[HN Gopher] WebRTC Samples
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WebRTC Samples
Author : simonpure
Score : 78 points
Date : 2020-09-14 19:07 UTC (3 hours ago)
(HTM) web link (webrtc.github.io)
(TXT) w3m dump (webrtc.github.io)
| ChicagoBoy11 wrote:
| For those in the know, if I'm running something on my own that
| may only at any given moment see in the few hundreds of people
| using it, what's the best approach in terms of server-side
| infrastructure to support WebRTC apps? Is there any SaaS offering
| that has a meaningful product?
| ajb413 wrote:
| You can use PubNub for reliable signaling and XirSys for TURN.
| I made an OS example of this a while back.
| https://github.com/ajb413/pubnub-js-webrtc
| Sean-Der wrote:
| A few hundred people should be fine! For fun I made
| https://github.com/pion/rtsp-bench and was doing 14k people
| before I saw my first hiccup.
|
| You just need watch CPU Usage and Network attributes (loss,
| congestion etc...) Getting this right will take a little bit of
| WebRTC knowledge. Happy to answer any specific questions :)
|
| There are also lots of hosted options as well! Real world
| WebRTC has a few different parts though,
| https://bloggeek.me/webrtc-server/ is a good intro I think.
| ChicagoBoy11 wrote:
| Thank you for these resources -- I'll check them out!
| amelius wrote:
| Any reason why this requires both Python _and_ Nodejs?
| Sean-Der wrote:
| The term 'WebRTC' is overloaded, it can mean the protocol or the
| API. These samples only cover in the browser. This repo is also
| kind of dead https://github.com/webrtc/samples/issues/1350
|
| If you are interested in WebRTC out of the browser there are lots
| of implementations and servers! I am also working on a book
| 'WebRTC for the Curious'[0] that tries to explain the protocol
| and the history behind it.
|
| [0] https://webrtcforthecurious.com/
| suyash wrote:
| Share GitHub repo with latest code, love to play around
| Sean-Der wrote:
| The project I work on is https://github.com/pion/webrtc.
| https://github.com/pion/awesome-pion is a list of interesting
| projects that use it, my favorite thing to share :)
|
| For other WebRTC implementations check out.
|
| * https://github.com/aiortc/aiortc
|
| * https://github.com/shinyoshiaki/werift-webrtc
|
| * https://github.com/rawrtc/rawrtc
|
| * https://gstreamer.freedesktop.org/documentation/webrtc/inde
| x...
|
| There is also https://github.com/node-webrtc/node-webrtc this
| takes Google's WebRTC implementation (used in Chromium) and
| throws a node.js API on top of it!
| lxgr wrote:
| Isn't the protocol more or less "just" SDP, DTLS-SRTP and ICE,
| or in other words a normal SIP stack without the SIP part?
|
| Or is the WebRTC subset/combination of these standards
| sufficiently different better described as its own protocol?
| Sean-Der wrote:
| Yea pretty much! The only other thing is SCTP.
|
| One nice thing is that the protocol is less flexible. The
| clients are expected to support NACK, Congestion Control and
| some other stuff.
|
| I haven't done SIP stuff in 10 years. I just remember you had
| lots of variety between who you interacted with. Maybe this
| has gotten better!
| gxqoz wrote:
| This blog post from 2018 from Discord indicated that the
| browser implementation of WebRTC wasn't as good:
| https://blog.discordapp.com/how-discord-handles-two-and-half...
|
| Is this still the case? Would you still hypothetically see
| better audio performance of something like Discord in a
| standalone app?
| Sean-Der wrote:
| One big issues was poor crypto performance. I managed to
| convince Google to turn AES-GCM on with SRTP! https://bugs.ch
| romium.org/p/chromium/issues/detail?id=713701...
|
| I am sure there are other issues as well. The WebRTC
| implementation in Chromium is controlled by Google (they have
| all the commit bits). If they don't agree with your change it
| isn't going to land. It's a tough position to be in for other
| businesses and Open Source projects even.
| suyash wrote:
| If you want to build a multiple session (multiple people)
| connecting to a website with their camera and audio on, would you
| use WebRTC or WebSocket + getUserMedia ? Why ?
| Sean-Der wrote:
| WebRTC is going to be better for most users (IMO). WebRTC
| provides congestion control so video stays real-time. It works
| pretty well with very little effort.
|
| Some companies have decided to roll their own,
| https://webrtchacks.com/zoom-avoids-using-webrtc/ is a cool
| read.
| suyash wrote:
| how about if you want to integrate real time chat, can webRTC
| data channel accomplish that or one needs WebSockets for
| that?
| moron4hire wrote:
| Go do your own homework
| kristianpaul wrote:
| There is also
| https://janus.conf.meetecho.com/screensharingtest.html just to
| have more resources to check
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(page generated 2020-09-14 23:00 UTC)