Ask HN: Why isn't VoIP better? - <AUTHOR> I'm very sensitive to bad call quality. I'm very easy going about most things, but poor sound quality or an intermittent connection on a call, make it completely impossible for me to maintain normal communication and also makes me extremely cross.
As a result I am very attuned to the general quality of different call services. I always know when I'm on a PSTN that terminates to another PSTN - yes maybe the voices are a bit trebly but the connection is so consistent and wonderful. I actually appreciate the quality of those calls.
Skype, Viber, our own corporate VOIP system (whatever the hell it is) all drive me completely nuts. I have 240MB/s at home, but still VOIP lets me down: about 1 in 3 calls is of a quality I am happy with.
I guess my question is: will this ever be solved to my satisfaction, or is it simply the nature of the beast and "good enough" for most people? <comment><USER2> I'm surprised that no one has mentioned bufferbloat yet. In most home routers, people can see _seconds_ of delay when there's significant other traffic on the link. (And remember, "other traffic" can be ordinary web browsing, where pages average 2 mbytes these days...)

The latency (and jitter) caused by bufferbloat has been a solved problem for almost five years: fq_codel, and the newer cake qdisc's are are in the Linux kernel, and people's home routers now could install LEDE/OpenWrt ([https://lede-project.org](https://lede-project.org)). Two links:

What can I do about bufferbloat? [https://www.bufferbloat.net/projects/bloat/wiki/What_to_do_a...](https://www.bufferbloat.net/projects/bloat/wiki/What_to_do_about_Bufferbloat/)

Test your network for bufferbloat: [http://dslreports.com/speedtest](http://dslreports.com/speedtest)

<comment><USER3> Bufferbloat is a side affect of congestion or microbursts, and not the primary cause of poor voice calls. Either packets get dropped (shallow buffer) or queued (enough buffer), and in both cases the quality of the call will suffer.

Mitigation: Don't congest. If you need to, mark VOIP traffic with a DSCP value that has queues that are serviced prior to all others (e.g., EF, DSCP46).

<comment><USER4> Applications do it already, but edge routers are adept at ignoring these flags or applying them only to their own service.

So that you use it and not competition.

Net neutrality would be the antidote.

<comment><USER5> If those flags provide better quality of service, why am I not applying them to all of my traffic? Any attempt to "know" if my use is legitimate would undermine network neutrality.

<comment><USER2> Sure, you could make yourself crazy trying to prioritize all the traffic with varying kinds of flags.

Or you could just let an algorithm (SQM - fq_codel or cake) automatically determine which flows are sending more than their fair share of traffic into the bottleneck link, and offer backpressure to slow those applications down, so other applications work well.

re: net neutrality. That applies to whether my ISP (or any provider upstream in the Internet) should prioritize packets/data from various sources toward (or from) me. The answer should be, No. My contract with my ISP is that I get X mbps of data. The contract doesn't say the ISP will provide higher priority or data rate for service Y over service Z...

SQM has no effect on net neutrality: I'm simply configuring my own router to prioritize the data that I'm requesting (or sending). My ISP should handle them in a totally neutral way...

</comment></comment></comment></comment></comment><comment><USER6> Lots of good answers, but there is another one that's important too: echo cancellation. Most telco equipment has very good echo cancellation hardware in it. It's not impossible to do a decent job in VOIP, but not all of the software does it well. This can be _very_ important when you have high latencies or if you are doing conference calls.

The other main overlooked problem is automatic gain control in software. This one drives me crazy. Most people are too lazy/clueless to set the gain properly on their microphone, so VOIP client makers tend to use automatic gain control to adjust the input. However, it works _very_ differently that what you are used to in a telephone. If someone laughs loudly, or coughs you end up losing the beginning of the next sentence. It can work extremely badly in conjunction with echo cancellation depending on the software -- to the point where the audio can be very difficult to understand.

It's probably my biggest gripe with Google Hangouts because there is no consistent way to shut of AGC on all platforms.

I've written VOIP clients before and it is possible to have incredibly good audio (much better than normal PSTN -- mainly because the codecs are better). But it's very tricky to get it to work on random hardware, with random network conditions and with users who don't want to configure their client optimally.

<comment><USER7> I'm going to agree with you and take this opportunity to plug [http://cleanfeed.net/](http://cleanfeed.net/) We run a free 'VOIP' system that does zero processing on your audio.

It's actually for broadcasters who just want to get live audio from A to B, untouched. But as it's two way (and multi-way) we're finding users contacting us to say they're using it for regular 'conferencing'.

I suppose we are also doing it ourselves; we used Cleanfeed to communicate when developing Cleanfeed.

Elsewhere in the thread people mention jitter etc. issues. In my experience, 200-300ms of latency has much less negative impact on rapport than aggressive audio processing.

<comment><USER8> Thanks for the recommendation!

FAQ #1 should be "how does this remain free/why is this not going to disappear"?

<comment><USER7> Thanks, good suggestion. It's actually on the front page that our free offering is a spin-off of something commercial. But a FAQ to clarify this particular point would be sensible.

</comment></comment></comment></comment><comment><USER9> Jitter is one of the words you're looking for. 20 milliseconds does not sound like much, but when it's a delay in some (but not all) packets in a call getting to you, you can sure hear it!

Jitter is why you could have a 1 GB/s connection at home, and still have the same occurrence of poor call quality.

QoS (Quality of Service) can help, but only if it is applied at every router the traffic for a call goes through. QoS-assured bandwidth is more expensive, and cost-conscious VoIP providers don't seem to be willing to pay for it.

(edited to add) I'm a Skype for Business admin.

<comment><USER10> > QoS (Quality of Service) can help, but only if it is applied at every router > the traffic for a call goes through.

You really only need it on routers where a bottleneck is. If a router is forwarding 100% of packets immediately there is nothing for QoS to do.

In practice what this means is that the most important place to have it is at the border where you go from high bandwidth LAN to lower bandwidth internet.

In particular if you control a router there, make sure it knows to limit traffic to what the next hop link speed actually is. In will often see the link as gigabit ethernet even if it's slower than that and then forward all traffic when you really want it to be the device that chooses which traffic gets dropped.

Unfortunately if your ISP supports burst bandwidth it's not always easy to get it right without breaking that.

<comment><USER9> QoS depends on tagging (DSCP), and if one of the routers along the way doesn't support it or isn't configured for it, those tags are often dropped - or at least that's what I've seen in our company's network.

<comment><USER11> Like so many other things in tech, "it depends".

Network gear can be configured to treat frames/packets in various ways -- leave the ToS/DSCP bits alone, strip them out, or remark them. At the edge of a network, it's common to mark them. In the core of a network, you'll usually leave them alone (but honor them). On connections to third-parties (such as ISPs), it's not unusual for them to strip your QoS markings -- unless you're paying extra for them not to.

For a random packet that leaves your home network and hits the Internet (even VoIP traffic), I'd be surprised if it ends up at its destination with those same QoS bits set.

Like _zrm_ said, though, if there's no congestion on links, prioritizing traffic typically isn't going to do you much good. It's only when there's contention that moving packets to the front of the line helps out.

<comment><USER4> Except Internet routing is not smart, the routers cannot send you through a low bandwidth but uncontested and low latency pipe when your packets are unmarked because you didn't pay a premium for a peering agreement.

It actually can hurt the Internet performance as a whole, as the tiny VoIP packets prevent Jumbo throughput ones from working really well.

It takes just one bad hop to ruin latency.

<comment><USER11> Sure they can, especially on MPLS circuits sold for exactly this reason -- and with SLA commitments to deliver packets w/ specific markings within specific parameters (i.e. up to 5 Mbps of traffic marked ef will have maximum latency of 30 ms between A-Z).

Not on your home Internet connection, perhaps, but if you're a company and you want guaranteed performance levels (i.e. for VoIP traffic), you can certainly get it.

</comment></comment></comment></comment></comment></comment><comment><USER12> I think your complaint is about VoIP over the internet. And that's simple, the internet is best effort. Packets get dropped and that's expected. Also, packets can take a long time to reach their destination and there are no guarantees for jitter. This absolutely sucks for voice quality and there's nothing anyone can do about it.

<comment><USER13> Your landline is now VOIP too, it may be twisted pair to the central office, but then it's digitized and transmitted on the carriers owned networks.

</comment></comment><comment><USER14> Both ends need to support wideband codecs, as does your phone (hard or soft phone). This includes the VOIP providers needing to be able to use wideband codecs between each other.

If you achieve a wideband setup the call quality is amazing. I've used it for years. It really sucks when you have to call someone on a legacy POTS line or using a narrow band codec (bad voip, cell phones, etc)

On that note it's unfortunate that cell phones which can do wideband codecs within the same carrier (HD Voice!) won't work if your call leaves their network.

<comment><USER15> > On that note it's unfortunate that cell phones which can do wideband codecs > within the same carrier (HD Voice!) won't work if your call leaves their > network.

This is quite annoying, but it's because the PSTN overall is stuck on G.711 and G.729 - my SIP provider doesn't support anything else because they would be forced to re-encode the signal anyway and lose quality doing so.

Ultimately, we need to be putting pressure on the telco's to enable support for G.722 - until that happens we're going to be stuck with 4KHz audio for a long, long time.

</comment></comment><comment><USER16> First, as others pointed out, you will need a decent internet connection, with minimized packet loss and jitter. Most metro area data connections and quality WiFi access points will be sufficient. However, the wifi router that comes from your provider and stuffed away on the other side of the house might not work very well.

Secondly, good codecs make a world of difference. This is where I'll add the disclaimer that I'm in the collab side at Cisco. But I have the opposite experience using systems on my phone, my mac and with big telepresence units: the quality in all 3 settings far surpasses what's on PSTN, and if you're using good speakers or headsets, you can really appreciate the good quality.

In your case, it sounds like you might benefit from a better network access point at home.

Shameless plug #1: check out Cisco Spark. We still have some rough edges but I think the call quality is what you're looking for. I'm able to carry on a call and switch from wifi to LTE without dropping, or turn on VPN in the middle of a call and just get a few seconds pause before it recovers.

[https://www.ciscospark.com/](https://www.ciscospark.com/)

Shameless plug #2: check out the new Sparkboard. It's got 12 microphones and does beamforming in order to filter our room noise away from whoever is speaking.

[http://www.cisco.com/c/en/us/products/collateral/collaborati...](http://www.cisco.com/c/en/us/products/collateral/collaboration- endpoints/spark-board/datasheet-c78-738316.html)

<comment><USER17> Does it automatically report your political and religious affiliation to chinese ministry of public security? Lol, zing.

<comment><USER16> It's end-to-end encryption. For large sites, customer owns the encryption keys on their hardware so we can't intercept/decrypt even if we're ordered to.

</comment></comment></comment><comment><USER18> VOIP calls are very sensitive to intermittent connectivity.

Sometimes wifi will work in batches. No packets will get through for a few hundreds of milliseconds (or seconds), then they will get sent all at once.

Sometimes UDP traffic is blocked completely and the call has to get relayed over TCP. Now no packets can get dropped and you get delays due to retransmits.

Sometimes 3g/4g connection disappears because someone entered a building an overpass or passed a corner in a city.

Sometimes the sides of the conversation are far and the audio delay is high.

If any side of the conversation encounters a problem described above both will get a bad quality.

<comment><USER9> Among other things, I'm a Skype for Business admin. We've found that having our users use Ethernet when working from home instead of WiFi eliminated a lot of audio quality complaints.

<comment><AUTHOR> At home I use Ethernet, a Blue Yeti USB microphone and a solid pair of Philips USB headphones. Of course I don't know what my counterparts on the other end are using.

I guess my problem seems to be that we swapped a system that had a consistent call quality for one that is inconsistent, for a whole bunch of reasons. It seems to me, as if the general quality of telephony took a huge step backwards. And I don't understand, in an age of technical progress, why we should (or do) put up with that.

<comment><USER9> The only thing that could make VoIP voice quality as consistent as PSTN is a consistent application of QoS.

And that is why I have very mixed feelings about Net Neutrality. Sure, I don't think that Verizon should be allowed to penalize NetFlix or $randomVideoStartup by favoring its video traffic over theirs, but there is no technical way for voice to be consistently good without a consistent handling of its data traffic.

<comment><USER19> Network neutrality doesn't really have anything to do with QoS. Network neutrality says you can't prioritize packets from Comcast over packets from Netflix. It doesn't say you can't prioritize packets a customer has marked as more important over packets the same customer has marked as less important.

<comment><USER20> How much of the VoIP contention is on the user end vs the ISP end though?

I have a 1 Gbps connection at home and can still have a VoIP call across the world mess up since I might not have the best route to whoever I'm calling.

</comment></comment></comment></comment></comment></comment><comment><USER21> If you use Wifi, keep in mind that it's built on a very fragile physical layer. In fact, the MLME expects up to seven failed transmissions before calling a packet dropped (otherwise it would destroy TCP's performance due to TCP's default congestion control mechanisms). The delays each way from two people speaking over noisy Wifi networks can already add substantial lag.

</comment><comment><USER22> Old fashioned PSTN lines would secure the full bandwidth needed end-to-end when the original call was established (which was really in efficient from a data utilization perspective).

With VoIP you are:

1\. Beholden to the bandwidth/latency issues of the crappiest link in your end-to-end path.

2\. Intentional use of poorer quality codecs (by your ISP and/or VoIP software to reduce aggregate bandwidth use). Ironically this can be inverted, there are codecs that sound much better than PSTN calls (but use more bandwidth).

<comment><USER15> > 2\. Intentional use of poorer quality codecs (by your ISP and/or VoIP > software to reduce aggregate bandwidth use). Ironically this can be > inverted, there are codecs that sound much better than PSTN calls (but use > more bandwidth).

I mean, people use G.729 for example because, yes, it is more efficient than G.711. But ultimately, if you want to communicate with the PSTN those are your only two choices and it's easier to just say "G.729 for everything" instead of wasting processing time on the server re-encoding everything that goes off the network.

You want better codecs? Bitch at the phone companies who have drug their feet on implementing G.722 while more and more IP phones and mobile devices continue to add support for wideband audio.

</comment></comment><comment><USER23> I've been in the VoIP industry for over a decade now, and in almost all cases of 'bad VoIP' I can trace the blame to a bad implementation of it. Let me give you an extreme examples...

\- It's fairly common practice in countries with poorly enforced telco law for people to resell access to the PSTN. They will connect to the PSTN (either via a GSM gateway or a server with an E1 card) and resell that access to a 'wholesale VoIP business'. \- That 'wholesale VoIP business' will have an infrastructure, cobbled together with open source technologies with no ability to scale, in the operational or technical sense. \- The same 'wholesale VoIP business' will undercut the cost to access the countries PSTN (compared to accessing it directly via a tier 1 carrier) and sell the access to other tier 1 carriers. \- The other tier 1 carriers lap up this cheap access, an embrace it because international calls fulfil a small portion of their multi-million revenues. \- Therefore, when someone in another country tries to call that country with poorly enforced telco law, it will route via the resold route until it's shut down by the law or it breaks because it's poor infrastructure. \- This is currently happening on a big scale in lots of African countries, some central american countries and in the middle east.

I'll leave the context of 'office to office' comms to the other answers as it's been covered in detail elsewhere. But I will say VoIP, all it's forms, is an old technologies, and problems relating to poor bandwidth.. voice quality.. backwards compatibility have been solved for many years.

</comment><comment><USER24> This is almost always a latency issue, and because nobody has figured out a way to milk extra cash for providing inter-network packet prioritization. In other words, this isn't an engineering problem, it's a money problem.

</comment><comment><USER25> Skype used to be fantastic when you could establish a point-to-point call. Today, I think everything gets routed through Microsoft so they can provide access to law enforcement. Some days it's good, often it's pretty bad.

<comment><USER8> I believe FaceTime is still point-to-point.

I've heard good things about WeChat and it's support for low-bandwitdth networks but haven't worked up the courage to try it out.

</comment></comment><comment><USER26> I don't believe what you ask for can ever exist. "Clear, Distortion Free, Low- Latency. Pick any two" comes to mind.

When you're doing voip you're doing so over UDP because TCP has far too much latency to be agreeable, but this means a lost packet will be a digital distortion. The nature of the internet is that packets get lost or arrive out of sequence- that's why TCP exists in the first place.

<comment><AUTHOR> 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.

<comment><USER14> VOIP is considerably better than PSTN if configured correctly. There's no comparison. Your sysadmins don't understand VOIP.

<comment><USER14> 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.

<comment><USER27> 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?

<comment><USER28> [http://voip.ms](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.

</comment></comment></comment></comment></comment></comment><comment><USER29> It depends. I've noticed that sometimes my AT&T cell calls are of a higher quality, depending on who I'm talking to. I think this has something to do with their HD Voice product.

I like it, the quality reminds me of what I've experienced in the past with the Cisco UC platform on a well-provisioned network and UC platform.

The goal of Skype, Viber, etc. are to always give you some level of audio across all networks, so the codecs they are using have a different goal than Cisco UC.

All these codecs have low bandwidth requirements, much lower than what you have. If you have a residential ISP, QoS might be an issue with some of the higher sampling rate codecs, but you probably won't notice them with Skype.

Try running an open source VoIP MCU at home on your LAN to see how good it could be.

</comment><comment><USER30> All quality problems would disappear if QoS would be implemented across the entire public internet. Your ears are so much more sensitive to small anomalies.

I've been deploying VoIP since 2003, both in the US and in Europe. In Europe it is easier to find cheap DSL with QoS applied, which allows laying in a separate circuit for VoIP only. In theses cases there are no complaints.

VoIP also works well on fiber. But as soon as one part of the network path travels the public internet all bets are off because there's no guarantee you packets aren't dropped because of traffic spikes.

Everybody who thinks the internet is stable in anyway is way off. Route flaps, packet loss, overloaded hops, last-mile problems happen all the time.

I could talk about this for hours.

</comment><comment><USER2> And there's good news on the Wi-Fi front as well. Many of the same people who brought us fq_codel/cake have new algorithms for "Making Wi-Fi Fast".

Their paper tells how they've reduced bufferbloat as well as increasing "airtime fairness" in the Wi-Fi stack. The paper, "Ending the Anomaly: Achieving Low Latency and Airtime Fairness in WiFi", is now available as a preprint at: [https://arxiv.org/abs/1703.00064](https://arxiv.org/abs/1703.00064) The conclusion of the paper states:

"We have developed a novel two-part solution to two large performance problems affecting WiFi: Bufferbloat and the 802.11 performance anomaly.

"The solution consists of a new integrated queueing scheme tailored specifically to eliminating bufferbloat in WiFi, which reduces latency under load by an order of magnitude. Leveraging the queueing structure, we have developed a deficit based airtime fairness scheduler that works at the access point with no client modifications, and achieves close to perfect fairness in all the evaluated scenarios, increasing total throughput by up to a factor of 5.

"Our solution reduces implementation complexity and increases accuracy compared to previous work, and has been accepted into the mainline Linux kernel, making it deployable on billions of Linux-based devices."

Fun reading! Oh, and it's available in LEDE today.

</comment><comment><USER22> So VoIP can be better... at the expense of needing more bandwidth. During the session initialization the two VoIP endpoints handshake over which codecs will be used (kind of like the Accepts header in HTTP). Just like images, higher quality codecs tend to use more bandwidth (though like images its much more nuanced than that). Similarly like image formats there's different amounts of loss in the encoding/decoding of the voice data.

Voice has a major difference though. Its generally a real time protocol (its data packets are called RTP and sent over UDP generally). With images you can afford to wait longer for high quality images to load over a slow network. In a phone call that delay gets a bit annoying (to put it lightly).

Most people would rather trade quality for timeliness. You ISP would rather throttle your bandwidth and may not care about your quality (to a point).

Its quite a bit more complicated than that, but in a nutshell with VoIP, "higher quality" = "more bandwidth". If you don't have the bandwidth you'll have a very hard time getting better quality. If you have the bandwidth locally, then your provider or some network link along the way either doesn't have the aggregate capacity or may be actively throttling your rate to limit usage.

<comment><USER31> It really depends on which codec you're using. We had an implementation of G.711, which is not bad, but far from great. However, the license costs for implementing G.729 were so outrageous that my previous company simply couldn't afford it (or wouldn't). G.729 offers better call quality for the same bitrate, or similar call quality for much lower bitrate. Typically, G.711 would use ~64Kbps, whereas G.729 can achieve the same call quality for ~8Kbps (there's also a variable encoding that scales between 8 and 32Kbps). This being said, G.729 also requires a lot more CPU processing power.

If I'm not mistaken, each party of each leg of a call needs a license, which means that for a typical call centre with an IVR server between the agent and the customer, you need:

Summary:

\- PSTN converter to IVR: 2 licenses / per customer

\- IVR to agent: 2 licenses / per agent

\- IVR to supervisor: 2 licenses / per supervisor.

It's pretty normal to see 300-agent call centres, and you typically want at least 300-900 calls waiting. Let's say there's between 10 and 30 supervisors.

300 * 2 + 900 * 2 + 30 * 2 = 2460 licenses. At $6-10 per license, that's more expensive than most hardware PBX solutions alone!

For IVR companies that handle multiple hundreds of thousands of calls per day, it's not really worth it.

Edit: Formatting

<comment><USER32> That licensing scheme is _INSANE_ , especially since codecs like Opus exist, and do a much better job.

Once you touch the PSTN though, you're uLaw anyway. Why would you pay that much just for in-network voice quality?

</comment></comment></comment><comment><USER33> Single anecdotal counterpoint:

My corporate VOIP system is made of Polycom phones connecting via SIP to local Asterisk servers, and then those have interconnects to PSTN gateway services.

The quality on-campus is uniformly extremely high.

Some people have the Polycom phones at home (for working remotely) and calls to/from those are of similarly high quality, even in conferences.

Connections to cellphones range from mediocre to bad.

Connections to POTS land lines range from bad to pretty good.

Overall I would say that our VOIP system doesn't have a problem.

</comment><comment><USER34> Latency and speed are two very different things. For good audio quality, the first is way more important than the later.

Also, lossy codecs usually drop a good chunk of information in favour of compression, working with larger packets for better efficiency and adding even more latency. In that sense, G.711 or higher-quality PCM is the best if you want lowest latency possible.

I've worked for some time with telecom to know that each time you add a complex codec (G.72x) you may end up doing a few conversions on the way: in most gateways, if audio comes in as codec X, it gets converted to PCM internally and then re-converted to codec X again before going to the other end. Adding bypass logic adds little benefit as you have to scale the system for worst case scenario (all channels performing the costlier compression), so bypass only complicates your code.

So those "X ms of algorithmic latency" for complex codecs may end up being multiplied a few times: throw in echo cancelling, jitter buffers, etc and you get a call where you end up interrupting each other all the time due to audio delay.

</comment><comment><USER35> You might try Google Fi. Not Google Fiber, Google Fi ([https://fi.google.com/](https://fi.google.com/)).

I felt the same way about the difference in quality between cell phones and old-fashioned landlines. "Telephone voice" used to mean "bad audio quality," but compared to cell phones, the old telephone sounds golden. The nicest thing about PSTN-to-PSTN calls is their timing. Cell phones have a delay that always trips me up and keeps us from getting into any kind of conversational rhythm. Again it's another thing that I didn't appreciate back in the '80s, when landlines were all there was.

Finally I got Fi. The difference in quality is refreshing, even when talking to people on cell phones with different carriers (which is all the time, because no one else seems to know about Google Fi). The latency is much less, and the sound is clearer.

Ready to kick it into high gear? I can make phone calls on my laptop. Plug in earbuds with a built-in mic, and phone calls never sounded so good.

No, I don't work for Google.

</comment><comment><USER36> I find I agree.

\- lossy vs lossless codecs (PSTN isn't analogue end-to-end any more, but is lossless as far as I'm aware. VoIP seems to have been about squeezing as much traffic down the pipe as possible, at the expense of quality.)

\- circuit vs packet switched networks (Congestion, contention, jitter on shared packet switched networks. While I'm sure PSTN doesn't have "physical" circuit switching any more, I do get the impression that high-bandwidth bearers are chopped into fixed-width chunks, like 64k ISDN, and will reach capacity and stop accepting demand rather than degrade individual connections)

For me, wideband audio helps, even with the jitter and lossy codec. This is starting to become common in the UK on cellular calls between smartphones, even between different network providers. Audio bandwidth is comparable to Facetime Audio.

<comment><USER37> The lossy part of PSTN occurs at the front-end where only a 3KHz band of audio is passed through the network.

</comment></comment><comment><USER38> Data Scientist for Republic Wireless here. We operate a hybrid mobile phone service that leverages both VoIP and traditional circuit switched networks to let our customers make calls.

When asked to rate the quality of a call, our users consistently rate pure VoIP calls higher than their pure circuit switched counterparts. A significant portion of that is due to our "bonded calling" technology - if a VoIP call over Wifi starts to degrade, we can supplement the audio stream with additional packets sent over LTE until Wifi issues calm down.

When we first rolled that tech out, we saw a marked improvement in call quality measures, and a noticeable drop in customer support complaints about call quality.

In the end, I can't say this issue will ever be solved to OP's satisfaction, but we're making some incredible strides.

</comment><comment><USER39> Have you tried Wire's audio? The creator of the SILK (Skype) and Opus codecs is on their engineering team.

[http://opus-codec.org](http://opus-codec.org)

[http://app.wire.com](http://app.wire.com)

<comment><USER40> Wire's audio quality is the best in my experience.

</comment></comment><comment><USER41> Ive seen some enterprise level voip providers in the UK not only provide clients with VOIP systems either on site or a cloud PBX, but also a dedicated leased ADSL line. Usually 2mbps both up / down (They say this is good for about 20 concurrent users). They also connect the the line directly to their voip server, so it dosnt have to make multiple hops.

Never tried one in practice though. We run our voip over our main 200mbps down, 15 up fibre connection. Have QOS and VOIP prioritisation setup on the router, SIP ALG off etc. Seems pretty good, but as you said, if you know what to look for you can always tell your on a voip call.

<comment><USER42> Is that ADSL? If it's 2 up/2 down, is that not SDSL?

<comment><USER43> While A in ADSL stands for Asymmetrical, it is primarily name of the used DMT- based technology, S(H)DSL is completely different thing (in fact it is related to 1000-Base-T, but with significantly smaller symbol rate) that does not work across non-dedicated lines (and usually needs carefully selected pairs to work reliably).

<comment><USER42> That's counter-intuitive! TIL, thanks!

</comment></comment></comment></comment><comment><USER44> When I first tried JamKazam (a-play-in-a-band-over-the-Internet-thing) and talked to my friends over low-latency, high-quality gear, I was amazed but how much it felt like being in the same room.

That's when I realized that most conferencing solutions simply have too high latency. It's a matter of milliseconds - if there's too many of them people start interrupting each other by talking at the same time, and the feeling of having a natural conversation is lost.

They'd probably do better financially by selling their technology as a conference solution. ;-)

</comment><comment><USER45> We have several VoIP lines on several locations. The quality is actually great. If you want guaranteed quality, use a dedicated voice VLAN or some sort of QoS.

Even in places where we don't have this in place, it seems to be working just fine for us. I can also say that I've used Skype to a PSTN in a separate continent over a slow mobile link and it also worked.

However: I have to mention that the quality is never quite as good as normal Skype (so called "HD" quality).

</comment><comment><USER46> Your first paragraph could have been written by me so it was nice to discover I'm not alone. However, I find FaceTime audio calls to be of such higher quality than regular phone lines or most other VOIP services that it is my go- to for calls where I need to be able to concentrate. The only other service I've found to be comparable is Slack's calls.

</comment><comment><USER47> Just as an aside, a potentially good alternative for international calls are callback services (depends on the service provider of course, could also be worse) as you will be able to skip your local networking hardware with them.

I don't have any specific recommendations though, haven't used one for years.

</comment><comment><USER48> 8kHz pcm ulaw or alaw isn't exactly stellar, and high bandwidth Skype or sip is much much better really.

Maybe you are conditioned to like the can-effect, your friends have really shitty headsets, or there is something wrong with the available bandwidth...

<comment><USER49> To clarify this a bit, SIP (session initiation protocol) is just a signalling protocol, meaning it is used to set up/tear down media streams between endpoints using various codecs. The sip standard includes a section of the message which is included in specific types of negotiation messages (invite, 183, 200 ok) called the SDP (session description protocol) to determine/negotiate which codec that gets used for the call (or video session).

In SDP you specify a list of the codecs you want to use in the media stream. The switch you are hosted by (either your Telephone Service Provider or Carrier) then determines which of these codecs match the other leg of the call it will be bridging you to, and chooses one of the codecs you provide to use for the media session. Sometimes it will be decide to transparently transcodes the call to another codec if required by the destination network. The payload in the media stream (called the RTP stream or Real-time Transport Protocol stream) will be encoded in the codec you specify, but may be transcoded on the way to the destination if required by the intermediary hop.

Among the codecs you could use, there are high definition codecs like G722 that sound really great and are becoming much more popular (this is the skype high definition audio codec IIRC), but if a call traverses the PSTN at all, that call will get transcoded down to G711U (USA) or G711A (everywhere else) and the quality will be much lower. If a call stays pure voip, there is a chance it will stay at G722 but typically that is the problem, you will encounter when calling to other phone systems that are still on the old circuit switched infrastructure.

So ultimately the quality of the call is dependent on these factors, how much bandwidth you have available, what codecs your VoIP provider supports, the quality of the path call takes and the codecs supported by the carriers the call traverses to get from your provider to the destination phone, and the type of endpoint you are calling IE: if you are calling mobile phone, a phone on another voip provider, or an old POTS line.

A POTS (plain old telephone service) line is (now sort of) dedicated from point to point. You might be surprised to learn that on the backend, most carriers use a lot of voip for internal network or handing off calls to other carriers. The old phone system reserved a 64k circuit for the entire path of the call which only contained the voice data for that specific call. This provides some quality guarantees are there that do not exist for packets traversing the public internet. VoIP on a private network with sufficient bandwidth and QoS for the media stream is always going to at least sound as good, if not better than the POTS network because the compression to squeeze the voice data down to a 64kbps was not very good when that stuff was invented. With voip you can use the new codecs, which nearly always results in a better call quality. This all depends on the provider you are using though, if they are hosting their services on AWS and using crappy carriers, your call quality will suffer.

</comment></comment><comment><USER50> @scandox: Here's some insight from an Enterprise VoIP perspective that you may find useful, from CiscoLive last year: [https://youtu.be/zbjT3Fy4y1s](https://youtu.be/zbjT3Fy4y1s)

</comment><comment><USER51> Could it be the upload speeds or QoS settings? I don't use voip, my wife skypes long distance and it seems pretty good for both video and audio and our internet is only 2ish Mb down and 1ish Mb up!

</comment><comment><USER52> Some models of LG G5 have VoWiFi. You can make calls even when your cellular network is not available as long as you have an internet connection. Unfortunatly LG G5 H-860 doesn't support VoWiFi.

</comment><comment><USER53> Where do you live where you can get 2 gigabit speeds at home?

<comment><USER54> It's more likely he meant 240Mbit/s

<comment><AUTHOR> Yes sorry I did

</comment></comment></comment><comment><USER55> I always thought that cellphone calls were routed over ip networks and were using exactly the same low-quality codecs "real" voip is using.

</comment><comment><USER56> Skype also has to deal with routing some of the calls trough their servers. Audio is probably crippled to save bw.

</comment><comment><USER57> VoIP isn't getting better because vendors are not willing to delay output for better UX.

</comment><comment><USER58> Little trolling: so you say you have 1.6Gbps connection at home?

Maybe your connection has some latency issues?

<comment><AUTHOR> Yes sorry I'm a dufus about these things. The issue has been thoroughly gone into down below.

Mea culpa.

</comment></comment><comment><USER59> It's probably because most people have horrible quality microphones.

<comment><USER22> Sorry to down vote, but that's completely the wrong vector. Its a network issue (of some sort - bandwidth, latency, throttling).

<comment> FLUX-YOU How are games and voice software targeted at gaming managing to make things so much more clear?

If it's a network issue, it's likely unsalvageable because the network is just crap. We need to be using whatever those are using. It is usually crystal clear and people are easily understandable even during large gatherings (EVE Online battles).

I'm the same as the OP, and what's worse, people talk 20 feet from the cisco conference phones (because those are everywhere) and it makes everything sound completely terrible. I don't listen half the time. _That_ is definitely a microphone issue.

<comment><USER22> If you're audio sounds great for a network game and crappy for a phone call (to the same person you are playing a game with), its because some one (your ISP, your VoIP software, the other parties ISP, or VoIP software, or any network hop in the middle), is intentionally choosing poor quality codecs for you to lower your bandwidth utilization.

I've heard directly of novice VoIP engineers unintentionally taking out corporate networks trying to "help" improve audio quality, by picking a higher quality codec. It worked great when they tested phone calls in the middle of the night maintenance window. It really sucked the next day when packets started colliding. Crappy sounding codecs sound a heck of a lot better than 50% packet loss.

</comment></comment></comment></comment><comment><USER60> Lots of great feedback here. I know from my own experience that the leading culprit of bad VoIP (and video) quality is a poorly performing ISP connection.

I'm one of the co-founders of Firebind. We use active traffic to continuously assess network quality of last mile ISP links, with a focus on sites using VoIP and video. Our software agent tests back to the cloud (combo of AWS and customer chosen destinations) every 5 minutes and makes 11 measurements by putting traffic on the wire. Two of those measurements are upload and download packet loss based on a synthetic G.711 VoIP call that runs for 25 seconds in each direction to one of our AWS hosted agent clusters.

By graphing the upload and download loss 24x7x365, we can see how good the network quality is on an ISP circuit far better than you can with a TCP Bandwidth test (which is effectively a destructive test) or anything ICMP based. And since we use active traffic, we have visibility even at 3am when no calls are being made.

I've reviewed hundreds of agent charts personally just in the last few months, many times with the customer. By far the number one issue is a bad quality ISP circuit, followed by an occasional bad or misconfigured network device at the customer site. Every once in a while the hosted VoIP provider might have some temporary glitch, but it has been pretty rare in my experience. Similarly the WAN path beyond the local ISP is usually not the problem.

Oversubscribed links are a very prevalent issue, especially with cable-modem based circuits where the upload bandwidth is a fraction of the download. Some of the things I've seen recently...

\- Cable-based ISP circuit in Concord, CA whereby the upload packet loss increased in proportion to the daily high temperature \- Customer site who accidentally configured their backup process to kick in at 3pm instead of 3am and hence flooded the upload path during working hours \- Bad loss in both directions that was eventually traced to a bad coax line that had been nicked by a lawnmower \- A cable modem with a memory leak that would gradually drop packets at a rate increasing up to almost 20% over the course of 2 weeks, where a power-cycle would "fix it" for a few days until the leak grew again and dropped more packets. \- Incident with 2 Charter sites in 1 state and another in a different state whereby all 3 had transient 70+ percent download loss for three days back to our test site at AWS in US-East-1 (I sampled dozens of other agents around the U.S. and only the Charter sites saw this issue.) \- And of course, the Netflix effect. Dozens of times I've seen download loss spike during prime-time TV hours, and that's the only time each day that remote site has download loss.

The best test you can do with open-source tools is to use iperf out to the cloud (to a $5/mo Digital Ocean droplet perhaps) on regular intervals and simulate the codec you care about. We built our own iperf-like solution that is highly concurrent but you can copy our settings and aim for 87 Kbps with 218 byte payloads over UDP. That should give you 50 pps of VoIP.

And of course for the shameless plug, if you want to check out Firebind, you can find a free trial link at our site. [https://www.firebind.com](https://www.firebind.com).

<comment><USER61> I would be very interested in trying to see the results of the firebind stuff, with our stuff (fq_codel and cake based "sqm") in place on the link. Could you drop by the bufferbloat mailing list and chat with us?

Also: We use a tool that we consider much better than iperf alone, it's from "flent.org" and gives us the abilty to inject loads and measure the side- effects.

</comment></comment><comment><USER62> PSTN uses a digital backbone these days; it's just supersized carrier-grade digital backbone, not 'what hacky software can we get on the cheap'. Phone companies prioritise their voice traffic over their own backbone and have network admins monitoring it closely, whereas your skype calls are just 'more traffic from the proles' and get to compete with everyone else.</comment>


Why Working on Monoliths Is Bad for Your Career - <AUTHOR> https://lightstep.com/blog/why-working-on-monoliths-is-bad-for-your-career/ <comment><USER2> Despite having spent the last two years working specifically on addressing the operational issues caused by a monolith, I couldn't find anything apart from the title and the implied pro-microservice premise in this post that I agree with.

A monolith's tests need not take all day to run. Features don't need to take weeks or months to ship—one can practice agile even inside a monolith. A monolith does not preclude the use of modern tooling—we adopted CI/CD and distributed tracing despite having a monolith, and use them to great effect.

In some cases scaling a monolithic codebase may be _more_ instructive about how to scale, because it's not as simple as "pay AWS more money and use more instances." Instead, you're forced to understand bottlenecks and consider how to solve them with algorithmic changes. Circuit-breaking is a critical feature in a monolith, even more so since one bad type of request could otherwise ruin the day for everyone.

The inability to "rewrite your business" has more to do with how well- architected your platform is than whether you have microservices. Both monolithic and microservices can be built well and built badly; a clean monolith is straightforward to rewrite the same way that cleanly-designed microservices are.

My personal opinion aligns with that of the author and greatly favors microservice architectures. But the usual hatred of monoliths strikes me as more dogmatic than well-reasoned, and I find this post to ultimately fail in defending its claim by falling victim to that.

</comment><comment><USER3> Either the author is trolling or they are trying to convince us to switch to Microservices, so that once we have a big ball of mud Lightstep can come in to save the day.

</comment><comment><USER4> They're just describing the difference between good development practices and bad ones.

</comment><comment><USER5> Is it just me or is the author confusing the concepts of ci/cd and microservices?

</comment><comment><USER6> BS

</comment><comment><USER7> bad</comment>


SoundFocus (YC S13) Launches With an App for '20/20 Hearing' - <AUTHOR> http://techcrunch.com/2013/08/14/yc-backed-soundfocus-launches-with-an-app-for-2020-hearing-with-mysterious-hardware-on-the-way/

<comment><USER2> This is an interesting case of a disadvantage becoming an advantage. The CEO suffers from significant hearing loss, but that enabled him to see a huge market that most 20-something founders overlook.

<comment><USER3> What's cool is that in addition to this being a tremendous product for an overlooked market, the comments here seem to indicate that it has great value even for non-hearing-impaired users. Excellent potential here.

</comment></comment><comment><USER4> Love the idea, but it does precisely nothing for me at the moment.

I have significant hearing loss in one ear. The app doesn't seem to distinguish between ears(?) So when I try to calibrate it, I can hear practically the lowest noises on all the screens, so it doesn't really seem to do much whether I turn it on or off.

It's not uncommon for people to have one considerably worse ear; in fact, I would guess it's a lot more common for someone to have one bad ear (due to an infection or similar) than it is to have equal problems in both, though I may well be wrong about this, but even if still think it's worth considering. (Edit: I can't find specific figures comparing incidence of unilateral vs bilateral deafness, but there are certainly plenty of people out there in a similar position to me, e.g. see [0] and [1])

I'm actually in a privileged position of having the results of a professionally-taken hearing test, so I know exactly which frequencies my hearing is bad at too. But the app doesn't seem to show the results of the test it does, so I can't verify its accuracy.

Still, I hope it improves, because I've always wanted an app like this (or, more precisely, an app that would generate a custom corrective EQ curve for each ear). I'll watch with interest... Certainly won't be deleting it yet anyway, and it's nice to see the app itself being so polished.

Edit: I almost think the set-up screens are too simplified. If someone does have significant hearing problems, they'll be more than prepared to spend a few minutes setting it up if it results in a better correction (e.g. more frequency samples and one for each ear).

[0] [http://www.hear-it.org/Single-sided-deafness](http://www.hear- it.org/Single-sided-deafness) [1] [http://www.singlesideddeafness.com/ssd_report.pdf](http://www.singlesideddeafness.com/ssd_report.pdf)

<comment><AUTHOR> Co-founder here - thanks for the great feedback Osmium. You're quite right in that the number of people who have unilateral hearing loss is pretty high (it's close to 40% from some recent studies).

We're working on developing a test that works well for people with single ear hearing loss for a future app update - would you be interested in trying out our beta? Please drop me an email if you are: varun [at] soundfocus [dot] com.

<comment><USER5> I've had significant hearing loss in one ear since birth and finally got around to getting a hearing aid after finishing school last year.

Right now I typically put the standard Apple earbuds over my hearing aid when walking around, but strongly prefer my Bose QuietComforts when I'm coding and want noise-canceling. This obviously isn't ideal because my hearing aid and the QCs don't play nice together, so I end up just taking out my hearing aid to use them. Really looking forward to seeing what's in store, and happy to help however possible - I'll email as well re: beta testing.

<comment><AUTHOR> Alex (my co-founder) had this exact problem except in both ears, and he uses our app every day for the same reason - hearing aids just don't play well with earbuds or headphones.

Thanks for signing up whbk, it's always great to get more beta testers to validate the changes to the sound processing as we keep improving it.

</comment></comment></comment></comment><comment><USER6> I was expecting to be pretty underwhelmed by this app, but I have to say, I'm impressed.

My office has a pretty loud HVAC system, and it totally kills the low end when I listen with a pair of cheap Apple earbuds.

I took the little SoundFocus hearing test, turned on their EQ, and the difference was pretty pronounced. It killed a bit more high end than I would have liked (seemed to roll off after about 12k), but the lows were definitely spot-on.

I'll probably use this during my commute on the super loud city busses.

 _edit:_

I don't actually suffer from hearing loss. I'm just picky about audio quality (former recording/mixing engineer) and have an unfortunate tendency to spend a lot of time in noisy places.

<comment><USER7> Adding noise can damage the ears, and adds more stress to the body.. just saying..

<comment><USER6> Not sure what you mean by "adding noise".

<comment><USER7> I assume if external noise is drowning out the music / changing how it sounds, then the music will be modified to let you hear certain sounds better via making them louder.

<comment><AUTHOR> loceng, we try to avoid that - a simple equalizer will create an amplification of frequency bands without gain control, so you get a sound of higher amplitude/magnitude, which can have a negative impact as you pointed out.

We use some techniques to rebalance your music - by making sounds that are softer in the mix louder, while keeping the very loud sounds at the same level. We also try to change the relative volume of the frequency bands you do not hear well, while preserving the overall volume.

Since psychoacoustics vary a bit from person to person, it may be perceived as mildly louder or softer than the original, but overall it will be at a similar volume level.

</comment></comment></comment></comment></comment><comment><USER8> My dad recently went to the doctor for his hearing and ended up getting a hearing aid. We had joked for a few years that he needed one - but were kind of surprised when it turns out he actually did.

He told all of his friends and family about how much his hearing and in turn day-to-day life has improved since getting them. Two of my uncles and one of his friends got their ears checked and ended up getting hearing aids as well.

I believe it when I read that 1 in 5 people need one. Goodluck to the SoundFocus team!

<comment><USER9> I think the article was saying that only 1 in 5 people who need a hearing aid have one, rather than that every 5th person should have one.

<comment><USER8> I read it correctly but wrote the sentence incorrectly.

</comment></comment></comment><comment><USER10> I love the idea of prefacing a hardware release with an app that is both useful to many people and could get some people interested in the hardware who otherwise wouldn't be. I myself am pretty sure I have perfect hearing but am interested in trying this out when I get home from work tonight.

</comment><comment><USER11> Excellent and long awaited initiative! I have a genetic hearing loss on both ears. It's a bit different on both ears though. Now I use Poweramp for Android with its excellent EQ to tune the audio as good as possible after my "cookie cutter" audiogram. On my computers I use the system wide EQ "Hear" [ [http://www.prosofteng.com/products/hear.php](http://www.prosofteng.com/products/hear.php) ]. One inherent flaw with all solutions I've come across so far though (as with yours?) is that EQ can't be tuned differently for each channel (left and right).

Nice idea though. But I'll have to wait for a system wide Android EQ (there is one, but it has too few bands [https://play.google.com/store/apps/details?id=com.smartandro...](https://play.google.com/store/apps/details?id=com.smartandroidapps.equalizer) ). Preferably combined with your hearing test based EQ (can't say much about it as I don't use an iOS device other than at work).

<comment><USER11> Slightly related: [https://developer.android.com/reference/android/media/audiof...](https://developer.android.com/reference/android/media/audiofx/Equalizer.html)

</comment></comment><comment><USER12> This is really cool. I have a friend who is a very talented multi- instrumentalist but who also has hearing loss in the low frequency range. I sent him this app and look forward to finding out how / if it helps him listen to his own recorded music.

<comment><AUTHOR> Thanks Muraiki - we'd love to hear what your friend thinks of the app.

<comment><USER12> I asked him if it helped, and he unfortunately said: "Actually no :( All it did was boost the bass up so much that it just got muddy." But I think he has a pretty severe impairment in terms of hearing bass frequencies, so I don't know if there's much that could feasibly be done.

</comment></comment></comment><comment><USER13> How are you processing the sound?

Enabling the algorithm significantly reduces sibliance, and alot of details in the upper mids disappear. Drums and guitar attacks in particular. Not especially pleasing. I tried playing with the equalizer presets, and they didn't seem to exhibit the same artifacts.

Then again, I'm hardly your target audience, and I was demoing on very revealing studio headphones. I'm gonna give it a shot on my commute tomorrow with earbuds. I'm guessing it improves clarity on vocals quite a bit.

</comment><comment><USER14> Question for the founders:

I'm completely deaf in one ear. But, I can't use normal headphones - the sound coming in on my right (bad) ear causes some health issues. As a result, I usually cut the right earbud wire entirely. This seems like a really dumb solution.

Does SoundFocus let me make all the sound come into the left earbud?

<comment><AUTHOR> keiferski, SoundFocus doesn't do this today. But stay tuned, we are working to help people who have single ear hearing loss as well.

I'm actually interested in learning more about your particular use case, I'll drop you a line on the email address in your HN profile.

<comment><USER14> Thanks, and feel free to email me!

</comment></comment></comment><comment><USER15> Dumb question: after completing the test, I didn't see "soundfocus autotuned" (or whatever) as an EQ option. Not sure if it is automatically applied and then you pick additional EQ settings to go on top of it, or what. Hope that makes sense

<comment><USER16> In the bottom left, there is an on/off toggle if that helps.

<comment><USER15> Thanks - I figured it out. There was a brief message on playback start that I must have missed a few times which let me know soundfocus was unavailable due to DRM. The bottom left icon says "\---- DRM" in that case so until I saw the message I didn't know what I was missing

</comment></comment></comment><comment><USER17> Ignoring the sound-loss test, I love this app simply because it's the only free equalizer for the iPhone that works with Spotify!</comment>


Samsung Gear VR now available for preorder - <AUTHOR> https://www.oculus.com/en-us/blog/samsung-gear-vr-now-available-for-pre-orders-at-99/ <comment><USER2> Oculus is in such a weird position right now. You can buy (well, pre-order, but at least it has a ship date) a version of their hardware right now, but it only works if you have a Samsung device, or are willing to buy one. Or you can wait for the "real" Oculus device, but all they'll say about when that'll come is "Q1 2016" \-- which isn't encouraging when Q1 2016 is, like, six weeks away. But Gear VR _is_ a real Oculus device, Samsung says. And then Oculus agrees... in a halfhearted way that makes you wonder just how much they really mean it.

None of which would matter much if they weren't heading into the prime shopping season of the year. But since they are... ugh.

</comment><comment><USER3> If I had to guess, they're using the Gear VR and the next few months to test out their services / infrastructure (such as Oculus Arcade, Oculus Video, Oculus 360 Photos, Oculus Social) before they do a much bigger launch of the highly anticipated Oculus Rift.

</comment><comment><USER4> They say "Galaxy Note 5 require a software update from your carrier."

I wish I knew if Verizon wil be offering this update. I just bought a Google cardboard type device (cheap!) for my Note 5 but the lenses are small so the experience ends up not being very immersive.</comment>


Ask HN: Why not link AND text? xor vs. or [meta] - <AUTHOR> It is becoming quite a common thing (specially with the Show HN) that when you submit a link you'd also submit a comment explaining more about what you made, relevance to HN, specific questions, etc. So my question is simple:
When you submit a story you can choose "url" or "text", but why not both? What are the disadvantages of just setting them as optional? (and requiring at least one)
Meta: I'd suggest changing the "or" to "xor" to display this mutual exclusivity. Or just leave it as "or" and allow both <comment><USER2> When links and text are available together, many of the top level comments will be direct responses to the original poster's text rather than discussion about the link itself.

By forcing the original poster to follow the same commenting rules, it greatly cleans up the top level discussion by confining direct responses to the poster in their own tree.

This is one of the reasons why the AskReddit subreddit moved to the headline- only model: It encourages people to share their own stories at the top level rather than respond to the poster, whose comment also needs to be able to stand on its own merits rather than get a default high visibility position.

<comment><USER3> Totally makes sense. Where can I learn more about things like that?

</comment></comment><comment><USER4> Have to be honest, I've wondered why Reddit doesn't do this. Seems like a simple change that'd open up a lot of possibilities for aggregators.

Especially given how many popular posts have a top rated comment by the author which does nothing more than explain stuff that could have been explained in the post itself (had the service given them room to do so).</comment>


Show HN: Huffgram helps you memorize data - <AUTHOR> https://www.huffgram.com/ <comment><AUTHOR> I made this a few years ago and use it to memorize phone numbers, API keys, etc. Here's a short writeup about how it works: [https://www.stephanboyer.com/post/111/huffgram-convert- strin...](https://www.stephanboyer.com/post/111/huffgram-convert-strings-into- memorizable-phrases)

Happy to answer any questions.

</comment><comment><USER2> > Huffgram will convert any piece of text into a word phrase that is easier to > memorize.

Huffgram for the above::

        what a glowing description of a dim and ill treatment
        of her him so think we were suddenly become the next day in
        your now he was more but some of thought creeping close
        attention was not help our part of those handsome sum had
        no bad so as for instance if bob or later when he ever will
        of of the avenger is there was in scribbling suit and that
        would not seem so fix like so the secret and laugh for a
        woman early an image of your seventeen already he wow"

</comment><comment><USER3> Would be much better if the text it generate is grammatically correct. My OCD is killing me.</comment>


In the Bay Area, technology has gone hand in hand with imperialism for 500 years - <AUTHOR> https://www.salon.com/2018/12/09/in-the-bay-area-technology-has-gone-hand-in-hand-with-imperialism-for-500-years/ <comment><USER2>
      n the late 18th century, the newly-arriving Spanish

That would make it the last _300_ years, not 500.

The Ohlone hadn't even employed the _wheel_ , let alone other technologies.</comment>


Ask HN: Tips, and resources on staying focus and engadged at work for dev - <AUTHOR>

How do you stay focus at work ? Have you done/read something that help you increased you ability to work longer hours and be better focus? Please share <comment><AUTHOR> Starting myself : 1 - Maybe the most effective way to worked longer for me is to take a nap during my break. My afternoon are so much more productive after a good nap. 2 - Not eating anything heavy during lunch

</comment><comment><USER2> Know when to knock off work. Dont stay up late to finish something. Take your evening off and forget all about computing.

When you start work the next day produce something of real value before you check your email, hn and the like.

I am very picky about my lighting. Not all people are but I am. I like to point several task lamps at the ceiling as well as away from me so I cannot see any glaring lightbulbs.</comment>


Dark crystals: the brutal reality behind a booming wellness craze - <AUTHOR> https://www.theguardian.com/lifeandstyle/2019/sep/17/healing-crystals-wellness-mining-madagascar <comment><USER2> _" According to Pew Research Center data...42% think spiritual energy can be located in physical objects such as crystals."_

Urgh. Is it really 42%? That's depressing.

<comment><USER3> Science education in this country is seriously broken. The elementary schools jump right into teaching procedural skills such as the 'scientific method', which is really just experimental design and is useless without teaching the basic critical thinking and philosophy of science. Those are way more important for the majority of students, who are not going to go on to be scientists.

We need to start the education at the preschool level, teaching the kids how easy it is to be fooled and why we have to always question. You can then build on that at the elementary school level with concepts of objectivity and repeatabily. By the end on elementary school you can then begin philosophy of rationality and empiricism. This can all happen in parallel to teaching 'science facts' that we do now.

But none of this is currently happening at any kind of large scale.

<comment><USER4> Here's an alternative view: we do an ok job of science education, but a deeply unsatisfactory job of teaching people that it is OK to be themselves and likes the stuff they like.

Crystal spiritualism to me is just a reaction to the fact that people overascribe meaning to things that they like and enjoy, as a justification beyond "I enjoy it". It goes hand in hand with the whole spiritualism around rock climbing or stand up paddle boarding or yoga - they're all great fun, but, modern life under emphasizes the role of fun and preference and so people justify it with spiritualism.

<comment><USER5> The simple counter-example here is: everybody likes cocaine, but nobody likes what cocaine does to you over long term, so you should probably not do it.

<comment><USER6> i don't particularly like what breathing oxygen does to my body over the long term but i still do that. granted, out of necessity.

anyway i agree with the point you're making, with one caveat: id say it's perfectly okay to do cocaine responsibly. it's just really difficult to do so evidently! if one were to only occasionally do it then i would think that the overall negative impact of it would be significantly less than what would happen if one were to do it consistently for a long period of time. granted my experience is with methamphetamine, which is, uh... an entirely different beast! but either way.

if someone likes crystals, it's ok to like them and to collect them, even to believe some woo things about them. but it's a problem when they start trying to convince everyone else that their experience is absolute and not just anecdotal. the same concept in a roundabout way goes for cocaine: if you like it, by all means, enjoy it! just be responsible with it. personally i don't like cocaine at all, just the way that it smells /s

</comment></comment></comment></comment></comment><comment><USER7> I predict that there will be a huge market in giant crystals, once rich people get hold of this.

Giant crystals are one of the few remaining truly scarce assets, more rare than that limited edition Bugatti, and unlike gemstones (diamond, ruby, ...) they have a novelty factor (not to mention sheer size).

[https://i.pinimg.com/originals/88/b9/04/88b9046dcf1acc04c63d...](https://i.pinimg.com/originals/88/b9/04/88b9046dcf1acc04c63da1be9e58a517.jpg)

</comment><comment><USER8> I was at a dinner party and I was talking to the man's wife about my work in software and cybersecurity.

She told me that she keeps a crystal next to the family computer to help protect it from computer viruses.

<comment><USER9> I would not know what to do in this situation.

Do you just shut down and walk away and find a different conversation to join?

Do you humor them?

Do you ruin your reputation and friendship by telling the man's wife "You're stupid"?

Do you try to politely explain something to someone who is dumb as a crystal and will probably never be able to understand?

Do you call 911 because they might be having a stroke or psychotic episode?

<comment><USER10> Sometimes just park your own view and genuinely ask questions and find out why she believes in them. Sometimes you can have an interesting conversation that way.

But most of the time, change the subject.

<comment><USER9> Thanks.

</comment></comment></comment></comment><comment><USER11> You'd think it would be an easy sell that unethically mined crystals harbor "bad" energy, not "good". But I guess nobody involved in sales & distribution actually cares about anybody's well being.

</comment><comment><USER9> A friend of my wife and I is into crystals and is also currently dating. She wants to find someone who is open about the magical powers of crystals.

Thinking of that made me realize that I would rather date someone on crystal meth than someone on crystals.

<comment><USER12> Reminds me of when I lived with my methhead landlord in college who was also really into crystal magic...

<comment><USER13> My previous roommates (a couple) were both big into xanax and crystals. They'd go on walks around the town, f'd up on scrips, and pick up rocks, bring them home and gush at me about what they found 'look at this ruby!" They'd proclaim (it was just a shard of broken beer bottle, I kid you not). By the end, the side of our house had accumulated a several hundred pound collection of (totally mundane and worthless) rocks. Their plan was to crack these rocks open by throwing them at the (concrete) ground, hoping to find crystals inside. In my opinion the benzos infantilized them, because they were not like that when they moved in, it was a downward spiral of a transition over two years.

This wasn't even thier worst quality. But I will always remember thier 'crystal' hobby.

</comment></comment></comment><comment><USER14> I don't buy into any of this stuff myself, but if it gives people comfort (or even a placebo effect), perhaps it's not all that bad?

<comment><USER15> The article does not mainly focus on whether the belief in crystals' healing powers is bad. It is mostly about the mining operations that bring the crystals out of the ground to begin with, and how a surge in demand for them is driving that mining.

</comment></comment><comment><AUTHOR> I was flabbergasted at the justifications of the sellers for supporting child labour in this industry:

>>If anything, Stamison wonders if the circumstances of miners in Madagascar, "makes the pieces a lot more special. Because I know some person in a little baby hut was actually polishing it by hand, and they're setting their intentions into it, too," she said. "People's intentions and people's energy are put into the stones as they're producing it."

"So the circumstances they're mined in, they are embedded into the stone somehow?" I asked.

"I think so. A little bit, it has to be. It has to be."

</comment><comment><USER16> The scientific version of this is mechanical watches.

</comment><comment><USER17> "Granola Crunchers or simply Crunchers, terms comprising the prototypical sandals, unrefined fibers, daffy arcana, emotional incontinence, flamboyantly long hair, extreme liberality on social issues, financial support from parents they revile, bare feet, obscure import religions, indifferent hygiene, a gooey and somewhat canned vocabulary, the whole predictable peace-and-love post- Hippie diction" -Brief Interviews with Hideous Men</comment>


CLiki : Benchmarks - <AUTHOR> http://www.cliki.net/Benchmarks <comment><USER2> For everyone interested: This is about Lisp.</comment>


Ask HN: Who is working on repairing nerves? - <AUTHOR> This is my first ask HN so, I hope I do it the right way.
For quite a time, I have found increasingly hard to accept that we still cannot repair nerves, leaving people prisoners of wheelchairs or worse. Conceptually, we are talking about something as basic as an electric wire being cut between a central computer and various types of sensors and actuators.
I have been explained that this is a very very hard problem, but I have also had a difficult time understanding who is actively working on this problem.
So, who in the world is working on this problem? Where do we find the cutting edge people? How can a hacker contribute to this?
Thanks! <comment><USER2> You've somewhat answered your own question, it's not so much a 'hard' problem as these are people and experimenting/hacking on them is not recommended. It's one of those things that needs to be done with the highest regard to ethics and patient welfare.

You could maybe contribute as a software developer by optimising the tantalum metal industry, medical equipment manufacturing or medical records and patient care in general?</comment>


Ask HN: Is it possible to do dead reckoning on the iPhone? - <AUTHOR>

Can I use the accelerometer to figure out how far the iPhone has moved from its original position?  I'm mostly interested in the scale of inches and feet.  If it's possible, how accurate would it be? <comment><USER2> [http://www.lmgtfy.com/?q=iphone+dead+reckoning&l=1](http://www.lmgtfy.com/?q=iphone+dead+reckoning&l=1)

decent answer in the second comment of the first google result.

Basically: There are 6 degrees of freedom: 3 translational, 3 rotational. To do dead reckoning you don't have enough information.

If you can get over that issue (fix the phone or something), then the main issue is gravity, if you misalign by 1 degree even after 30s you will be way off.</comment>


Scribblenauts Caught Being Accidentally Racist - <AUTHOR> http://www.escapistmagazine.com/news/view/94784-Scribblenauts-Caught-Being-Accidentally-Racist <comment><USER2> No -- it's not them being accidentally racist. It's you misunderstanding something you read, and _leaping_ to the conclusion that it's racism. This is a very common reaction, unfortunately, in recent months.</comment>


Hard-Wired to Hate Exercise? - <AUTHOR> http://online.wsj.com/article_email/SB10001424127887324432004578304250252788528-lMyQjAxMTAzMDEwOTExNDkyWj.html

<comment><USER2> What a crock of shhhhi!

Who are these scientists? Hard-wired to enjoy TV? Hard-wired to surf the net? Does it ever stop? Most people hate working hard. Be it studying, working out, working. The only things that make those enjoyable is a clear goal of why you are doing them. If not, it will be tedious and very hard.

I was hard-wired to hate exercise, until I stopped looking at exercise as a way to lose weight or be happy or have a nice body. It was simply a challenge of mind over my body, the same challenge that applies in studying for long hours, working for long hours, delaying gratification and subjecting myself to pain to reap the reward that lies ahead. Once I did that, I enjoy exercise. I skip ropes for the most part with light weight training. I skip for an hour straight. My weight drops 2-3 lbs everytime I enter the gym. I work that hard because I have a reason. Prior to this? I walked around the gym for an hour touching and sampling various equipments. What I now do in a single day is greater than what I use to do in an entire week. Tbe great thing about humans? We are hardwired to change anytime we make the choice. That's the only thing we need to remember.</comment>


Show HN: Simple, opinionated task list for developers - <AUTHOR> http://blncd.io <comment><AUTHOR> Hi everybody,

I made this while playing around with node and react. It's free, open source, allows anonymous registration and has an open api.</comment>


Death by 1k Clicks: Where Electronic Health Records Went Wrong - <AUTHOR> https://khn.org/news/death-by-a-thousand-clicks/ <comment><AUTHOR> Very long article - but a good read. It's incredible to me how incentives misaligned in such a way that essentially every medical practitioner in the US has to use software they hate.

Anecdotally via the physicians I interact with its' terrible how much time it takes away from actual patient care, when the promise was the opposite.</comment>


Razorfish's site background is a low-res, live, web-cam feed  - <AUTHOR> http://www.razorfish.com/

<comment><USER2> Cool effect, but I don't think it's live. I find it hard to believe that every single office has someone walking around in it 24x7 (which always seems to be the case).

<comment><USER3> Agreed. I'm thinking that they just have different sets of videos for different local times of the people viewing the page.

Awesome though.

</comment></comment><comment><USER4> Pretty cool. Anyone know how they're doing this, exactly?

<comment><USER5> Since I get a blank white page in Chrome, I'm going to guess Flash has something to do with it...</comment></comment>