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UC: WebRTC #20

Open xxyzzzq opened 7 years ago

xxyzzzq commented 7 years ago

From @mounirlamouri on June 29, 2015 15:55

AFAIK, WebRTC output is usually piped to a <video> that means that the element will be part of a MediaSession of some sort. However, even if it's close to content, it's not really content because it wouldn't make sense to pause the video when there is a notification or have a pause button in the device's lock screen.

We should try to see how we can make that UC fit into the spec model.

Copied from original issue: WICG/mediasession#81

xxyzzzq commented 7 years ago

From @richtr on June 30, 2015 10:58

FWIW, transient-solo probably makes the most sense for a WebRTC-based media session right now. However, that is not really ideal for telephony longer-term since other transient-solo content could then still pause those media elements (without pausing its associated media stream).

Related: there was some discussion on this issue yesterday in the #WHATWG IRC channel http://krijnhoetmer.nl/irc-logs/whatwg/20150629#l-186.

We could always add a 'telephony' media session category. Alternatively we could add special handling when mediaElement.srcObject is a MediaStream object or when mediaElement.src is a Blob object generated from a MediaStream object.

Perhaps ideally WebRTC could start returning RTCMediaStream objects so we can distinguish WebRTC-originating MediaStream objects from other non-WebRTC sources of media streams. Then we could provide telephony with a suitable session only when it really is a telephony-generated media stream.

interface RTCMediaStream : MediaStream {}
xxyzzzq commented 7 years ago

From @foolip on July 1, 2015 16:30

From the IRC discussion with @annevk I can see two main points:

  1. Telephony has higher priority on some systems, certainly on iOS and Firefox OS, but "even" Android has PhoneStateListener which makes it possible for others to know that a phone call is ongoing.
  2. If a phone call is ongoing, you might want the incoming call sound to be different.