WebRTC 1.0 telephony app using “hifi” Opus codec with up to 500kbs bitrate

Mumble uses Opus by default. You can definitely tell the audio sounds much better than competitors although mumble is more fussy about mic and background noise as it does less processing than Skype for example.<p>Opus was originally intended to be a general purpose codec that maintained fidelity at even low bitrates which makes it suitable for telephony as it&#x27;s also low latency. On the other hand it also sounds good at higher bit rates and I use it as my codec for my music when I convert them from flac for mobile devices.

Timur is a really fun guy. He also wrote and maintained the (since quite famous) kernel patches and userspace utils for Nexus7 devices to allow USB host mode and charging simultaneously. The hack itself was modest enough but it since enabled a whole new world for DIY in-car entertainment enthusiasts, providing the modding community with a simple and affordable way to fully integrate a working tablet into their cars with USB DACs, DAB dongles, back-up cameras, etc - which was pretty slick stuff back in 2012-2013.

I am the developer of this web-app and you can use the link (say, for the next hour or so) to give me a direct voice call. I am interested in finding out about audio quality in the real world. Thanks

I had no idea what I was getting into when I clicked &quot;Call Now&quot; but wound up having a great conversation with a new friend-- Timur! The audio quality was excellent. This project has great potential. Best of luck to the author!

Opus at over 40 or 60Kbps for voice sounds like overkill, nevermind 500kbps. What&#x27;s the rationale for using such high bitrates?

I made a lot of phone calls tonight. Two quick observations: 1) Many people think I&#x27;m a bot. And they want to test me with smart questions in order to find out if I&#x27;m a bot. How do you prove that you are not a bot? 2) It seems that aprox 90% of the calls are getting relayed. I expected more calls to come through as P2P. The audio quality is still excellent in most cases. If both sides use a headset, it is almost better than sitting next to each other. This is much better than normal telephony.

Ooh... reminds me of a throwaway experiment I did a few years back - WebRTC for radio outside broadcasts. It never went far as we ended up going down the SIP path instead. Nice to see the idea going further though there are commerical products now that work on this basis (IPDTL is one that rings a bell).<p>Both approaches were a great improvment over the need for an ISDN line but you could run into problems running higher bitrates on &quot;spotty&quot; networks. Thankfully you didn&#x27;t need more than 128&#x2F;192kbps for broadcast quality most of the time and could get away with even less for voice.<p>Call Server: <a href="https:&#x2F;&#x2F;bitbucket.org&#x2F;marcsteelesoftware&#x2F;g-rtc-call-server&#x2F;src&#x2F;master&#x2F;" rel="nofollow">https:&#x2F;&#x2F;bitbucket.org&#x2F;marcsteelesoftware&#x2F;g-rtc-call-server&#x2F;s...</a> Frontend: <a href="https:&#x2F;&#x2F;bitbucket.org&#x2F;marcsteelesoftware&#x2F;g-rtc&#x2F;src&#x2F;master&#x2F;" rel="nofollow">https:&#x2F;&#x2F;bitbucket.org&#x2F;marcsteelesoftware&#x2F;g-rtc&#x2F;src&#x2F;master&#x2F;</a>

Tried to call a few times and got the busy signal. Just a thought but I would maybe not log the number in the console as it might be leaking your actual phone number to the wider world. Hope it&#x27;s a burner number.

Adding a &quot;I am not a robot&quot; button to initiate the call would be really cool way to avoid bots in the future if this ever catches on.

&gt; It works best in Chrome, Chromium and Firefox (all v80+).<p>I&#x27;m assuming Edge also?

Is there any way this could be integrated with Twilio Voice?