This page contains the Updates for the First Edition (September 2012) of the book "WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web". There are three types of updates: corrections, omissions, and new developments.
Internet draft draft-ietf-codec-opus has just been published as RFC 6716 "Definition of the Opus Audio Codec". This updates pages 97, 102, and 104, and Table 6.2.
The audio parts of draft-cbran-rtcweb-codecs have been adopted as a Working Group item by the RTCWEB Working Group and have been published as draft-ietf-rctweb-audio. This updates page 97 and would add a new row to Table 6.1.
The WebRTC specification now contains a new statistics reporting capability on RTCPeerConnection. The new getStats() method uses a callback to return a new RTCStatsElement structure for the given MediaStreamTrack. This structure contains, for both the local and remote ends of the track, an RTCStatsResult object whose getValue method can be used to query the value of any statistic for that track. Although a variety of IETF protocols have standardized network statistics that can be reported, the WebRTC specification does not have yet have a registry defining the set that must be supported for WebRTC. This subject will be discussed more fully in the next edition.
The WebRTC specification now contains a new identity verification capability on RTCPeerConnection. Although DTLS, SRTP, and other protocols already in use in WebRTC can ensure that data is secured from the local brawser to the remote brawser, they do not guarantee that the individual user at the remote end is the one the local end expects. To address this, the specification now provides a mechanism for verifying the remote party is the expected one. It does this via an Identity Provider, a known site with a protocol for allowing a named user to log in or otherwise verify the user's identity. The Identity Provider can be set in the browser itself or via RTCPeerConnection.setIdentityProvider(). If either is set, an identity will be requested whenever createOffer() or createAnswer() is called. However, the process of obtaining an assertion can be started before any SDP is generated by calling RTCPeerConnection.getIdentityAssertion(). Once verified, the identity will be saved in RTCPeerConnection.peerIdentity and a negotiationneeded event will be fired.
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