[OSS4?] WebRTC - Cargo Cult Revolution in Web Conferencing

OSS specific Linux discussion (x86/amd64)

Moderators: cesium, dev, kodachi, hannu

[OSS4?] WebRTC - Cargo Cult Revolution in Web Conferencing

Postby igorzwx » Sat Jun 25, 2011 10:15 pm

WebRTC is coordinating an effort to let people call each other and hold videoconferences just by visiting a Web site. If they're successful, Web developers will be able to add these features to their sites just by using some relatively simple JavaScript code. Even better, the audio and video codecs WebRTC uses have free software implementations, and come with patent licenses that offer legal protection to users and developers. This project is still in its early stages; right now, the effort is focused on adding the necessary support to different browsers.
http://www.fsf.org/blogs/community/skyp ... t-projects


It is not simply an "open-source" replacement for Skype, it is a global Cargo Cult revolution.

You see, there is already FaceFlow ("free group video chat" and more) http://www.faceflow.com/
You can sign up for free on FaceFlow and make video conferencing with your friends and family.
Now imagine that anyone, who has a website, can easily create his own "FaceFlow" and host video conferences.
Any educational institution may easily create own e-learning sites, or add "these features" to their Moodle, or Mahara
http://en.wikipedia.org/wiki/Moodle
http://en.wikipedia.org/wiki/Mahara_(software)
Mahara is a very popular "open-source" Cargo Cult, you can try it for free here http://demo.mahara.org/
It is not true that Cargo Cults always fail. There are successful ones, for example: Moodle, Mahara, LimeSurvey http://en.wikipedia.org/wiki/LimeSurvey
To be successful, a Cargo Cult should meet certain "necessary and satisfactory conditions".

WebRTC was "open-sourced" by Google on June 1, 2011 http://en.wikipedia.org/wiki/WebRTC
Google Video Chat is said to be migrating to the "open-sourced" webRTC.
Notice that Google Video Chat failed to become "popular". Windows users, as a rule, do not like Google Video Chat because of low quality of audio and video. There are also problems with "usability", etc.

Google Talk Video Chat plugin does not work with OSS4.
Google has been always with ALSA (because it said to be "very advanced").
It is very probable that Google experts are not competent in audio and video technologies.
This means that WebRTC project may fail like many other Cargo Cults.
There is a little hope that it might be possible to make it work with OSS4.

cesium wrote:adding OSS support (if it's not there) should not be difficult - it was easy to add for mangler... viewtopic.php?f=3&t=4261#p16849


WebRTC is a framework that was open sourced on June 1, 2011 that allows web browsers to conduct real-time video chat.
Its inclusion in the World Wide Web Consortium (W3C) standards is supported by Google, Mozilla and Opera.[1][2][3]
It is licensed under the BSD-3 clause license and the code is based on products from Global IP Solutions — a company Google acquired in May 2010.[4][5][6][7] WebRTC uses the iLBC, iSAC, G.711 and G.722 codecs for audio and VP8 for video[8] and work is underway to migrate the google talk video chat plugin to the webRTC framework.[9] http://en.wikipedia.org/wiki/WebRTC


Google battles MicroSkype with 'open' VoIP protocol
Earlier this year, Microsoft purchased popular VoIP services Skype in a deal valued at $8.5 billion, making it the biggest acquisition in Microsoft's history. But Google is taking a very different tack. In addition to adopting Jingle, Google recently open sourced a framework for real-time video and audio inside the browser. Known as WebRTC, the framework is based on technology Google acquired with its $68.2 million purchase of Global IP Solutions (GIPS) last year. http://www.theregister.co.uk/2011/06/24 ... to_jingle/

See also: http://sites.google.com/site/webrtc/
igorzwx
Supporter
 
Posts: 987
Joined: Sun Jun 28, 2009 9:31 pm

Re: [OSS4?] WebRTC - Cargo Cult Revolution in Web Conferenci

Postby igorzwx » Wed Jun 27, 2012 7:24 am

Mozilla WebRTC integration[Demo Video]
http://www.youtube.com/watch?v=lifNIRfWQXA

TRANSCRIPT:
Hi, I’m Anant from Mozilla Labs and I’m here at IETF where we are demonstrating a simple video call between two BrowserID-authenticated parties, using the new WebRTC APIs that we are working on.

This is a special build of Firefox with WebRTC support, and also has the experimental SocialAPI add-on from Mozilla Labs installed. On the right hand side you can see web content served by demosocialservice.org, to which I will sign with BrowserID. Once I’m signed in, I can see all my online friends on the sidebar. I see my friend Enda is currently online, and so I’m going to click the video chat button to initiate a call.

Here, I see a very early prototype of a video call window served up by our demo social service. Now, I can click the Start Call button to let Enda know that I want to speak with him. Once he accepts the call, a video stream is established between the two parties as you can see. So, that was a video call built entirely using JavaScript and HTML!

You can check out the source code for this demo, as well as learn how to contribute to the ongoing WebRTC efforts at Mozilla in this blog post. Thanks for watching!
http://hacks.mozilla.org/2012/04/webrtc ... t-mozilla/

The WebRTC standard has been endorsed by a number of browser vendors. The standard group’s reason for being is that “Currently, there is no free, high quality, complete solution available that enables communication in the browser." WebRTC aims to make that happen. http://phys.org/news/2012-04-webrtc-vid ... owser.html


WebRTC (Web Real-Time Communication) http://en.wikipedia.org/wiki/WebRTC
WebRTC.org http://www.webrtc.org/
Testing WebRTC on Chrome http://www.webrtc.org/running-the-demos

W3C: WebRTC 1.0: Real-time Communication Between Browsers
http://www.w3.org/TR/webrtc/

Google open sources $68.2m realtime comm platform
Audio and video chatter inside the browser

http://www.theregister.co.uk/2011/06/01 ... es_webrtc/

Architecture
1 Your Web App
2 Web API
3 WebRTC Native C++ API
4 Transport / Session
4.1 RTP Stack
4.2 STUN/ICE
4.3 Session Management
5 VoiceEngine
5.1 iSAC
5.2 iLBC
5.3 NetEQ for Voice
5.4 Acoustic Echo Canceler (AEC)
5.5 Noise Reduction (NR)
6 VideoEngine
6.1 VP8
6.2 Video Jitter Buffer
6.3 Image enhancements
http://www.webrtc.org/reference/architecture


SRTP (encryption) now on by default, meaning all audio and video data will be encrypted. http://www.webrtc.org/blog/webrtcchange ... devchannel
igorzwx
Supporter
 
Posts: 987
Joined: Sun Jun 28, 2009 9:31 pm


Return to Linux

Who is online

Users browsing this forum: No registered users and 1 guest