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[webrtc-pc] RTCPriorityType is not documented at all

From: Iñaki Baz Castillo via GitHub <sysbot+gh@w3.org>
Date: Mon, 11 Jun 2018 09:21:25 +0000
To: public-webrtc@w3.org
Message-ID: <issues.opened-331108678-1528708883-sysbot+gh@w3.org>
ibc has just created a new issue for https://github.com/w3c/webrtc-pc:

== RTCPriorityType is not documented at all ==
When using simulcast, N encodings must be provided to `RtpSender.send()`. Each encoding may have a different `priority` field with value "very-low", "low", "medium" or "high".

As per https://w3c.github.io/webrtc-pc/#dom-rtcprioritytype:

> low
> See [RTCWEB-TRANSPORT], Section 4. Corresponds to "normal" as defined in [RTCWEB-DATA]

In https://tools.ietf.org/html/draft-ietf-rtcweb-transports-17#section-4:

> For media, a "media flow", which can be an "audio flow" or a "video flow", is what [RFC7656] calls a "media source", which results in a "source RTP stream" and one or more "redundancy RTP streams".  This specification does not describe prioritization between the RTP streams that come from a single "media source".

But it happens that simulcast is about RTP streams coming from a **single media source**.

So, to be perfectly clear, the meaning of `RTCPriorityType` is not documented at all. AFAIS people usually assigns "low" priority to the minor bitrate stream and "high" priority" to the largest bitrate stream without too much rationale behind.

Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/1888 using your GitHub account
Received on Monday, 11 June 2018 09:21:28 UTC

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