VoRS: Vo(IP) Simple Alternative to Mumble

http://www.vors.stargrave.org/

by stargrave • 2 years ago

86 71 2 years ago

vors.stargrave.org

speps• 2 years ago

It mentions WebRTC but this project wouldn't work in browsers as it's a Go CLI tool. Is there any alternative to WebRTC nowadays? How far along is WebTransport? Anything else that could rely on older web tech?

Karrot_Kream5• 2 years ago

The project just mentions WebRTC as bloated, it's not WebRTC compatible at all. WebTransport isn't there yet so browsers are hamstrung by not having access to real sockets.

ekr____• 2 years ago

Of course, WebTransport also isn't real sockets. It's more like WebSockets but with more modern transport semantics such as unreliable data and multiplexing without head of line blocking (as long as it's run over QUIC).

For security reasons, browsers don't allow Web sites access to raw TCP or UDP sockets.

jacknews• 2 years ago

This is great but would need to run on windows for the typical mumble use-case of gaming.

It would also be great to have this running on ESP32 or similar, so you could make dedicated IP desk intercoms - I envisage a star-trek style intercom, with each button being a channel that you can join by pushing it in (can join multiple simultaneously).

account423• 2 years ago

This is great but would need to run on windows for the typical mumble use-case of gaming.

You can game on Linux just fine.

mtndew4brkfst• 2 years ago

You can certainly play a number of popular and/or high quality games on Linux, but it's hardly ubiquitous. If you're interested in a given specific game I would say your odds of it having a viable Linux experience are worse than 50/50 yet, and if it's a AAA title I'd say it's closer to 10-25% IME.

If not for Valve/SteamDeck this would be even more dire, and many of those experiences are still emulation-based i.e. Proton.

If you want to be able to play any given computer game being sold this year, that's only reasonably certain on Windows. Gaming is the only reason I still have a Win10 install.

cl3misch• 2 years ago

This reminds me of a ESP32 Walkie-Talkie: https://www.reddit.com/r/esp32/comments/mbwq6f/walkietalkie_...

numpad0• 2 years ago

It's so nerd-sad to see Ukrainian Army doing just fine with Discord screen sharing on Windows[1], realistically obsoleting a large bulk of proper intercom solutions. I want to have to have a working RTS KP-32 on my desk!

1: https://i.redd.it/hjcg9hdnqvqa1.jpg

yvely• 2 years ago

I'm more concerned about the winrar in that photo. Someone please tell them about 7z

account423• 2 years ago

It's clown world level to see any military using a consumer chat application for sensitive operations, especially one made by a foreign cooperaton. They really don't seem to take opsec seriously even when leaks have already cost lives.

squarefoot• 2 years ago

It would also be great to have this running on ESP32 or similar

ESP32 has its own port of the Codec2 library, which allows intelligible communications using very low bandwidth (a few kb/s). It could make the ideal solution for creating small cheap intercoms scattered around a small area using WiFi, or employing very low bandwidth radio connection such as LoRa for wider coverage. I still didn't see any real application, beside a few simple proof of concept videos on YouTube, though.

More info:

http://www.rowetel.com/?page_id=452

https://www.arduino.cc/reference/en/libraries/esp32_codec2/

AshLeece• 2 years ago

Anyone able to find the source code? Can't see it listed anywhere

subbz• 2 years ago

It's here: http://www.vors.stargrave.org/Install.html

Sammi2• 2 years ago

I see the source is in the tar.zst file and there is documentation for uncompressing it.

Only I spent minutes looking for the git repo link... There is none?

Karrot_Kream5• 2 years ago

I wonder if there are plans to offer a "headless" mode where call-status is streamed to an FD instead of using a TUI. I'd love to be able to compose/wrap this tool outside of a terminal emulator.

I've been idly hacking on something quite similar but using QUIC instead. One of the downsides with QUIC of course is that it assumes TLS by default which makes it inefficient for use on encrypted transports (like wireguard, zerotier, tailscale, or yggdrassil.) I'll definitely give this a go as I'm a big fan of NNCP, another project by the author.

cies5• 2 years ago

This is REALLY important software nowadays imho.

I'm old enough to have spoken to people on analogue land lines: the sound was crisp, you could hear small background noises, you could hear someone breathe.

Nowadays we usually speak to people on digital lines that are highly compressed (to the extend that is messes with the sound quality), low freq range (no bass, very high sounds) and cut up (without enough sound or when then other party makes more sound the stream is completely interrupted).

And it does not have to be like this! All of this is in favour of the network operator (or centralized chat servers e.g. whatsapp) trying to save some data/money. While many of us have paid for unlimited data!

On top of that much of the conversations are not properly e2e encrypted!

I've used Mumble to speak to people I love over long distance and the quality is just so much better: it's like the analog experience of my childhood. Hearing ever breath, background noise and all in high quality makes all the difference some times.

cies5• 2 years ago

Cool project anyone reading this may be interested in:

https://github.com/Johni0702/mumble-web

I've never used it but it should make having a p2p conversation through Mumble as easy as pointing your browser to some URL. UX matters (Mumble clients, including mobile apps, are not very user friendly last time i checked: they require some level of skill to use them)

Unmaintained for the last 4 years, sadly.

thaumasiotes2• 2 years ago

I'm old enough to have spoken to people on analogue land lines: the sound was crisp

You're old enough to have forgotten what land lines sounded like.

They intentionally dropped frequencies from the audio signal so that they wouldn't have to carry the data contained at those frequencies. This is why nobody ever sounded like themselves over the phone.

cies5• 2 years ago

They intentionally dropped frequencies from the audio signal so that they wouldn't have to carry the data contained at those frequencies. This is why nobody ever sounded like themselves over the phone.

I remember switching from early GSM phone to a landline because I like to hear my love breathe

Sammi2• 2 years ago

Old phone signals (Plain old telephone service or POTS) cut off frequencies above 4 khz. You cannot hear the difference between f and s, as these are higher frequency sounds at around 8 to 12 khz. You'd have to say f like in Fred or s like in Steve? Cause you literally could not hear the difference.

This is the reason broadband internet (ADSL) is called broadband. Because dial up internet used the POTS frequency band below 4 khz, and broadband used the (broad) frequencies above.

It's also the reason you'd have to get a technician out to your house in order to get ADSL. They would install a frequency cutoff filter on the phone line into your house, and hookup the landline to the below 4 khz side and the ADSL modem to the above 4 khz side, so the POTS signal would not interfere with the ADSL signal.

lhamil64• 2 years ago

I wonder if it was more of an issue with the mic and/or mic placement on phones? Old landline phones had a large mic right by your mouth, whereas cell phones have little pinhole mics near the bottom of the phone.

I feel like I've actually noticed the opposite. My parents still have a landline, although it's basically VoIP (through their cable company) but it's connected to the analog lines in their house. I've noticed they generally sound clearer on their cell phones than they do on the landline.

lxgr12• 2 years ago

How old of a landline are you talking about here?

Anything newer than (heavily depending on the country, probably) the 70s or 80s or so would have been very likely PCM u-law or a-law at 64 kbps (i.e. 4 kHz audio bandwidth at 8 bit), which is literally a mandatory codec in WebRTC.

It would have to be a really old, purely analog baseband line without filters (maybe a local call between offices), frequency modulation etc. to preserve more than the typical 4 kHz of audio bandwidth you'd get on these. Inter-trunk connections were often frequency multiplexed to fit more channels onto a physical wire, which also limited them to 4 kHz.

Today, 64 kbps gets you much farther using a modern codec like Opus. WhatsApp sounds better than any landline or native mobile phone connection I've ever used in my life.

(or centralized chat servers e.g. whatsapp) trying to save some data/money

WhatsApp uses P2P for (non-group) calls if at all possible.

There's also a "save data for calls" option in the settings which is off by default.

Modern codecs are so good, adding even more data would literally not make any discernible difference. A sizable fraction of all data transmitted/received by modern VoIP is IP and UDP framing overhead.

aftbit3• 2 years ago

Mobile connections are often especially crap, using something much lower bit rate than G.711 µlaw. Even modern landlines use voice activity detection and comfort noise generation instead of just passing through the background sounds.

I've never used WhatsApp to call. I have used decent SIP connections with G.722 Wideband or OPUS. They sound better than the old landlines. Discord sounds better too. Signal, much worse.

I think often the problem is that cell phones really have crappy speaker and microphone placement for calls, as basically nobody actually makes calls on them anymore.

uyzstvqs• 2 years ago

users tend to complain about [Mumble's] newer client versions quality and convenience.

I have never heard anyone complain about Mumble's UX. Especially not that it has gotten worse. Or it's resource usage, which is practically unnoticeable on any average system sold past 2005. It's the best-in-class and has only gotten better.

thfuran3• 2 years ago

And even if I had, I'm quite sure they wouldn't have meant they wanted it in the terminal instead.

tarxvf• 2 years ago

Well cover me in foil and charge extra, I guess I'm rare.

Why would I ever want a GUI for a voice application?

upofadown• 2 years ago

...its server side is still written on Qt, which requires hundreds of megabytes of additional libraries to build it up.

See:

https://github.com/umurmur/umurmur

fionaellie• 2 years ago

unrelated, but I wonder if anyone can identify the terminal font used in the screenshot? I am looking for a good (free) serif monospaced font.

lagniappe• 2 years ago

That's Go Mono https://go.dev/blog/go-fonts

BuildTheRobots• 2 years ago

The _biggest_ selling point with mumble (though I've not used it in anger in about a decade) was the way it tied in with the game you were playing and gave you in-game spacial audio.

Stand next to someone in game and they came through loud and clear. Move away from them sideway and it's now quieter and only through one headphone... I found it extremely novel and immersive when I used to play FPS games; everyone on mumble but absolutely not like being in a chatroom.

lxgr12• 2 years ago

WebRTC-based solutions are insane bloated incredible monsters [...] They work mainly only if you use the same kind of software and codecs, for example Chromium[...]

That's just not true anymore these days. I've used WebRTC between Firefox, Safari, and Chrome without any issues.

only if you use the same kind of software and codecs, for example Chromium, that requires dozens of gigabytes of disk space and much RAM, CPU time to build it.

There seem to be non-browser implementations [1] [2], although I can't vouch for their quality.

[1] https://liburtc.org/

[2] https://github.com/awslabs/amazon-kinesis-video-streams-webr...

Sean-Der3• 2 years ago

libdatachannel[0] is exciting me the most for 'low footprint'. str0m[1] is especially exciting, but you have to be in a device/ecosystem that allows rust.

only if you use the same kind of software and codecs

With WebRTC I can connect clients with varying support of H264, VP8, VP9, AV1, Opus, ulaw and alaw. WebRTC has plenty of flaws, but 'lack of negotiation' is one of them.

[0] https://github.com/paullouisageneau/libdatachannel

[1] https://github.com/algesten/str0m

lxgr12• 2 years ago

Oh, this looks great! Also thank you for Pion :)

Edit: Woah, I somehow completely missed that Pion does WebRTC media too (I've only seen it in a data channel project I've toyed around with in the past). Definitely add that to my list above, and I can somewhat vouch for it, it's cool!

Karrot_Kream5• 2 years ago

I'm generally not a person who rails at bloat, but WebRTC is definitely a package. It's a suite of SIP, SRTP, ICE, STUN, TURN, and codecs to do voice/video calls. This makes it versatile and seamless but yes also heavy and opinionated. A good example is just how few blessed ways there are to use WebRTC outside of the browser.

Sean-Der3• 2 years ago

What makes something a 'blessed way'? Are you looking for community support, large corporate users etc..? Not being snarky, curious about how people see the space. Lots of implementations exist and I have used most of them for different projects, not all these projects went to production though.

* https://github.com/elixir-webrtc (Elixir)

* https://github.com/pion/webrtc (Golang)

* https://github.com/webrtc-rs/webrtc (Rust)

* https://github.com/algesten/str0m (Rust)

* https://github.com/sepfy/libpeer (C/Embedded)

* https://github.com/awslabs/amazon-kinesis-video-streams-webr... (C/Embedded)

* https://github.com/paullouisageneau/libdatachannel (C++)

* https://webrtc.googlesource.com/src/ (C++)

* https://github.com/shinyoshiaki/werift-webrtc (Typescript)

* https://github.com/sipsorcery-org/sipsorcery (C#)

* https://github.com/aiortc/aiortc (Python)

* GStreamer’s webrtcbin (C)

d-z-m4• 2 years ago

You have to verify downloaded tarballs authenticity to be sure that you retrieved trusted and untampered software.

I'm not able to access the site over TLS, so this is currently impossible. Anyone else have better luck?

    (from the installation instructions)

    $ [fetch|wget] http://www.vors.stargrave.org/download/.   vors-2.3.0.tar.zst
    $ [fetch|wget] http://www.vors.stargrave.org/download/vors-2.3.0.tar.zst.sig
    [verify signature]
    $ tar xf vors-2.3.0.tar.zst

Guarantees nothing, if you're actually being attacked. You can't serve out the tarball and the public key[0] and the signature insecurely and get any guarantees about authenticity.

[0] http://www.vors.stargrave.org/PUBKEY-SSH.pub

someguydave• 2 years ago

I can see your point about a man-in-the-middle attack on HTTP, but realistically if the the attacker can do the MITM he can probably also figure out how to modify the contents on the webserver, in which case TLS does nothing

d-z-m4• 2 years ago

realistically if the the attacker can do the MITM he can probably also figure out how to modify the contents on the webserver, in which case TLS does nothing

Sorry, I don't see how that follows. If you had access to the webserver you would never even attempt a MITM attack in the first place.

ranger_danger2• 2 years ago

Most websites these days use Crimeflare or equivalent already anyways which is a MITM proxy, so you're still not safe there either.

d-z-m4• 2 years ago

Yes, however there's a big difference between Cloudflare and an arbitrary MITM. If you think Cloudflare is mangling traffic in a malicious way you would need a shred of evidence to substantiate that claim.

I would also push back that "most websites" use a 3rd party TLS terminating CDN/proxy layer in front of the actual webserver.

aftbit3• 2 years ago

No GUI requirement. Why would someone need a GUI for voice application? But a fancy real-time refreshing TUI would be desirable. Mumble tends to output no information, sometimes hiding the fact of a problem and that everything stopped working.

Eh why would someone need a fancy real-time refreshing TUI for a voice application? Just write logs to stderr and status updates to stdout like regular people.

Honestly, go a step further. Take microphone audio in on stdin and produce speaker audio out on stdout. Then you can skip ALSA/OSS/jack/pipewire/pulseaudio support and just leave that to the user to compose.

rnhmjoj• 2 years ago

You've practically described baresip.

kimi• 2 years ago

I'm not sure I understand the reason why; for non-browser, you can set up a FreePBX or something in 20 minutes and it will work with any softphone. But if I was to use a chat today, I'd use something in-browser, so based on WebRTC and incidentally peer-to-peer (well, kinda, but voice/video flow will be). So not sure I understand who/what this is for.

VoRS: Vo(IP) Simple Alternative to Mumble

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