of Erlang which has decades of experience and refinement of heavy-duty telecom usage baked into the framework. If anything the experience of previous development trickles down and younger engineers benefit from that knowledge. But I'm not sure I buy the idea that you can only build great systems if you've only been doing that one category for your whole career. The rate of jobs changes faster and in front-end development the pace of toolset changes pretty fast. But even in front-end development it's still the same languages. usually just the tools are the trendy part. And if it's switching languages every 5-7 yrs that knowledge of languages is transferable. I've only benefited as a programmer from exposure to many other languages (and projects). Regardless, the hardcore backend guys you get to build this type of infrastructure isn't the same hipster front-end devs that Twitter hired in the early years to build out a consumer product. Consumer is very different than building out telecom infrastructure in Erlang/C++.

l1ambda 10y ago

The Opus codec ( http://opus-codec.org/comparison/ http://opus-codec.org/comparison/ ) can enable very low latency audio. 26.5 ms algorithmic delay (the time it takes to go through the algorithm) by default, but can be set to a very impressive 5ms. Digital audio can't achieve the near no-latency analog POTS has, but it can get pretty darn close e.g. if Opus is used in the very low latency mode. I believe Mumble and maybe TeamSpeak let you choose the Opus latency. I know some other apps like WhatsApp, Wire, and Signal use Opus as well. I don't know what settings they use though. Another problem is cell phones add ~150ms latency (pre VoLTE) and people that route calls through Google Voice adds an additional 150ms+ latency on top. If you ever talk to someone on a cell phone on google voice, it sounds like they're calling from the moon. 150ms is about the human limit before turn-taking and talking over each other becomes an annoyance. That's why cell-to-landline conversations (150ms one way) were tolerable, but cell-to-cell (300ms one way) conversations are abysmal. And that's all in the same area code. Once you add long distance, it gets worse.

doctorshady 10y ago
Just fyi, POTS is digital up until the very last mile. While the network is digital, it uses time multiplexed serial PCM streams under the hood. Aside from the delay imposed by fiber transmission at the trunk level (~40 milliseconds from one side of the US to the next. If you're using a long distance network that packetizes the connection or sends a call on some weird route, that can certainly increase), the g.711 encoding delay should be fixed at 0.125 milliseconds.
- l1ambda 10y ago
  Interesting. I didn't know g.711 was so low. Typical VOIP latency must then stem from D/A and A/D conversion and internet latency.
  - makomk 10y ago
    g.711 has such low latency because it doesn't take advantage of inter-sample redundancy in any way whatsoever - it's just raw 8-bit samples at 8 ksps with a compander on either end (traditionally on the analog side of the DAC and ADC).

doctorshady 10y ago

Unless you're hitting an IP node that transcodes to g.729 or something, landline phones should sound pretty good; the mu-law codec uses companded 8-bit PCM that roughly equates to about 14-bit linear in terms of precision. If you're sending a lot of very low frequencies (most phones and line cards have a pass filter in them to remove stuff below ~60-100 hertz, so while the network is very capable of sending it, it doesn't always in practice), you'll start to hear companding noise, but it's rarely the case that it gets loud enough to cause an issue. As for the sample rate, 8 khz is a bit low, but you can make it sound significantly better with a bit of equalization; usually turning down frequencies near 1 khz a little goes a long way. Sometimes too, it doesn't hurt to gently (and I stress, gently) reduce frequencies near ~300 hertz if it sounds too muffled.

77pt77 10y ago

Try using Skype over a 3G connection. The latency is almost palpable.