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.
While we’ll share here results from this weeks discussions, last week I had the privilege to co-run a webinar on WebRTC standardization with Amir Zmora. This Webinar was hosted by Upperside’s WebRTC expo and sponsored by webrtcHacks and Quobis. The slides are here and the recording from the webinar can be seen here.
Besides this, I’m happy to announce on August I’m running a webinar on IMS/VoLTE & Telco WebRTC (in Spanish) and on September I’m co-running a WebRTC workshop in London with Tsahi, our latest addition to the webrtcHacks team.
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