I sent a letter to the Soviet Union asking if they were ok with me using the name, never heard back!
4,710 karma · joined November 18, 2012
Socials: - github.com/Sean-Der - linkedin.com/in/sean-dubois - siobud.com - pion.ly
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I sent a letter to the Soviet Union asking if they were ok with me using the name, never heard back!
Then I am all done fixing WebRTC bugs for life :')
If you get a chance to try it out would love to hear your thoughts! Should be available soon in pion/dtls@v4 :)
What motivates you, certain topics or building things in a certain area? what are you most interested in?
I am underwater dealing with it. I am getting outlandish PRs against Pion. So I spend my time triaging and feel burned out. The 'AI future' I was expecting would be people swooping in and clearing up the backlog in high quality ways.
Instead I get people trying to add new APIs when it can already be done with existing APIs and then arguing about it with me.
I wonder how carpenters felt when powertools started taking off?
When people are learning WebRTC they are really confused by the Offer/Answer. You have this big blob of text and no idea what it actually does. You look up values one-by-one and forget as you go. its annoying
Excited that the next generation learning WebRTC (for VoiceAI) has better tools.
Self-hosting is pretty these days with docker. Exposing it to the internet is the annoying part. I do Jellyfin via netbird, but that means I don't really share it with my friends. I hope something like this getting baked into self-hosting flows is killer.
My other pet peeve is that people use 'Cloud Services' for File transfer. Sometimes when they are in the same LAN!
Nice work again and I hope it catches on/people understand how great this really is :)
Accepting 'Big Changes' from people is VERY frustrating. These thoughts run through my head.
* Idea is usually good! Even if I don't understand it could help lots of others users.
* The contributor is very focused on just getting their feature in. The impact on the larger project isn't as much a concern.
* New contributors often don't have the grit to see it out. They will disappear before things are done. So I am left picking up the pieces (which is harder then doing it all myself)
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What I try and remember is that their happiness/experience matters more then any code. I try to help the contributor learn/grow as much as possible and even see some career benefits out of it. Pion will cease to matter eventually, so I hope to help as many programmers with it as possible.
I am grateful the code got written, but even better people got careers out of it/learned new stuff. If you are on the fence about taking on a project I encourage you to do it!
I added this in Pion here[0] and I remember testing against Chrome + FireFox and it seemed to work great!
[0] https://github.com/pion/webrtc/commit/e4ff415b2bff31382bdb80...
If https://github.com/pion/sctp/issues/12 had happened (not just in Pion but across all implementations) this could have been fixed years ago. The hardcoding we all settle for is tragic.
I hope this site can convince more people to check this stuff out. If you are curious or have feedback I would love to hear.
I hope this site can convince more people to check this stuff out. If you are curious or have feedback I would love to hear
After that I want to spend the weekend just closing out Pion bugs/relaxing :)
The 'co-watching/co-streaming' is the best. I love watching a crappy horror movie with friends and bantering.
I could get it lower by encoding once and then syncing to keyframes. It would make the code more complicated though. If someone asks for it/gets excited would love to do it though :)
I should reword the README though. If someone is savy enough they could totally grab the video. For most users it is like a Google Meet though. If you click `Show Controls` you can pause the video that is it.
With things like Insertable Streams[0] you can totally grab the video.
[0] https://developer.mozilla.org/en-US/docs/Web/API/Insertable_...
In this project I don't set anything though.
[0] https://webrtc.googlesource.com/src/+/refs/heads/main/docs/n...
2.) The model isn't processing just text anymore. Also taking into account breathing/emotion etc... not just spitting out big responses anymore. As it generates them it is taking into account the users response.
3.) It works with the LB setup today. Clients are sending ICE traffic, if it roams we lookup the ufrag and route appropriately.
4.) With DTLS 1.3 it is 1 RTT with SNAP[0] for WebRTC session. SCTP info goes in Offer/Answer, DTLS is packed into ICE. You are totally right about signaling though! [1] was my answer for doing WebRTC without signaling, couldn't get anyone to care though.
5.) I don't have anything that I need to tune. If I want to increase (or decrease) latency [3] is something I put into Transceiver. Otherwise I can't think of any 'change this WebRTC behavior' that has been asked by users/developers.
[0] https://datatracker.ietf.org/doc/draft-hancke-tsvwg-snap/
[1] https://github.com/pion/offline-browser-communication
[3] https://webrtc.googlesource.com/src/+/refs/heads/main/docs/n...
[0] https://developers.openai.com/api/docs/models/gpt-realtime-t...
> …but as a user, I would much rather wait an extra 200ms for my slow/expensive prompt to be accurate
This is the opposite of the feedback I get. Users want instant responses. If you have delay in generating responses/interruptions it kills the magic. You also don't want to send faster than real-time. If the user interrupts the model you just wasted a bunch of bandwidth sending 3 minutes of audio (but only played 10 seconds)
> TTS is faster than real-time
https://research.nvidia.com/labs/adlr/personaplex/ Voice AI for the latest/aspirational is moving away from what the author describes. It is trickled in/out at 20ms
> We really hope the user’s source IP/port never changes, because we broke that functionality.
That is supported. When new IP for ufrag comes in its supported
> It takes a minimum of 8* round trips (RTT)
That's wrong. https://datatracker.ietf.org/doc/draft-hancke-webrtc-sped/
> I’d just stream audio over WebSockets
You lose stuff like AEC. You also push complexity on clients. The simplicity of WebRTC (createOffer -> setRemoteDescription) is what lets people onboard easily. Lots of developers struggled with Realtime API + web sockets (lots of code and having to do stuff by hand)
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I think if I had my choice I would pick Offer/Answer model and then doing QUIC instead of DTLS+SCTP. Maybe do RTP over QUIC? I personally don't feel strongly about the protocol itself. I don't know how to ship code to multiple clients (and customers clients) with a much large code footprint.
You run into issues around AudioContext and resumption etc... it's a PITA to have to handle all those corner cases :(
I hope it’s getting better with education/more libraries. It’s also amazing how easy Codex etc… can burn through it now
However things like 'Call center helpline' turn based actually seems better! You don't want to be interrupted when giving information back and forth (I think?)
You can't beat Websockets :) Especially since you have so much tooling/existing stuff that works with HTTP.
I have been trying to get a website off the ground that does Datachannels + SQlite in the browser and then users sync between each other. I have gotten distracted so many times though.