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								 Alessandro Ros | f07886db5f | print the reason why a source is started or stopped (#2322) 
		
	
	
		
			
				
	
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				 | 2023-09-09 23:37:56 +02:00 |  | 
			
				
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								 Alessandro Ros | 95baade478 | srt: fix memory leak during reader disconnection (#2273) 
		
	
	
		
			
				
	
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				 | 2023-08-31 23:01:47 +02:00 |  | 
			
				
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								 Alessandro Ros | 5fb7f4e846 | force all readers to use an asynchronous writer (#2265) needed by #2255 | 2023-08-30 11:24:14 +02:00 |  | 
			
				
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								 Alessandro Ros | b72f3577c8 | print warning when the write queue is full (#2251) 
		
	
	
		
			
				
	
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				 | 2023-08-26 19:45:10 +02:00 |  | 
			
				
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								 Alessandro Ros | cf86dbb303 | switch to gortsplib/v4 (#2244) | 2023-08-26 18:54:28 +02:00 |  | 
			
				
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								 Alessandro Ros | bf8e69ea89 | rename readBufferCount into writeQueueSize (#2248) | 2023-08-26 13:25:21 +02:00 |  | 
			
				
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								 Alessandro Ros | dd91abae9b | api: add transport to RTSP sessions (#2151) | 2023-08-05 17:10:48 +02:00 |  | 
			
				
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								 Alessandro Ros | bc3084ae7b | support proxying WebRTC streams (#2142) 
		
	
	
		
			
				
	
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				 | 2023-08-03 23:12:05 +02:00 |  | 
			
				
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								 Alessandro Ros | 72b1d233df | normalize channels and methods (#2127) 
		
	
	
		
			
				
	
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				 needed by #2068 | 2023-07-30 23:53:39 +02:00 |  | 
			
				
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								 Alessandro Ros | e3d4856b4f | update gortsplib (#2126) | 2023-07-30 23:39:24 +02:00 |  | 
			
				
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								 Alessandro Ros | db3862cf0d | move stream in a dedicated package (#2121) needed by #2068 | 2023-07-30 22:34:35 +02:00 |  | 
			
				
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								 Alessandro Ros | b42154fa6a | return an error in case the random number generator fails (#2120) | 2023-07-30 22:30:41 +02:00 |  | 
			
				
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								 Alessandro Ros | 9b491499bc | webrtc: speed up track detection (#2105) 
		
	
	
		
			
				
	
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				 | 2023-07-24 20:32:28 +02:00 |  | 
			
				
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								 Alessandro Ros | 1fa53b49d4 | webrtc, hls: prevent brute-force attacks by waiting before sending responses (#2100) 
		
	
	
		
			
				
	
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				 | 2023-07-23 20:18:58 +02:00 |  | 
			
				
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								 Alessandro Ros | 0137734294 | webrtc, hls: show IP in logs in case of failed authentication (#2099) | 2023-07-23 20:06:16 +02:00 |  | 
			
				
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								 Alessandro Ros | 36298f8bc8 | webrtc: send session ID to external auth server (#1981) (#2098) | 2023-07-23 19:31:34 +02:00 |  | 
			
				
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								 Alessandro Ros | af23609d47 | api: fix crash when calling /v1/webrtcsessions/list just after session creation (#2097) | 2023-07-23 18:40:06 +02:00 |  | 
			
				
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								 Alessandro Ros | 473c075d89 | webrtc: fix memory leak during shutdown or session kick (#2079) | 2023-07-19 12:31:50 +02:00 |  | 
			
				
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								 Alessandro Ros | 22b120ef22 | update list of supported codecs inside error messages (#2058) (#2073) | 2023-07-19 00:14:50 +02:00 |  | 
			
				
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								 Alessandro Ros | 5066ba403c | webrtc: fix race condition that caused random crashes during handshake (#2072) | 2023-07-18 23:41:52 +02:00 |  | 
			
				
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								 Volodymyr Borodin | 47317ea8e5 | api: add path to RTMP connections, RTSP sessions, WebRTC sessions (#1962) (#2022) * api: add path to rtmp response
* add 'path' to RTSP and WebRTC sessions too
* add tests
---------
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com> | 2023-07-05 21:20:26 +02:00 |  | 
			
				
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								 Alessandro Ros | 1a748bb971 | webrtc: allow using special characters in ICE server credentials (#1953) (#2000) | 2023-06-30 16:47:10 +02:00 |  | 
			
				
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								 Alessandro Ros | 20a3b07d0a | webrtc: move codec and bitrate settings on client side (#1990) | 2023-06-27 22:37:06 +02:00 |  | 
			
				
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								 Alessandro Ros | 79ee4e06f3 | webrtc: add option to disable audio effects (#1908) (#1989) | 2023-06-27 22:36:29 +02:00 |  | 
			
				
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								 Alessandro Ros | 4aef466103 | webrtc: allow setting Opus bitrate (#1908) (#1985) | 2023-06-27 22:15:50 +02:00 |  | 
			
				
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								 Alessandro Ros | 6663f7b474 | webrtc: forbid publishing zero tracks (#1991) | 2023-06-27 13:58:06 +02:00 |  | 
			
				
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								 Alessandro Ros | fb1f8ff81d | webrtc: fix bitrate not being applied (#1984) | 2023-06-24 18:35:19 +02:00 |  | 
			
				
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								 Alessandro Ros | c46d2156b6 | webrtc: fix memory leak when publishing or reading (#1884) (#1983) | 2023-06-24 13:30:36 +02:00 |  | 
			
				
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								 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 | 2023-05-24 17:06:06 +02:00 |  | 
			
				
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								 Alessandro Ros | b93eed64bc | api: add /get endpoints (#1577) (#1823) this allows to get entities by ID or name after /list endpoints were
changed in v0.23.0. | 2023-05-18 15:07:47 +02:00 |  | 
			
				
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								 Alessandro Ros | 503a131097 | webrtc: return 404 when a stream is not present (#1805) | 2023-05-16 18:01:05 +02:00 |  | 
			
				
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								 Alessandro Ros | 39c072edd6 | change repository owner (#1801) | 2023-05-16 16:14:20 +02:00 |  | 
			
				
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								 Alessandro Ros | a14246d776 | webrtc: support publishing with WHIP and reading with WHEP (#1800) | 2023-05-16 15:59:37 +02:00 |  |