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jlaine
aiortc
Python2018–2018
jlaine / aiortc

aiortc

WebRTC and ORTC implementation for Python using asyncio

4Distinct people
6Mentions
0Explicit recommenders
146GitHub stars
REPOSITORY AT A GLANCE

What to know before exploring

What it is

WebRTC and ORTC implementation for Python using asyncio

GitHub

Where it is used

No documented product use has been verified yet.

What people are saying

No recommendation phrase was found in the indexed comments.

No classified warning is attached to this page. Read the discussions before treating silence as approval.

Also in these discussions

No other repository is repeatedly linked in the same discussions.

This week on Hacker News

Nov 7–Nov 14 UTC · Direct repository links and separately labeled related project stories. Indexed through 11/14/2026, 5:04:57 PM UTC.

No direct repository links were found in the indexed HN discussions over the past seven days. Older discussions are available below.

Mentions over time

HN mentions by week6 dated mentions

Observed mentions: 2018-10-08 12:23:11 UTC to 2018-12-11 14:36:50 UTC. Weeks start Monday, UTC.

01232018-10-08 00:00:00 UTC to 2018-10-15 00:00:00 UTC (exclusive): 3 mentions; 1 submissions, 2 commentsOct 82018-10-15 00:00:00 UTC to 2018-10-22 00:00:00 UTC (exclusive): 0 mentions; 0 submissions, 0 comments2018-10-22 00:00:00 UTC to 2018-10-29 00:00:00 UTC (exclusive): 0 mentions; 0 submissions, 0 comments2018-10-29 00:00:00 UTC to 2018-11-05 00:00:00 UTC (exclusive): 0 mentions; 0 submissions, 0 comments2018-11-05 00:00:00 UTC to 2018-11-12 00:00:00 UTC (exclusive): 0 mentions; 0 submissions, 0 commentsNov 52018-11-12 00:00:00 UTC to 2018-11-19 00:00:00 UTC (exclusive): 1 mentions; 0 submissions, 1 comments2018-11-19 00:00:00 UTC to 2018-11-26 00:00:00 UTC (exclusive): 1 mentions; 0 submissions, 1 comments2018-11-26 00:00:00 UTC to 2018-12-03 00:00:00 UTC (exclusive): 0 mentions; 0 submissions, 0 comments2018-12-03 00:00:00 UTC to 2018-12-10 00:00:00 UTC (exclusive): 0 mentions; 0 submissions, 0 comments2018-12-10 00:00:00 UTC to 2018-12-17 00:00:00 UTC (exclusive): 1 mentions; 0 submissions, 1 commentsDec 10

Counts come from the repository’s dated source evidence. Index cutoff: 2026-11-02 08:16:59 UTC.

View mention counts and date ranges
UTC ranges include their start and exclude their end. Select a period to filter the source evidence.
PeriodStart (UTC)End (exclusive, UTC)MentionsSubmissionsComments
Oct 82018-10-08 00:00:00 UTC2018-10-15 00:00:00 UTC312
Oct 152018-10-15 00:00:00 UTC2018-10-22 00:00:00 UTC000
Oct 222018-10-22 00:00:00 UTC2018-10-29 00:00:00 UTC000
Oct 292018-10-29 00:00:00 UTC2018-11-05 00:00:00 UTC000
Nov 52018-11-05 00:00:00 UTC2018-11-12 00:00:00 UTC000
Nov 122018-11-12 00:00:00 UTC2018-11-19 00:00:00 UTC101
Nov 192018-11-19 00:00:00 UTC2018-11-26 00:00:00 UTC101
Nov 262018-11-26 00:00:00 UTC2018-12-03 00:00:00 UTC000
Dec 32018-12-03 00:00:00 UTC2018-12-10 00:00:00 UTC000
Dec 102018-12-10 00:00:00 UTC2018-12-17 00:00:00 UTC101

What Hacker News says

6 sources
https://github.com/jlaine/aiortc may be a better option than janus. It's a much simpler to use implementation and is python. quick summary of the acronyms... STUN/TURN/ICE are all related to NAT traversal (ie. when you are behind a router or firewall of some kind and your computer's IP is different from your public IP). The reason you need this is because UDP is not connection-based like TCP. So STUN is a service that helps your computer find its public IP. ICE is a protocol for finding and reporting peer candidates (after they've learned their public IPs via STUN) so they can establish a peer-to-peer connection. TURN is when you reflect the connection off a public server instead of establishing a peer-to-peer connection. SDP is session description...basically information on what media formats and other things each client will accept
Read the rest
. RTP is how the media (or other data) packets are framed RTCP is out of band reporting for things like packet loss and other info so that each side can adjust media properties The nice thing about WebRTC is it takes care most of this stuff for you. If you're always connecting to a server with a public IP you'll just need a few things: 1) WebRTC server that can accept media data (like janus or aiortc) 2) Some signalling mechanism, Websockets work well for this 3) you _may_ need a STUN server, google has stun servers available but you could also use coturn for this. The rest you shouldn't need to worry as much about because webrtc handles it all under the hood
In Adventures in Video Conferencing Part 1: The Wild World of WebRTC
There have been a few libraries popup that tie libnice + OpenSSL + libsrtp + usrsctp together, it is nice that the stack is all there. Just have to jump through lots of hoops :) The creation of them also seems to be accelerating! I think the P2P excitement has helped a lot. * https://github.com/xhs/librtcdc (This was the first non-chrome one afaik) * https://github.com/chadnickbok/librtcdcpp * https://github.com/jlaine/aiortc * https://github.com/rawrtc/rawrtc * https://github.com/pions/webrtc * https://github.com/nodertc * https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gs... There are also a few that aren't public, I haven't used them directly though. Only by talking to other devs.
In Show HN: WebRTTY – Share a terminal session over WebRTC
There are several examples available from the project's github repo: https://github.com/jlaine/aiortc/tree/master/examples I personally like the "server" and "apprtc" demos as you can easily talk to a browser. They illustrate slightly different things: - "apprtc" relies on a third-party signaling service, and shows you how you can play media from a file / record it to a file, or generate video frame-by-frame. - "server" shows you how you can combine media and signaling into a single Python-based server, and how you can apply image processing on the fly to the received video.
In Show HN: WebRTC implementation for Python using Asyncio

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