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Sean-Der
searching PlanetScale…
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24 ms
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391.
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by
Sean-Der
3y ago
This article is incorrect about WebRTC. I don’t know about other protocols and what they offer. * Clock Recovery I have had no problems with measuring NTP drift. As the clocks change I would measure. * Common Clock for Audio and Video Sende
392.
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Sean-Der
3y ago
I would agree with that. Chrome's WebRTC implementation has the most specification coverage of any implementation. What is your goal? Do you want something that is easier to build, more performant, $x?
393.
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Sean-Der
3y ago
This is why I am personally excited. * Simulcast - WebRTC has the concept of uploading multiple quality levels baked in. This means they don’t need to be generated server side anymore. Huge savings on cost/complexity for broadcast serv
394.
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Sean-Der
3y ago
kabes description is excellent. It just gives senders lots of info about receivers. Since it is WebRTC all the way through this is possible. You can read about it in more detail https://webrtcforthecurious.com/docs/06-m
395.
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Sean-Der
3y ago
I don't think that will ever be possible unfortunately. WebRTC requires DTLS+SRTP for the mandatory security, don't ever see that being made optional. For RTSP/WebRTC bridging I have used https://github.com/Al
396.
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Sean-Der
3y ago
Sorry about that! Fixed the AWS link, got truncated on phone.
397.
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Sean-Der
3y ago
I tried to do this with OBS. The code worked really well. It was so stable and performant. Life is significantly better not debugging SEGV. However it puts too much burden on the maintainers. So much work goes into packaging etc… I don’t ha
398.
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Sean-Der
3y ago
Lots of other WebRTC implementation exist (in many languages). Happy to help if you have any questions * https://github.com/aiortc/aiortc (Python) * GStreamer’s webrtcbin (C) * https://github.com/shinyo
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Sean-Der
3y ago
I am really excited for this PR to land. Being able to use WebRTC for Broadcast and Playback is going to be huge. Instead of doing fixed interval keyframe we can use receiver feedback (Massive reduction in bandwidth) Instead of server side
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WebRTC support being added to FFmpeg
(github.com)
201 points
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Sean-Der
3y ago
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28 comments
401.
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Broadcast to Twitch in 50 lines of JavaScript. No libraries or software required
(jsfiddle.net)
4 points
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Sean-Der
3y ago
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0 comments
402.
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Sean-Der
3y ago
If Google’s WebRTC implementation is the only one that works how does anything WebRTC work? The majority of WebRTC media calls aren’t P2P. What does Google WebRTC do that you can’t find anywhere else? What do you think would be better then
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Pion WebRTC v3.2.0 Released
(github.com)
5 points
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Sean-Der
3y ago
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1 comments
404.
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Tell HN: Twitch.tv now supports WebRTC ingestion (Broadcast from your browser)
2 points
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Sean-Der
4y ago
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5 comments
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Sean-Der
4y ago
The lack of collective 'internet user' ownership makes me sad. Many people want to improve RTMP, but an open discussion doesn't exist. If RTMP was in the IETF my comment wouldn't apply anymore.
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Sean-Der
4y ago
You wouldn’t open ports, you are in the same network. I don’t think P2P is the answer to everything, in the same manner servers aren’t the answer to everything.
407.
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Sean-Der
4y ago
You don't have a server (and in my cases you don't want) * I don't want to upload my video from my security camera to a remote host. It has security and privacy implications * I don't have the bandwidth available * I don
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Sean-Der
4y ago
> Does codec negotiation really make sense On startup you could be connecting two clients that have completely different sets of codecs. What if you have a low powered device that only has a H264 hardware decoder? What if you have two de
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Sean-Der
4y ago
Project leads created the original PR + RFC * PR https://github.com/obsproject/obs-studio/pull/7192 * RFC https://github.com/obsproject/rfcs/pull/43 I tried to keep the size as
410.
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Sean-Der
4y ago
I use WebRTC today and am happy with it! Works great for broadcasting from Web, Mobile and tools that support it (GStreamer) I have a PR open to OBS for it right now https://github.com/obsproject/obs-studio/pull&#x
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Sean-Der
4y ago
I am heavily conflicted on this. I am happy that people get access to better codecs. Better compression and those that aren't encumbered with patents/licensing (AV1) I am sad that we are extending RTMP instead of adopting somethin
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Sean-Der
4y ago
Yes this is a real problem! I reported it here https://github.com/rtcweb-wg/mdns-ice-candidates/issues/121 I am not able to use my reproduce on Chrome/FireFox/Safari anymore. Not sure how they are d
413.
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Sean-Der
4y ago
Yea! WebRTC supports TCP candidates, so webrtc-rs can do the same. If that unblocks you happy to implement. Getting people with interested use cases involved in the project helps a lot.
414.
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Sean-Der
4y ago
Would it help if I added the ability to gather on 127.0.0.1 to webrtc-rs' SettingEngine? That exists in Pion, but landed after webrtc-rs was started.
415.
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Sean-Der
4y ago
Would you mind explaining more the issue you are having with WebRTC? It sounds like you want playoutDelay (no latency) added by the receiver? You could also use insertable streams and modify/drop frames with some arbitrary logic.
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Sean-Der
4y ago
Does anyone know if any deals/exemptions exist for Open Source? Don’t want all my images for github.com/pion/webrtc to not be available, but paying is a bit much.
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Hmm
(github.com)
2 points
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Sean-Der
4y ago
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0 comments
418.
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Sean-Der
4y ago
What do you find impenetrable about WebRTC? In most cases I have found the complexity necessary. When you find yourself in unique situations it’s a protocol that gives you the knobs. I created https://webrtcforthecurious.com to
419.
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Sean-Der
4y ago
You can run all your WebRTC traffic off a single port. You use the remote 3 tuple (IP, Port, Protocol) to demux traffic. https://github.com/pion/webrtc/tree/master/examples/ice-sing... is one examp
420.
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Sean-Der
4y ago
The third scenario is fixed with https://datatracker.ietf.org/doc/html/rfc8260 This just needs to be implemented in libwebrtc’s SCTP implementation. It isn’t implemented in Pion either I believe?
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