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How to create a video call application with WebRTC

441 points by xueyongg 6y ago

xueyongg 6y ago
Took some time over the weeks to play and figure out WebRTC. Made a simple app out of it. Do check it out!
- EGreg 6y ago
  We started a simple webrtc app in 2018. Thought it would be simple. Now two years later we are still tweaking the code and dealing with handshakes and codecs across browsers, as well as edge cases involving firewalls and what to do if someone disconnects for longer than the timeout. Finally we had to invent workarounds for Cordova: https://mobile.twitter.com/qbixapps/status/1156484156425039872 https://mobile.twitter.com/qbixapps/status/11564841564250398... Did anyone set up a WebRTC that was super easy and worked rock solid with just a few lines?
  - tim-- 6y ago
    Twillio has this, but it is crazy difficult to do DIY something in the real world.
    - EGreg 6y ago
      Yes. Twilio and maybe Jitsi. We open sourced our implementation. I am still wondering if we missed some simple solution.
  - gligorot 6y ago
    First off, I'm a total noob when it comes to WebRTC, but having read the docs Google provide for it (and the accompanying mini video app tutorial) it seemed like it's dead simple to implement and use - I understood that the complexities and basically everything you talked about above was already handled. Is your implementation different than theirs or did I maybe misunderstand the value propositions of WebRTC?
    - EGreg 6y ago
      See the sister comment to yours
  - sandGorgon 6y ago
    curious to know - how do you scale webrtc ? I mean do you use kubernetes and stuff - what load balancer do you use (I assume sessions are "sticky"), etc
    - ethanwillis 6y ago
      Scaling doesn't always involve kubernetes. But the way you scale WebRTC is by using multiplexers between clients. Let's say you have 2 clients. They can directly send data to each other. Let's say you have 3 clients. Client 1 sends data to Clients 2, 3 Client 2 sends data to Clients 1, 3 Client 3 sends data to Clients 1, 2 As you add more clients the scaling requires n(n-1)/2 edges because it forms a Complete Graph. The idea then is to use a server in the middle. That server could be 3rd party or maybe even one of the participants within the graph.
      - sandGorgon 6y ago
        sure - but ultimately you will have to scale the server horizontally. or do you mean that one server can scale till millions of users
        ethanwillis 6y ago
        If you have millions of users sure use kubernetes. If you don't have millions of users just setup a handful of decently sized VMs.
  - Mulpze15 6y ago
    Aren't those codec exchanges and handshakes managed by ICE? Why would you need to go into this layer yourself?
    - fictorial 6y ago
      One example – H.264 is hardware accelerated on iPhone so one might prefer this over VP8 which could drain the device's battery pretty quickly when used in a P2P mesh setup.
figers 6y ago
WebRTC seems easy when you're creating a proof of concept with-in your own network, once you get into complex situations behind firewalls across the internet it's a whole different story. The article mentioned "This section we will just touch and go about when do you need a TURN server. It is not needed in all situations but a component needed if you have to deal with slightly less straightway use cases." a TURN server is a must in the real world...
- chrisco255 6y ago
  True but there are some open source TURN servers you can use for that.
- mothepro 6y ago
  Not necessarily, TURN servers should only be required for strict NAT. It seems that 92%* for connections are not behind strict NATs. *https://developers.google.com/talk/libjingle/important_concepts?csw=1#portssocketsconnections https://developers.google.com/talk/libjingle/important_conce...
  - FR10 6y ago
    In my tests with 10 people 5 needed TURN, ISPs in my country are really something special.
  - bkanber 6y ago
    STUN fails under symmetric NAT, not strict NAT. That google document makes no citations of that 92% figure, but I assume that's for desktop traffic only. Pretty much all mobile/cellular connections would require TURN too.
    - cma 6y ago
      I haven't found Verizon or AT&T to need TURN, but maybe clients on them would if connecting to each other.
- hardwaresofton 6y ago
flyGuyOnTheSly 6y ago
I'm eager to create a higher quality video broadcasting (not web meeting, one way only) app for some local yoga studios I help out with and am hoping this article gives me a push in the right direction. The audio quality on zoom is just terrible no matter if you disable DSP or not. So many yoga classes require high quality music. It's frustrating that chaturbate provides top notch video and audio quality for free essentially, while paying $20/mo for zoom gives you what looks like 380p video quality and audio quality I have yet to find a poor comparison for... Does anyone know how one could emulate what chaturbate does? Any good articles outlining how they do what they do? Ideally, the teacher would just plop their phone down in front of them, hit broadcast, and a few seconds of buffering later 1080p video and quality audio would be visible through a browser. Why is that so tough to do??? I haven't been able to find a single article that simplifies or distills it at all.
- GuiA 6y ago
  “Chaturbate is the best video chatting product” is a catchy line. Has anyone done a comparison post?
- bredren 6y ago
  While the video streaming isn’t that great on zoom, my experience is the audio quality piped through their custom audio thing for maxis is pretty good.
  - mkl 6y ago
    I think Zoom's audio filtering (which in my experience is pretty good) is designed only for speech, and so other sounds like music can get messed up.
    - bredren 6y ago
      Have you tried the share using the internally piped audio option? The quality Is good.
- markn951 6y ago
  It is a truism that porn always pushes the Internet’s state of the art forward.
talkingtab 6y ago
I've been looking into webrtc and used the "webrtc samples" which are good in many ways. It is fairly easy to get something up and running, but I found several areas that were difficult. * debugging. One users sound just doesn't work while it works perfectly for me with different machines. I am clueless as to how to debug it. * ice. while it works, I had a hard time understanding, tracking and debugging what was going on. * closing and restarting connections * multiple clients in one room? * echo cancellation. This was frustrating for users. * Turn. Is there a tool or way to know which clients need a turn server? Are using a turn server? I ended up guessing that getting it to be a product would actually be fairly time consuming
- FR10 6y ago
  The debugging bit is so frustrating, I almost spend a whole week trying to find a bug with my PeerConnections only to find out that it was that the TURN Server was misconfigured (although Trickle ICE was successful). And even then just setting up a TURN server consumed a whole day.
- bkanber 6y ago
  WebRTC doesn't do everything for you; it's really just responsible for tying together ICE with media streams. Signaling is up to you to figure out. For instance, multiple clients in one room: this is part of the signaling layer and is not WebRTC's responsibility (I built this into zonko.chat if you want to see how it works though). Closing and restarting connections is signaling layer stuff, ie your responsibility. Echo cancellation is really supposed to be application layer and up to you as well, but I think this will probably shift to be the browser's/WebRTC's/getUserMedia's responsibility at some point. Re. TURN: ICE is the process that works out whether a specific client needs to relay through a TURN server. The question is: do you need to implement a TURN server? The answer is: yes, you need a TURN server. If you built a P2P app that you want to work for all users, you will always need a TURN server. You can run coturn on the same box that you serve your app from. Most likely a side project will never hit the scale requiring more than a $5 digitalocean box for TURN. And yes, it should not be a surprise that products are time consuming to build :) WebRTC is plumbing; you probably were expecting something more like Jitsi.
hardwaresofton 6y ago
For those interested in seeing how WebRTC can really scale check out some of the media servers/SFU (Selective Forwarding Units)s that are out there: - janus[0][1] - mediasoup[2][3] - Medoze[4][5] It's never been easier to start your own video streaming platform. [0]: https://janus.conf.meetecho.com/ https://janus.conf.meetecho.com/ [1]: https://www.youtube.com/watch?v=zxRwELmyWU0 https://www.youtube.com/watch?v=zxRwELmyWU0 [2]: https://mediasoup.org/ https://mediasoup.org/ [3]: https://www.youtube.com/watch?v=_GhdFOZTWTw https://www.youtube.com/watch?v=_GhdFOZTWTw [4]: https://github.com/medooze/media-server https://github.com/medooze/media-server [5]: https://www.youtube.com/watch?v=u8ymYTdA0ko https://www.youtube.com/watch?v=u8ymYTdA0ko
- deleted 6y ago
  [deleted]
- Sean-Der 6y ago
  https://stackoverflow.com/a/24879451/5472819 https://stackoverflow.com/a/24879451/5472819 is a good list also, quite a few out there! Also quite a few 'WebRTC as a service' providers. I really think it has gotten way easier in the past couple years. I am biased though, don't want to think my effort has all gone to waste :p
- deskamess 6y ago
  Any solutions where WebRTC can be transcoded to an Mpeg2 transport stream?
  - hardwaresofton 6y ago
    What you want are transcoding features -- some of the servers offer it (ex. Kurento[0] for example which is not listed above), but some don't (ex. mediasoup[1]) and some offer recording but you need to wrangle formats yourself (ex. janus[2]). [0]: https://www.kurento.org/tags/transcoding https://www.kurento.org/tags/transcoding [1]: https://mediasoup.org/faq/#does-mediasoup-transcode https://mediasoup.org/faq/#does-mediasoup-transcode [2]: https://janus.conf.meetecho.com/recordplaytest.html https://janus.conf.meetecho.com/recordplaytest.html
bkanber 6y ago
I also decided to learn WebRTC and built a video chat app project: https://zonko.chat https://zonko.chat The last time I did any p2p networking was back in 2002 or something when you still had to do it all manually. We used all sorts of fun tricks like NAT hole punching, and using little script endpoints to capture and forward along port and public IP address information. It was fun to see that all of this has since been formalized under the "ICE framework". I was surprised to see that the STUN spec is only 12 years old now, despite the techniques involved being used for at least 20 years, probably more like 30+. So if anyone who's new to this whole p2p world feels that WebRTC and the ICE framework is confusing or onerous, I would point out that just a short while ago these were basically just a handful of heuristic techniques developed through trial and error over the years. It's really much easier nowadays! zonko.chat only took me 12 or so hours to build (and seems to be well-supported by chrome and ff, even mobile). Edit: Upon reflection, I don't even remember how I learned about some of them. The concept of TURN was probably one that I, and many thousands of others, invented from scratch due to necessity (failed to punch the hole? fall back to this custom relay I wrote in perl). STUN was an easy one to figure out yourself, too. I don't remember how I learned about hole punching though. Probably a forum or a book. Or possibly just an experiment ("what if the two connections touch somewhere in the internet at the same time... hey wait, it worked?") What's interesting to me is that the core "ICE" concepts (hole punching, STUN, TURN) are still pretty simple even in their mature, formalized, scientific form. But the concept of "SIP" is much more sophisticated today than it was back then.
- dvirsky 6y ago
  I worked for a company developing a custom P2P video streaming protocol. It was amazingly hard to get working properly and testing all the quirks in different routers from different manufacturers. The combination of all the NAT strategies (I think there were 5 main ones) created quite a matrix of possible ways that things can work, and you had to implement and test them all before there was any kind of standard way to do it. We built a mock internet with dozens of consumer routers, as many as we could get our hands on. We eventually got it to work and that company exists to this day, but I think they switched to WebRTC long ago.
- vxNsr 6y ago
spicyramen 6y ago
It's interesting the raise again for unified communications. All this technology specifically WebRTC is being around for few years now. The Innovation is minimum, why? Most do the problems are solved. When a technology is mature most of the focus is on security or applying other technologies to improve it such as Machine Learning. In the case of VoIP and Video apps, is very mature since the inception of H323, SIP, SCCP, RTP, sRTP, most recently JS and WebRTC.
davidsawyer 6y ago
A 19-year-old[0] built a video call app[1] with WebRTC and open-sourced[2] it. Here's a podcast interview[3] about how he did it. [0]: https://github.com/ianramzy https://github.com/ianramzy [1]: https://zipcall.io https://zipcall.io [2]: https://github.com/ianramzy/decentralized-video-chat https://github.com/ianramzy/decentralized-video-chat [3]: https://syntax.fm/show/256/webrtc-and-peer-to-peer-video-calling-with-ian-ramzy https://syntax.fm/show/256/webrtc-and-peer-to-peer-video-cal...
- fictorial 6y ago
  This is an iteration on Twilio Video's example: https://www.twilio.com/blog/2014/12/set-phasers-to-stunturn-getting-started-with-webrtc-using-node-js-socket-io-and-twilios-nat-traversal-service.html https://www.twilio.com/blog/2014/12/set-phasers-to-stunturn-... https://github.com/philnash/video-chat/tree/intro https://github.com/philnash/video-chat/tree/intro
perenzo 6y ago
https://brie.fi/ng https://brie.fi/ng - a modern pure open source WebRTC implementation. It can even blur peoples background for visual privacy. Sources at https://github.com/holtwick/briefing/ https://github.com/holtwick/briefing/
- perenzo 6y ago
  I added an own entry here https://news.ycombinator.com/item?id=23523830 https://news.ycombinator.com/item?id=23523830 Learn more about the details: https://brie.fi/ng#help https://brie.fi/ng#help Also see https://webrtc-security.github.io/ https://webrtc-security.github.io/ WebRTC is end-to-end encrypted by default. There is a signaling server that helps establishing the connections between users in a room, but after that the communication is encrypted. Also those TURN and STUN servers are only required for technical reasons to get peer-to-peer working. So no content is ever passed unencrypted. That's the difference to other services like Zoom and Jitsi, where a server in the middle is receiving the video streams unencrypted and then redistributes. Although Jitsi is adding encryption support for that as well soon.
kbumsik 6y ago
Can it be used as an UDP alternative for server-to-client communication, not browser-to-browser? If so, are there any projects implementing it?
cf 6y ago
One thing I've wanted to make is something like gather.town where I can remix the audio and video so that different users sounded louder or quieter. But, I never figured out where in the WebRTC API that is done. It seems like I need to set up my own SFU and put the necessary logic over there.
- Sean-Der 6y ago
  Are you trying to adjust the volume of the remote audio streams? Can you change value of the `audio` element of the DOM? I think you could also do it with the WebAudio API. If you throw up a repo would love to try and help :) having a backend makes it so much harder to deploy/maintain stuff.
  - cf 6y ago
    Yes I want to adjust remote audio streams but I don't see where in the API I can iterate over each user's audio feed.
    - moron4hire 6y ago
      It's not in the WebRTC API. You can change the volume on the audio tags you create, or you can pipe the audio media stream into a WebAudio graph and modify it there. I've built a number of WebRTC apps over the years. Recently, I built just such a thing as you described and open sourced it: https://www.calla.chat https://www.calla.chat. I opted to build it on top of Jitsi Meet this time. It's actually advantageous that it's not through the WebRTC API because Jitsi doesn't give access to the raw WebRTC commands. But hijacking the audio elements it creates is completely doable.
      - cf 6y ago
        Oh man this is like 90% of what I wanted to make! Thanks for making this open-source!
rergaerg 6y ago
In real life, the STUN server rarely works, and thus, the myth of this peer to peer utopia was never realised, and why webrtc did not receive any attention
jpgvm 6y ago
IPv6 is becoming increasingly common here in Asia (I'm based in Thailand but travel a lot around Asia). For context I have used 3 different ISPs here and all have dual stack, both my cellular connections have also been dual stack. This has meant NAT is less of an issue for native IPv6 endpoints, including P2P. Hopefully when IPv6 is finally widespread in US/Europe we will see stuff taking more advantage of this fact.
- Orphis 6y ago
  It's not just about NAT but firewalls too. I have native IPv6 but you can't reach my computer directly from anywhere.
shyamady 6y ago
Twilio costs but not a bad idea. I created Remotehour(https://remotehour.com https://remotehour.com) which allows you to have an 'open-door' policy video call easily. It works with Twilo :)
julius_set 6y ago
This article is a strange one. They mention WhatsApp and some other mobile products but then proceed to frame everything into the context of a browser. WebRTC works without a browser too FYI
remotists 6y ago
Most ISPs here in Asia have dual stacks.
panpanna 6y ago
Does anyone have a better tutorial on webrtc? I didn't find the article particularly good.
- erikbye 6y ago
  https://lmgtfy.com/?q=webrtc+tutorial https://lmgtfy.com/?q=webrtc+tutorial
buboard 6y ago
sadly webrtc p2p does not scale. Sadly current HTML-based solutions for media servers are slow and highly CPU intensive. Even more sadly, adobe flash has solved the problem of multi video chat decades ago, but we have decided to deprecate without alternative
mcjiggerlog 6y ago
I have some experience in this from developing https://p2p.chat https://p2p.chat a while back. As others have mentioned, building a simple project is fairly simple. The difficulty comes when you want to scale to more than ~4 users without the app becoming unusable. Adjusting audio/video constraints to ensure that you get optimal media streams is quite difficult, also. Nevermind dynamically tweaking them!
gandutraveler 6y ago
Does webrtc also work on native Android iOS apps ?
- Orphis 6y ago
  There are WebRTC SDKs for Android and iOS.
ronlobo 6y ago
Pretty cool! Check out https://github.com/meething/meething https://github.com/meething/meething dWebRTC Video Meetings MESH/SFU hybrid using GunDB, MediaSoup and Beyond! This seems to be one of the most promising projects in the WebRTC space with support from Mozilla Builders.
ryanrolds 6y ago
A small group of friends and I are working on a virtual karaoke club using WebRTC and Go, https://github.com/ryanrolds/club https://github.com/ryanrolds/club. 100% agree with the WebRTC being easy to create proof-of-concepts, but there are a lot of edge cases and browsers differences that have to be worked through.
- xueyongg 6y ago
  What are your plans to deal with the scalability issue thus far? I think that is always the biggest challenge.
  - ryanrolds 6y ago
    The current plan is to keep everyone in "groups", think friends at a table, small (max 6). The server will maintain peer connections with everyone in the "room" and broadcast the singer via that peering. As the singer changes, the server will simply allow the KJ to pick who is getting broadcast over the other server -> client peer.
xueyongg 6y ago
I really learnt so so much from the entire thread discussion here today! WebRTC I've gathered is oftentimes easy to create, but the real challenge is in the scalability. From the way it seems, scalability is only possible with the forwarding architecture with the use of the Selective Forwarding Unit or of like. I always wonder if there is a way to think outside of this 'box'.
suprfsat 6y ago
> Google handout
zumachase 6y ago
The issue with webrtc is once you step out of the side-project domain, you have to confront the endless implementation differences between browsers, whether it's undocumented SDP behavior, different codecs, non-conformant behavior for low level calls, etc. We've built a push-to-talk walkie talkie system called Squawk[0] which holds long lived webrtc connections in the background throughout the day. We use simplepeer[1] as the base to help bootstrap some of the browser shimming, but it's not perfect. So ultimately we've had to build all sorts of checks into our protocols like an audio keepalive where we send periodic frames (20ms) of silence down the media channel, and verify that we received some additional header bytes on the remote end, because otherwise webrtc would let the connections rot and you wouldn't know until you needed them which in a push-to-talk situation is too late. [0] https://www.squawk.to https://www.squawk.to [1] https://github.com/feross/simple-peer https://github.com/feross/simple-peer
- elk235 6y ago
  I am really interested in this and would love to read a technical blog post on how you built Squawk. Was it in Electron? Especially, how do you maintain all connections so users can listen to all channels at once.
- gurjeet 6y ago
  I loved the idea and interface etc. so I registered and downloaded the app. But upon launching, my corporate (ge.com) proxy blocked it: Not allowed to browse Newly Registered Domains category URL: https://app.squawk.to/ This is perhaps because your whois records are bare minimum! $ whois squawk.to Tonic whoisd V1.1 squawk noah.ns.cloudflare.com squawk kim.ns.cloudflare.com You might want to look into that. Also, to increase adoption, perhaps give the users a link to directly use the app.squawk.to URL to use as well. I tried that on my personal device and it works as advertised.
whoatethedonut 6y ago
This reminds me of Icecomm[0] from a few years back. Unfortunately, it didn't stick around for too long. It was pretty easy to use, as well, and a lot of people here ended up in a video chat together[1]. LOL! [0]: https://news.ycombinator.com/item?id=8952880 https://news.ycombinator.com/item?id=8952880 [1]: https://medium.com/@icecomm/how-launching-icecomm-on-hacker-news-created-the-most-curious-chat-roulette-ever-731fb22dc072 https://medium.com/@icecomm/how-launching-icecomm-on-hacker-...

441 points by xueyongg 6y ago

xueyongg 6y ago

Took some time over the weeks to play and figure out WebRTC. Made a simple app out of it. Do check it out!

EGreg 6y ago
We started a simple webrtc app in 2018. Thought it would be simple. Now two years later we are still tweaking the code and dealing with handshakes and codecs across browsers, as well as edge cases involving firewalls and what to do if someone disconnects for longer than the timeout. Finally we had to invent workarounds for Cordova: https://mobile.twitter.com/qbixapps/status/1156484156425039872 https://mobile.twitter.com/qbixapps/status/11564841564250398... Did anyone set up a WebRTC that was super easy and worked rock solid with just a few lines?
- tim-- 6y ago
  Twillio has this, but it is crazy difficult to do DIY something in the real world.
  - EGreg 6y ago
    Yes. Twilio and maybe Jitsi. We open sourced our implementation. I am still wondering if we missed some simple solution.
- gligorot 6y ago
  First off, I'm a total noob when it comes to WebRTC, but having read the docs Google provide for it (and the accompanying mini video app tutorial) it seemed like it's dead simple to implement and use - I understood that the complexities and basically everything you talked about above was already handled. Is your implementation different than theirs or did I maybe misunderstand the value propositions of WebRTC?
  - EGreg 6y ago
    See the sister comment to yours
- sandGorgon 6y ago
  curious to know - how do you scale webrtc ? I mean do you use kubernetes and stuff - what load balancer do you use (I assume sessions are "sticky"), etc
  - ethanwillis 6y ago
    Scaling doesn't always involve kubernetes. But the way you scale WebRTC is by using multiplexers between clients. Let's say you have 2 clients. They can directly send data to each other. Let's say you have 3 clients. Client 1 sends data to Clients 2, 3 Client 2 sends data to Clients 1, 3 Client 3 sends data to Clients 1, 2 As you add more clients the scaling requires n(n-1)/2 edges because it forms a Complete Graph. The idea then is to use a server in the middle. That server could be 3rd party or maybe even one of the participants within the graph.
    - sandGorgon 6y ago
      sure - but ultimately you will have to scale the server horizontally. or do you mean that one server can scale till millions of users
      - ethanwillis 6y ago
        If you have millions of users sure use kubernetes. If you don't have millions of users just setup a handful of decently sized VMs.
- Mulpze15 6y ago
  Aren't those codec exchanges and handshakes managed by ICE? Why would you need to go into this layer yourself?
  - fictorial 6y ago
    One example – H.264 is hardware accelerated on iPhone so one might prefer this over VP8 which could drain the device's battery pretty quickly when used in a P2P mesh setup.

figers 6y ago

WebRTC seems easy when you're creating a proof of concept with-in your own network, once you get into complex situations behind firewalls across the internet it's a whole different story. The article mentioned "This section we will just touch and go about when do you need a TURN server. It is not needed in all situations but a component needed if you have to deal with slightly less straightway use cases." a TURN server is a must in the real world...

chrisco255 6y ago
True but there are some open source TURN servers you can use for that.
mothepro 6y ago
Not necessarily, TURN servers should only be required for strict NAT. It seems that 92%* for connections are not behind strict NATs. *https://developers.google.com/talk/libjingle/important_concepts?csw=1#portssocketsconnections https://developers.google.com/talk/libjingle/important_conce...
- FR10 6y ago
  In my tests with 10 people 5 needed TURN, ISPs in my country are really something special.
- bkanber 6y ago
  STUN fails under symmetric NAT, not strict NAT. That google document makes no citations of that 92% figure, but I assume that's for desktop traffic only. Pretty much all mobile/cellular connections would require TURN too.
  - cma 6y ago
    I haven't found Verizon or AT&T to need TURN, but maybe clients on them would if connecting to each other.
hardwaresofton 6y ago

flyGuyOnTheSly 6y ago

I'm eager to create a higher quality video broadcasting (not web meeting, one way only) app for some local yoga studios I help out with and am hoping this article gives me a push in the right direction. The audio quality on zoom is just terrible no matter if you disable DSP or not. So many yoga classes require high quality music. It's frustrating that chaturbate provides top notch video and audio quality for free essentially, while paying $20/mo for zoom gives you what looks like 380p video quality and audio quality I have yet to find a poor comparison for... Does anyone know how one could emulate what chaturbate does? Any good articles outlining how they do what they do? Ideally, the teacher would just plop their phone down in front of them, hit broadcast, and a few seconds of buffering later 1080p video and quality audio would be visible through a browser. Why is that so tough to do??? I haven't been able to find a single article that simplifies or distills it at all.

GuiA 6y ago
“Chaturbate is the best video chatting product” is a catchy line. Has anyone done a comparison post?
bredren 6y ago
While the video streaming isn’t that great on zoom, my experience is the audio quality piped through their custom audio thing for maxis is pretty good.
- mkl 6y ago
  I think Zoom's audio filtering (which in my experience is pretty good) is designed only for speech, and so other sounds like music can get messed up.
  - bredren 6y ago
    Have you tried the share using the internally piped audio option? The quality Is good.
markn951 6y ago
It is a truism that porn always pushes the Internet’s state of the art forward.

talkingtab 6y ago

I've been looking into webrtc and used the "webrtc samples" which are good in many ways. It is fairly easy to get something up and running, but I found several areas that were difficult. * debugging. One users sound just doesn't work while it works perfectly for me with different machines. I am clueless as to how to debug it. * ice. while it works, I had a hard time understanding, tracking and debugging what was going on. * closing and restarting connections * multiple clients in one room? * echo cancellation. This was frustrating for users. * Turn. Is there a tool or way to know which clients need a turn server? Are using a turn server? I ended up guessing that getting it to be a product would actually be fairly time consuming

FR10 6y ago
The debugging bit is so frustrating, I almost spend a whole week trying to find a bug with my PeerConnections only to find out that it was that the TURN Server was misconfigured (although Trickle ICE was successful). And even then just setting up a TURN server consumed a whole day.
bkanber 6y ago
WebRTC doesn't do everything for you; it's really just responsible for tying together ICE with media streams. Signaling is up to you to figure out. For instance, multiple clients in one room: this is part of the signaling layer and is not WebRTC's responsibility (I built this into zonko.chat if you want to see how it works though). Closing and restarting connections is signaling layer stuff, ie your responsibility. Echo cancellation is really supposed to be application layer and up to you as well, but I think this will probably shift to be the browser's/WebRTC's/getUserMedia's responsibility at some point. Re. TURN: ICE is the process that works out whether a specific client needs to relay through a TURN server. The question is: do you need to implement a TURN server? The answer is: yes, you need a TURN server. If you built a P2P app that you want to work for all users, you will always need a TURN server. You can run coturn on the same box that you serve your app from. Most likely a side project will never hit the scale requiring more than a $5 digitalocean box for TURN. And yes, it should not be a surprise that products are time consuming to build :) WebRTC is plumbing; you probably were expecting something more like Jitsi.

hardwaresofton 6y ago

For those interested in seeing how WebRTC can really scale check out some of the media servers/SFU (Selective Forwarding Units)s that are out there: - janus[0][1] - mediasoup[2][3] - Medoze[4][5] It's never been easier to start your own video streaming platform. [0]: https://janus.conf.meetecho.com/ https://janus.conf.meetecho.com/ [1]: https://www.youtube.com/watch?v=zxRwELmyWU0 https://www.youtube.com/watch?v=zxRwELmyWU0 [2]: https://mediasoup.org/ https://mediasoup.org/ [3]: https://www.youtube.com/watch?v=_GhdFOZTWTw https://www.youtube.com/watch?v=_GhdFOZTWTw [4]: https://github.com/medooze/media-server https://github.com/medooze/media-server [5]: https://www.youtube.com/watch?v=u8ymYTdA0ko https://www.youtube.com/watch?v=u8ymYTdA0ko

deleted 6y ago
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Sean-Der 6y ago
https://stackoverflow.com/a/24879451/5472819 https://stackoverflow.com/a/24879451/5472819 is a good list also, quite a few out there! Also quite a few 'WebRTC as a service' providers. I really think it has gotten way easier in the past couple years. I am biased though, don't want to think my effort has all gone to waste :p
deskamess 6y ago
Any solutions where WebRTC can be transcoded to an Mpeg2 transport stream?
- hardwaresofton 6y ago
  What you want are transcoding features -- some of the servers offer it (ex. Kurento[0] for example which is not listed above), but some don't (ex. mediasoup[1]) and some offer recording but you need to wrangle formats yourself (ex. janus[2]). [0]: https://www.kurento.org/tags/transcoding https://www.kurento.org/tags/transcoding [1]: https://mediasoup.org/faq/#does-mediasoup-transcode https://mediasoup.org/faq/#does-mediasoup-transcode [2]: https://janus.conf.meetecho.com/recordplaytest.html https://janus.conf.meetecho.com/recordplaytest.html

bkanber 6y ago

I also decided to learn WebRTC and built a video chat app project: https://zonko.chat https://zonko.chat The last time I did any p2p networking was back in 2002 or something when you still had to do it all manually. We used all sorts of fun tricks like NAT hole punching, and using little script endpoints to capture and forward along port and public IP address information. It was fun to see that all of this has since been formalized under the "ICE framework". I was surprised to see that the STUN spec is only 12 years old now, despite the techniques involved being used for at least 20 years, probably more like 30+. So if anyone who's new to this whole p2p world feels that WebRTC and the ICE framework is confusing or onerous, I would point out that just a short while ago these were basically just a handful of heuristic techniques developed through trial and error over the years. It's really much easier nowadays! zonko.chat only took me 12 or so hours to build (and seems to be well-supported by chrome and ff, even mobile). Edit: Upon reflection, I don't even remember how I learned about some of them. The concept of TURN was probably one that I, and many thousands of others, invented from scratch due to necessity (failed to punch the hole? fall back to this custom relay I wrote in perl). STUN was an easy one to figure out yourself, too. I don't remember how I learned about hole punching though. Probably a forum or a book. Or possibly just an experiment ("what if the two connections touch somewhere in the internet at the same time... hey wait, it worked?") What's interesting to me is that the core "ICE" concepts (hole punching, STUN, TURN) are still pretty simple even in their mature, formalized, scientific form. But the concept of "SIP" is much more sophisticated today than it was back then.

dvirsky 6y ago
I worked for a company developing a custom P2P video streaming protocol. It was amazingly hard to get working properly and testing all the quirks in different routers from different manufacturers. The combination of all the NAT strategies (I think there were 5 main ones) created quite a matrix of possible ways that things can work, and you had to implement and test them all before there was any kind of standard way to do it. We built a mock internet with dozens of consumer routers, as many as we could get our hands on. We eventually got it to work and that company exists to this day, but I think they switched to WebRTC long ago.
vxNsr 6y ago

spicyramen 6y ago

It's interesting the raise again for unified communications. All this technology specifically WebRTC is being around for few years now. The Innovation is minimum, why? Most do the problems are solved. When a technology is mature most of the focus is on security or applying other technologies to improve it such as Machine Learning. In the case of VoIP and Video apps, is very mature since the inception of H323, SIP, SCCP, RTP, sRTP, most recently JS and WebRTC.

davidsawyer 6y ago

A 19-year-old[0] built a video call app[1] with WebRTC and open-sourced[2] it. Here's a podcast interview[3] about how he did it. [0]: https://github.com/ianramzy https://github.com/ianramzy [1]: https://zipcall.io https://zipcall.io [2]: https://github.com/ianramzy/decentralized-video-chat https://github.com/ianramzy/decentralized-video-chat [3]: https://syntax.fm/show/256/webrtc-and-peer-to-peer-video-calling-with-ian-ramzy https://syntax.fm/show/256/webrtc-and-peer-to-peer-video-cal...

fictorial 6y ago
This is an iteration on Twilio Video's example: https://www.twilio.com/blog/2014/12/set-phasers-to-stunturn-getting-started-with-webrtc-using-node-js-socket-io-and-twilios-nat-traversal-service.html https://www.twilio.com/blog/2014/12/set-phasers-to-stunturn-... https://github.com/philnash/video-chat/tree/intro https://github.com/philnash/video-chat/tree/intro

perenzo 6y ago

https://brie.fi/ng https://brie.fi/ng - a modern pure open source WebRTC implementation. It can even blur peoples background for visual privacy. Sources at https://github.com/holtwick/briefing/ https://github.com/holtwick/briefing/

perenzo 6y ago
I added an own entry here https://news.ycombinator.com/item?id=23523830 https://news.ycombinator.com/item?id=23523830 Learn more about the details: https://brie.fi/ng#help https://brie.fi/ng#help Also see https://webrtc-security.github.io/ https://webrtc-security.github.io/ WebRTC is end-to-end encrypted by default. There is a signaling server that helps establishing the connections between users in a room, but after that the communication is encrypted. Also those TURN and STUN servers are only required for technical reasons to get peer-to-peer working. So no content is ever passed unencrypted. That's the difference to other services like Zoom and Jitsi, where a server in the middle is receiving the video streams unencrypted and then redistributes. Although Jitsi is adding encryption support for that as well soon.

kbumsik 6y ago

Can it be used as an UDP alternative for server-to-client communication, not browser-to-browser? If so, are there any projects implementing it?

cf 6y ago

One thing I've wanted to make is something like gather.town where I can remix the audio and video so that different users sounded louder or quieter. But, I never figured out where in the WebRTC API that is done. It seems like I need to set up my own SFU and put the necessary logic over there.

Sean-Der 6y ago
Are you trying to adjust the volume of the remote audio streams? Can you change value of the `audio` element of the DOM? I think you could also do it with the WebAudio API. If you throw up a repo would love to try and help :) having a backend makes it so much harder to deploy/maintain stuff.
- cf 6y ago
  Yes I want to adjust remote audio streams but I don't see where in the API I can iterate over each user's audio feed.
  - moron4hire 6y ago
    It's not in the WebRTC API. You can change the volume on the audio tags you create, or you can pipe the audio media stream into a WebAudio graph and modify it there. I've built a number of WebRTC apps over the years. Recently, I built just such a thing as you described and open sourced it: https://www.calla.chat https://www.calla.chat. I opted to build it on top of Jitsi Meet this time. It's actually advantageous that it's not through the WebRTC API because Jitsi doesn't give access to the raw WebRTC commands. But hijacking the audio elements it creates is completely doable.
    - cf 6y ago
      Oh man this is like 90% of what I wanted to make! Thanks for making this open-source!

rergaerg 6y ago

In real life, the STUN server rarely works, and thus, the myth of this peer to peer utopia was never realised, and why webrtc did not receive any attention

jpgvm 6y ago

IPv6 is becoming increasingly common here in Asia (I'm based in Thailand but travel a lot around Asia). For context I have used 3 different ISPs here and all have dual stack, both my cellular connections have also been dual stack. This has meant NAT is less of an issue for native IPv6 endpoints, including P2P. Hopefully when IPv6 is finally widespread in US/Europe we will see stuff taking more advantage of this fact.

Orphis 6y ago
It's not just about NAT but firewalls too. I have native IPv6 but you can't reach my computer directly from anywhere.

shyamady 6y ago

Twilio costs but not a bad idea. I created Remotehour(https://remotehour.com https://remotehour.com) which allows you to have an 'open-door' policy video call easily. It works with Twilo :)

julius_set 6y ago

This article is a strange one. They mention WhatsApp and some other mobile products but then proceed to frame everything into the context of a browser. WebRTC works without a browser too FYI

remotists 6y ago

Most ISPs here in Asia have dual stacks.

panpanna 6y ago

Does anyone have a better tutorial on webrtc? I didn't find the article particularly good.

erikbye 6y ago
https://lmgtfy.com/?q=webrtc+tutorial https://lmgtfy.com/?q=webrtc+tutorial

buboard 6y ago

sadly webrtc p2p does not scale. Sadly current HTML-based solutions for media servers are slow and highly CPU intensive. Even more sadly, adobe flash has solved the problem of multi video chat decades ago, but we have decided to deprecate without alternative

mcjiggerlog 6y ago

I have some experience in this from developing https://p2p.chat https://p2p.chat a while back. As others have mentioned, building a simple project is fairly simple. The difficulty comes when you want to scale to more than ~4 users without the app becoming unusable. Adjusting audio/video constraints to ensure that you get optimal media streams is quite difficult, also. Nevermind dynamically tweaking them!

gandutraveler 6y ago

Does webrtc also work on native Android iOS apps ?

Orphis 6y ago
There are WebRTC SDKs for Android and iOS.

ronlobo 6y ago

Pretty cool! Check out https://github.com/meething/meething https://github.com/meething/meething dWebRTC Video Meetings MESH/SFU hybrid using GunDB, MediaSoup and Beyond! This seems to be one of the most promising projects in the WebRTC space with support from Mozilla Builders.

ryanrolds 6y ago

A small group of friends and I are working on a virtual karaoke club using WebRTC and Go, https://github.com/ryanrolds/club https://github.com/ryanrolds/club. 100% agree with the WebRTC being easy to create proof-of-concepts, but there are a lot of edge cases and browsers differences that have to be worked through.

xueyongg 6y ago
What are your plans to deal with the scalability issue thus far? I think that is always the biggest challenge.
- ryanrolds 6y ago
  The current plan is to keep everyone in "groups", think friends at a table, small (max 6). The server will maintain peer connections with everyone in the "room" and broadcast the singer via that peering. As the singer changes, the server will simply allow the KJ to pick who is getting broadcast over the other server -> client peer.

xueyongg 6y ago

I really learnt so so much from the entire thread discussion here today! WebRTC I've gathered is oftentimes easy to create, but the real challenge is in the scalability. From the way it seems, scalability is only possible with the forwarding architecture with the use of the Selective Forwarding Unit or of like. I always wonder if there is a way to think outside of this 'box'.

suprfsat 6y ago

> Google handout

zumachase 6y ago

The issue with webrtc is once you step out of the side-project domain, you have to confront the endless implementation differences between browsers, whether it's undocumented SDP behavior, different codecs, non-conformant behavior for low level calls, etc. We've built a push-to-talk walkie talkie system called Squawk[0] which holds long lived webrtc connections in the background throughout the day. We use simplepeer[1] as the base to help bootstrap some of the browser shimming, but it's not perfect. So ultimately we've had to build all sorts of checks into our protocols like an audio keepalive where we send periodic frames (20ms) of silence down the media channel, and verify that we received some additional header bytes on the remote end, because otherwise webrtc would let the connections rot and you wouldn't know until you needed them which in a push-to-talk situation is too late. [0] https://www.squawk.to https://www.squawk.to [1] https://github.com/feross/simple-peer https://github.com/feross/simple-peer

elk235 6y ago
I am really interested in this and would love to read a technical blog post on how you built Squawk. Was it in Electron? Especially, how do you maintain all connections so users can listen to all channels at once.
gurjeet 6y ago
I loved the idea and interface etc. so I registered and downloaded the app. But upon launching, my corporate (ge.com) proxy blocked it: Not allowed to browse Newly Registered Domains category URL: https://app.squawk.to/ This is perhaps because your whois records are bare minimum! $ whois squawk.to Tonic whoisd V1.1 squawk noah.ns.cloudflare.com squawk kim.ns.cloudflare.com You might want to look into that. Also, to increase adoption, perhaps give the users a link to directly use the app.squawk.to URL to use as well. I tried that on my personal device and it works as advertised.

whoatethedonut 6y ago

This reminds me of Icecomm[0] from a few years back. Unfortunately, it didn't stick around for too long. It was pretty easy to use, as well, and a lot of people here ended up in a video chat together[1]. LOL! [0]: https://news.ycombinator.com/item?id=8952880 https://news.ycombinator.com/item?id=8952880 [1]: https://medium.com/@icecomm/how-launching-icecomm-on-hacker-news-created-the-most-curious-chat-roulette-ever-731fb22dc072 https://medium.com/@icecomm/how-launching-icecomm-on-hacker-...