- From: Bernard Aboba via GitHub <sysbot+gh@w3.org>
- Date: Thu, 06 Jun 2024 00:50:16 +0000
- To: public-webrtc@w3.org
aboba has just created a new issue for https://github.com/w3c/webrtc-rtptransport: == RtpTransport and transport re-direction == One potential use case for RtpTransport would be to capture the RTP packets that would have been generated by WebRTC, and then redirect them over another transport (such as [WebTransport](https://w3c.github.io/webtransport/) or [P2P WebTransport](https://w3c.github.io/p2p-webtransport/)). Such a use case would require the following: - Ability to capture the RTP packets that would have been sent - Ability to turn off RTP congestion control (since CC is handled by QUIC) - Ability to specify the desired MTU for an audio or video stream - - Decrease MTU of audio packets slightly to allow for additional QUIC datagram overhead - - Use infinite MTU for video packets when transporting them over a QUIC reliable stream Please view or discuss this issue at https://github.com/w3c/webrtc-rtptransport/issues/48 using your GitHub account -- Sent via github-notify-ml as configured in https://github.com/w3c/github-notify-ml-config
Received on Thursday, 6 June 2024 00:50:17 UTC