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Extending Video to the Web through Open Source H.264

Cisco has been a champion of video communications for a very long time. We are committed to seeing video communications from board room to cubicle, and from CEO to intern. To achieve this vision, we’ve been investing in video solutions from top-end immersive telepresence to video capable soft clients like Jabber. Unfortunately, the one place we haven’t been able to fully go is the web. Video communications is not possible natively in the browser – yet. Work has been progressing on addressing this through an extension to HTML5 called WebRTC. However, this activity has hit a speed bump due to disagreements on choosing a video codec for the browser. Cisco and many others support H.264, which is the foundation of our products and those of most of our competitors.

Today, Cisco has taken a bold step to bringing video to the web. We plan to open-source our H.264 codec, and to provide it as a binary module that can be downloaded for free from the Internet. Cisco will not pass on our MPEG LA licensing costs for this module, and based on the current licensing environment, this will effectively make H.264 free for use in WebRTC. Furthermore, Mozilla has announced it will enable Firefox to utilize this module, bringing real-time H.264 support to their browser. This is an exciting step forward, as it will enable our products and those of the industry at large to move to the web and reduce further the barriers to open communications.

Read more about it in Rowan’s latest blog post: http://blogs.cisco.com/collaboration/open-source-h-264-removes-barriers-webrtc

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2 Comments.


  1. Its nice to hear that Cisco opens it Video Codec to public. But what is with audio codecs?
    AFAIK WebRTC prefers Opus and G.711. G.711 is compatible to a lot of VC systems, but has a horrible sound quality. Opus is great, but Cisco isn’t support it in their VC systems.
    So, what is the way for Cisco to go with audio and WebRTC?

       0 likes

  2. October 30, 2013 at 10:13 am

    Great question Juergen. It’ll be G.711 from webRTC for now, down the road we’re looking to get Opus more widely implemented. Opus is new for everyone including us.

       0 likes