Thursday, June 02, 2011

WebRTC - bringing real time communications to the web.

WebRTC (Real-Time Communications) is an open-source project supported by Google. WebRTC components come from Google’s acquisition of GIPS (Global IP Solutions)  formerly "Global IP Sound".

It's been released with a Google License that's similar to a BSD license.

Some experiments I was doing with P2P between browsers back in 1996/7 using Java had mixed results. UDP in java wasn't always available or would act like it's working but never send or recieve or just revieve.  There was no way to get TCP/IP through NAT's and Firewalls at the time, I did fine a method that did worked that I published in 2002.  I was stunned when I found out Skype was using my firewall bypass idea. See posts on Slashdot.

I have a feeling WebRTC is going to be getting a lot more attention over time. The GIPS (Global IP Solutions) product was using by Citrix Online, Cisco’s WebEX, Yahoo, IBM, AOL and many others in particular Skype itself. 


From GIPS orignial Marketing Materials:
VoiceEngine
The VoiceEngine family consists of comprehensive, packaged solutions that handle all the necessary voice processing tasks for IP networks, providing superior voice quality even under adverse network conditions. VoiceEngine allows application developers, service providers and hardware manufacturers the ability to easily build complex voice processing technology into their solutions, without worrying about the effects of delay, jitter, packet loss, background noise and echo. These products have been optimized for high performance on a number of platforms- from PCs to Mobile devices- allowing customers the flexibility to design almost any application or device around Global IP Solutions’ cutting-edge technology.

VideoEngine
Global IP Solutions’ VideoEngine products deliver world-class voice and video quality to PC softphones, Unified Communications applications and mobile devices. The VideoEngine suite consists of flexible, easy to integrate media processing frameworks that overcome the limitations of IP networks to provide a clear, consistent user experience.

  • Flexible solution delivers high quality voice and video to almost any application
  • High-level API for easy integration and accelerated time to market
  • Field-proven, best in class voice and video quality
  • Optimized to provide synchronized audio and video
  • Advanced techniques to handle jitter, packet loss, CPU and bandwidth constraints
  • Supports an array of audio and video codecs


Posted on Slashdot 6/2/2011:
Google WebRTC: Can It Replace Skype?
"Google WebRTC, all open source, is part of the web revolution that allows one browser to talk directly to another without the need for a server getting involved. WebRTC is an API that used the new P2P web API to allow developers to implement audio and video communications using direct P2P links between browsers. This really is a game changer."

And, while this feature doesn't seem to have gotten a lot of attention so far, Google Voice can call landline and cell phones for a small fee, just like Skype.



The GIPS wideband iSAC codec was used for the great audio quality on Skype calls before they parted ways in 2007 with the Skype 3.2 release.


There was a post to the main Skype blog that included this (my emphasis in bold):
And because we’ve replaced our audio engine in our most recent releases — it’s now fully built in-house — it’s worth bearing in mind that you may run into some bumps when a call is placed from an older version of Skype to newer versions.

Looks like Skype really shot them selves in the foot on this one. With Google opensourcing it with a BSD style license soon Skype may be history.
Which may explain why they sold off to Microsoft reciently for $8.5 billion.

Tester(591) on Slashdot said:
The WebRTC code that was released is missing many important bits that are required to compete against Skype. The most important is probably a bandwidth management engine, the code that's currently public just sends at a pre-configured bitrate. That means it can only do low resolution video with a shitty quality.
That said, Google Talk in GMail and Android have a dynamic bitrate stuff, and I expect they will be released at some point. I should also mention that Farsight2/Farstream using in Empathy and Pidgin are currently gaining the same kind of bandwidth management that Google is doing. So we should get at least two independent open implementations soon.

WebRTC FAQ


Why Google bought Global IP Solutions ZDNet 5/2010


I did an article on my other blog on this:
WebRTC - bringing real time communications to the web.

2 comments:

C said...

seems to me that if its released as open bsd license Skype could also start using it again.... So it isnt an advantage "against" Skype, but lowers the barrier of entry to other products.

Kristian Kielhofner said...

Skype designed and built SILK in house. There's already an IETF draft for SILK and OPUS is coming along as well. As far as audio codecs go you don't get more advanced than SILK. It's what makes the Skype experience Skype.