mediamtx

Commit Graph

Author	SHA1	Message	Date
Alessandro Ros	4553fc267c	hls, webrtc: in web page, add shadow to messages (#2959 )	1 year ago
Alessandro Ros	b32bc8dee9	hls, webrtc: in web page, prevent video from overflowing (#2958 )	1 year ago
Alessandro Ros	0433af66a3	hls, webrtc: in the web page, show connection errors to users (#2957 )	1 year ago
Alessandro Ros	94953f5d22	webrtc: support Chrome versions older than M72 (#2621 ) (#2814 )	1 year ago
Alessandro Ros	11988249df	move servers into internal/servers (#2792 )	1 year ago
Rui Lopes	9f5169ba26	webrtc: support WHIP/WHEP DELETE method (#2507 )	2 years ago
Alessandro Ros	412a48ac74	webrtc: fix reading Opus stereo tracks with Chrome (#2043 ) (#2470 )	2 years ago
Alessandro Ros	f1e28d053f	webrtc: fix ETag header name according to specification (#2396 ) (#2401 )	2 years ago
Daniel	36e608902d	Add video player options via query string (#2145 ) * Add video player options via query string. * use arrow-style functions * remove redundant check --------- Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com>	2 years ago
Alessandro Ros	1a748bb971	webrtc: allow using special characters in ICE server credentials (#1953 ) (#2000 )	2 years ago
Alessandro Ros	20a3b07d0a	webrtc: move codec and bitrate settings on client side (#1990 )	2 years ago
Alessandro Ros	2faca73749	hls, webrtc: in the web page, pass query parameters to inner requests (#1976 )	2 years ago
Alessandro Ros	815fa0586a	webrtc: fix using inline credentials in URLs (#1919 ) (#1966 )	2 years ago
Alessandro Ros	99aa0d0ac9	webrtc: fix WHIP/WHEP implementation (#1857 ) (#1861 ) offers and answers are now encoded in SDP in place of JSON; Location header is set by the server. This fixes compatibility with GStreamer and whipsink	2 years ago
Alessandro Ros	6b8c65f5d7	webrtc: fix exception in browser when webrtcICEServers is empty (#1817 ) (#1821 ) this fixes a regression introduced in v0.23.0.	2 years ago
Alessandro Ros	a14246d776	webrtc: support publishing with WHIP and reading with WHEP (#1800 )	2 years ago
Alessandro Ros	1688e5d2e5	support publishing with WebRTC (#1659 ) (#1786 )	2 years ago
kkocdko	1c70d627da	add viewport meta to hls and webrtc page (#1653 )	2 years ago
aler9	b20abbed6c	webrtc muxer: keep the WebSocket connection The WebSocket connection is kept open in order to use it to notify shutdowns.	2 years ago
aler9	777860f757	webrtc muxer: fix freeze on Firefox	2 years ago
aler9	bab5caee01	webrtc: fix connection in case of high latency When latency is high, one side of the peer connection switched to the "connected" state before the other one, and then closed the WebSocket connection since it's useless after the peer connection has been established. This caused the other side of the connection to detect a WebSocket error and to exit. The WebSocket connection must remain open, otherwise the "connected" state is not set by both parts.	2 years ago
aler9	0e68aedf12	webrtc: fix support for video + audio	2 years ago
aler9	ac371d8dca	remove sendrecv comment	2 years ago
Alessandro Ros	5efe97abf1	Support reading with WebRTC (#1242 )	2 years ago

5 Commits (f5d9fde60499c1a3ba51f91166f88753d9146256)