readyState="muted"

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Xander Dumaine

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Jun 23, 2016, 2:36:06 PM6/23/16
to discuss-webrtc
Can anyone clarify exactly what causes a remote stream to go readyState="muted" and muted="true"? Is it due to byte count, actively receiving, etc?

For context, we're getting streams from Jitsi with those states on the video track. And due to a chrome bug (cannot find the issue...), when the video track is muted, the audio track won't play. Trying to figure out if we're doing something wrong with jitsi, or if it's doing something wrong...

Warren McDonald

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Jun 24, 2016, 8:04:46 PM6/24/16
to discuss-webrtc
Sounds like Jitsi is doing something the browser is not ready for yet. I recently checked on the implementation of remote mute status and it was not supported and low pri in Chrome. Not sure about Firefox. Both sides of the peer connection would have to fully support this before it could be useful.

The support for readystate does not include muted yet or the events attached to the transition.

Philipp Hancke

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Jun 25, 2016, 3:44:27 AM6/25/16
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IIRC (and the reproduction I tried confirmed) that means RTCP is received while RTP is not. You should be seeing packet loss from getStats.

2016-06-23 20:36 GMT+02:00 Xander Dumaine <xander....@gmail.com>:
Can anyone clarify exactly what causes a remote stream to go readyState="muted" and muted="true"? Is it due to byte count, actively receiving, etc?

For context, we're getting streams from Jitsi with those states on the video track. And due to a chrome bug (cannot find the issue...), when the video track is muted, the audio track won't play. Trying to figure out if we're doing something wrong with jitsi, or if it's doing something wrong...

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vuse...@gmail.com

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Jan 26, 2017, 2:53:46 AM1/26/17
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Did you ever find a solution? I've run into the same issue and I'd appreciate any advice 

V User

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Jan 31, 2017, 10:54:21 AM1/31/17
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For posterity, the issue I was having was that I was sending the wrong ssrc-id in my XMPP messages, so what Jitsi was expecting didn't match what was on the RTP packets. The key clue was that in webrtc-internals the connection graph (conn-audio-0) was showing bytes transferred while the audio-recv and video-recv graphs were not seeing any bytes. 

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