- From: Ruslan Burakov via GitHub <sysbot+gh@w3.org>
- Date: Fri, 22 Mar 2019 14:31:23 +0000
- To: public-webrtc@w3.org
kuddai has just created a new issue for https://github.com/w3c/webrtc-pc: == Add setTargetJitterBufferDelay method to RTCRtpReceiver == Following discussion [here](https://github.com/w3c/webrtc-pc/issues/2109), can we add a new method to the `RTCRtpReceiver`, called `setTargetJitterBufferDelay`. It would allow to provide a preference for the user agent for the target jitter buffer delay, but depending on the congestion control and network condition actual delay may differ. The justification is essentially the same as in previous [issue](https://github.com/w3c/webrtc-pc/issues/2109). This extensions would open a room for quality improvements for both audio and video media sourced by an RTCPeerConnection as you will have additional time until the media rendered. For example, in the case of network issues you will have more video samples to play and recover. Or if the packets are lost then you will have more time for packets retransmission. Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/2139 using your GitHub account
Received on Friday, 22 March 2019 14:31:24 UTC