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Issue 625318 link

Starred by 3 users
Status: Duplicate
Merged: issue webrtc:5079
Owner: ----
Closed: Aug 2016
Cc:
webrtc-testing@google.com
Components:
EstimatedDays: ----
NextAction: ----
OS: Windows
Pri: 2
Type: Feature



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WebRTC: Need support for REMB style bandwidth estimation for audio.

Reported by rajsripe...@gmail.com, Jul 1 2016 
UserAgent: Mozilla/5.0 (Windows NT 10.0; WOW64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/51.0.2704.103 Safari/537.36

Steps to reproduce the problem:
Chrome supports Receiver Estimated Maximum BitRate (REMB) protocol which enables receiver to communicate this value in a RTCP packet.The support for REMB is negotiated in SDP. This is currently supported only for video. We need a similar support for audio as well.

What is the expected behavior?

What went wrong?
Not having the ability to estimate the bandwidth for audio may affect its quality.

Did this work before? No 

Chrome version: 51.0.2704.103  Channel: canary
OS Version: 10.0
Flash Version: Shockwave Flash 22.0 r0

Comment 1 by kavvaru@chromium.org, Jul 5 2016

Components: Blink>WebRTC
Labels: M-54
Status: Untriaged (was: Unconfirmed)
As the issue is of type feature marking it as Untriaged to get more inputs from dev team.

Thanks,

Comment 2 by jansson@chromium.org, Aug 31 2016

Mergedinto: webrtc:5079
Status: Duplicate (was: Untriaged)
This is work in progress and is tracked in the webrtc tracker.
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