One of the first posts we published on this blog a year ago was a ‘A Hitchhiker’s Guide to WebRTC standardization‘. Since then, the work has certainly progressed and we have been sharing here a number of updates on the topic. This week we’re having qn IETF meeting in Canada and when it comes to WebRTC some of the topics in the agenda include ALPN (Application Layer Protocol Negotiation), STUN Consent Freshness and audio (interop with legacy) and video (H.264 vs VP8 as mandatory codec is NOT discussed this week but you can see VP9 and H.265 are already mentioned in the slides) requirements. Several Working Groups other than RTCWeb will also discuss this week topics that are relevant to the WebRTC effort (e.g. MMUSIC WG). During the STRAW WG session, the working group I co-chair at the IETF, we’ll discuss some features of interest for those implementing WebRTC in servers like MCUs, Application Servers, WebRTC-SIP gateways or WebRTC-enabled SBCs: DTLS-SRTP, STUN and RTCP traversal/termination are examples.
webrtcH4cKS: ~ A Hitchhiker’s Guide to WebRTC standardization
Next week the IETF 87th standardization meeting will take place in Berlin, Germany. Most of the sessions I’m planning to attend are related to SIP, Diameter and of course WebRTC. When a week ago I started preparing some material for the meeting, a customer called and asked me to provide a training session on WebRTC standardization and implementation status for their R&D team. While this is something I’m planning to do in the next month, I thought I could start my contribution to this blog by providing a brief introduction to WebRTC standards and describe what’s going on in each group. This introductory post is meant to provide a very initial overview and I’m planning to go into technical details in future blog entries.