As discussed in previous posts, WebRTC standards do not specify a signaling protocol. In general this decision is positive by giving developers the freedom to select (or invent) the protocol that best suits the particular WebRTC application’s needs. This can also reduce the time to market since standards compliance-related tasks are minimized. WebRTC media and data protocols from the provider to the user are standardized, so the lack of a standardized signaling protocol does not hurt interoperability of subscribers within the same network service. The calling party just has to have a URL from the called party to download its web app and to establish a WebRTC session with them. They both connect to the same web server and will then utilize the same signaling scheme. This is a new paradigm that is often difficult to embrace for traditional telephony developers who are used using the SIP protocols for handling all signaling, including the User to Network Interface (UNI) and Network-to-Network Interface (NNI).
webrtcH4cKS: ~ The IMS approach to WebRTC
The first post we published on webrtcHacks was ‘A Hitchhiker’s Guide to WebRTC standardization’ (July 2013) where we gave some initial insight on activities in the 3GPP around WebRTC and IMS. Since then the situation has certainly evolved (well, probably not as fast as some would have expected). Since we regularly receive emails asking about the status/progress on WebRTC standardization within the 3GPP, we spent some time with our friend Antón Román, CTO at Quobis and author of the popular post ‘Anatomy of a WebRTC SDP’ to summarize the current status of the ‘WebRTC access to IMS’ effort.
Last week I attended the Illinois Institute of Technology Real-Time Communications (IIT-RTC) Conference in Chicago. This event has a history of attracting key players from around the RTC world. It features discussion that is distilled down to the key trends and technology challenges in the industry, with very little “fluff” on top. This year the IIT-RTC conference was co-located with IPTComm as well, adding to the quality of the content.
Topics at the conference touched on many segments of RTC, including IMS, RCS, E-911, OTT, and more. Our own Victor Pascual sits on the steering committee for the Web and Emerging Technologies track, where WebRTC was given particular focus. It began with a fantastic WebRTC tutorial from Alan Johnston (co-author of the SIP specification and a dozen other IETF RFCs) and Dan Burnett (co-editor of the W3C WebRTC specification). They are also both co-authors of “WebRTC: APIs and RTCWeb Protocols of the HTML5 Real-Time Web”, and provided a fantastic expert introduction to WebRTC APIs and methodologies. This set the tone for lots of excellent presentation, expert perspective, demonstrations, and discussion on WebRTC over the next few days. Here are some discussions I found particularly interesting:
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.