
Ask HN: Why is telephone “hold music” still such poor quality? - justinmolineaux
Even on new devices, telephone hold music seems to be rife with static&#x2F;noise. Just my experience? What&#x27;s holding this back?
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PaulHoule
Telephone hold music is never going to sound that good, at least not if you
are listening to it through the narrow bandwidth of a analog phone line or the
highly compressed signals used for cellular and VoIP.

~~~
Doxin
The average hold music definitely sounds worse than it just being band-
limited. There tends to be a fairly heavy noise component too.

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eliaspro
Relevant Tom Scott's "Things You Might Not Know"-Video: "Why Hold Music Sounds
Worse Now" \- [https://youtu.be/w2A8q3XIhu0](https://youtu.be/w2A8q3XIhu0)

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BjoernKW
The codecs and sampling rates used for phone communication are optimised for
the usual range (E2 to C4 depending on voice type) and overtones of the human
speaking voice. The usual sampling rate for phone lines is 8 kHz, which means
that only frequencies up to 4 kHz are captured. This is sufficient for making
vocal communication intelligible (if not exactly aesthetically pleasing).

Music typically goes beyond that range. When transferring music over a phone
line overtones in particular are cut off, which makes the music sound flat and
creaky.

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cadr
I've never understood why they didn't just make music that worked under the
constraints.

~~~
Something1234
Because that costs money and might be more grating. It might not improve the
customer experience much at all.

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hodl
What I also notice in Australia is when that music plays the volume slowly
tends to zero. I worry that I've lost reception, but when they pick up its
fine. I also notice the phone seems to "know" its on hold as there is an
audible cue, so perhaps the carrier is deprioritising on hold audio.

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refurb
A better question is why telephone quality is bad in general. I assume it's a
bandwidth issue related to the technological limitation when it was rolled
out.

Whenever I do a FaceTime voice call I'm always shocked by the quality of the
audio.

~~~
Slansitartop
IIRC, the cell phone voice codecs are at an absurdly low bitrate (1.2kbps?)
and _highly_ optimized for human voice _only_ , so voice is acceptable and
everything else sounds like trash.

Analog landlines used 64kbps codecs (after the analog loop to your house) that
were flexible enough to be abused to transport 56kbps of data.

I'm pretty sure nothing else lower quality codecs than cell phone voice.

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thewizardofaus
Limited bandwidth. Line is optimised for voice comms frequency.

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pcunite
It maybe conversion from the original MP3 format to the GSM, ULAW format some
PBX VoIP systems use.

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idunno246
the reason there is hold music at all is to let you know the phone didnt
disconnect. Cheap sound fulfills this so why spend money doing anything better

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Rjevski
The problem is that unlike the Internet which is lossless, the telephone
network is lossy and there is no standard as to how much loss is acceptable.

Even if all your equipment and your telco's equipment is perfect (which it
isn't), all it takes is one single weak link in the call chain (from a cheap
carrier somewhere) to completely slaughter the audio quality by passing it
through some bad equipment or even analog equipment.

The telco industry is the total opposite of the Internet industry - in the net
most of the technologies, standards, etc are open, there is mostly a culture
about sharing and openness, and while proprietary solutions exist, they are
often much better as far as complying with standards goes, and open-source
solutions are competitive as well (any Linux or BSD box can be used as a
router).

The telco world is the opposite, lack of openness, the standards, if they
exist, are often under NDAs, lots of security by obscurity, a lot of scammers
selling proprietary gear ("magic boxes" as I call them) that respect the
standards to the bare minimum or sometimes don't even try (which leads to lots
of fun when two different magic boxes both claim to support the standard and
yet fail in mysterious ways when trying to interoperate). An example would be
a magic box that only supports G711 (a crappy low-bitrate codec), so even if
the call chain is full IP and all the other boxes support better codecs (or
even yet, just handle SIP singling and let the peers talk directly via IP),
the call quality would be limited by that one magic box that insists on G711.
The lack of openness and knowledge prevents people from making an informed
decision and they have no choice but to trust the salesman that their magic
box is best. Open source solutions aren't up to scratch either, Asterisk (and
derivatives) is a start in the right direction but frankly doesn't really
scale and has problems when you try to make it highly-available.

I wish the telco world would just die and we move to SIP with media directly
over IP, that would eliminate all those problems. SIP is already used within
VoLTE and Wi-Fi calling but a magic box within the carrier's network (the
P-CSCF as they call it) will often still insist on re-encoding the media
stream instead of letting the packets flow between the two phones directly, so
you're still at the mercy of whatever codecs that box supports. Calling
outside your carrier will involve another magic box that will bridge your SIP
call to the other carrier's network, via legacy garbage E1/T1 links with
crappy codecs (instead of you know, just using SIP and letting the packets
flow directly).

If you have good equipment throughout the entire call chain, that ideally does
not interfere with the media itself (so the RTP stream is direct between the
two phones), and the phones both support a good codec (Opus?), you could have
MP3-quality calls.

