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: ~ orca.js: open real-time communications API
WebRTC promises to greatly simplify the development of multimedia realtime communications, without the need to install an application or browser plug-in. It enables this by exposing a media engine and the network stack through a set of specialised APIs. Application developers can use these APIs to easily add realtime communication to web applications. The defined use cases that have been identified in the early stages of the standardisation have a wide range of applicability. Many of the initial applications focus on communications between applications running on web browsers, but it is expected that WebRTC will also allow for interoperability with several existing communications protocols. For that and other reasons, the signalling interfaces have been determined to be out of scope, thus allowing the service providers to choose the best suited protocol or signalling mechanism for their users.
webrtcH4cKS: ~ What is a WebRTC Gateway anyway? (Lorenzo Miniero)
As I mentioned in my ‘WebRTC meets telecom’ article a couple of weeks ago, at Quobis we’re currently involved in 30+ WebRTC field trials/POCs which involve in one way or another a telco network. In most cases service providers are trying to provide WebRTC-based access to their existing/legacy infrastructure and services (fortunately, in some cases it’s not limited to do only that). To achieve all this, one of the pieces they need to deploy is a WebRTC Gateway. But, what is a WebRTC Gateway anyway? A year ago I had the chance to provide a first answer during the Kamailio World Conference 2013 (see my presentation WebRTC and VoIP: bridging the gap) but, since Lorenzo Miniero has recently released an open source, modular and general purpose WebRTC gateway called Janus, I thought it would be great to get him to share his experience here.