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• IP cameras
• Media servers
• Surveillance systems
• Conference servers

Supported codecs

• H.264
• H265 (since 5.2.1579)
• VP8
• AAC
• G.711
• Speex

Supported platforms and browsers

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h264_check_and_skip_annexb=true

Since build 5.2.946, this parameter is removed from settings, and AnnexB frames are detected and played automatically.

Audio codecs exclusion

Sometimes it is necessary to capture a stream from camera without audio, or disable some audio codecs to escapr sound transcoding. This can be done with the following parameter listing audio codec names to be excluded while capturing stream, for example

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If the name is set to the RTSP stream pulled, this stream can be played in CDN by its name (by default, CDN playback is not available for RTSP streams because they always are pulled locally).

Known issues

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Capturing H265 RTSP stream

Since build 5.2.1579 it is possible to capture RTSP stream published by camera as H265. To enable this, H265 should be added to supported codecs list

codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv,h265

and to exclusion lists

codecs_exclude_sip=mpeg4-generic,flv,mpv,h265
codecs_exclude_sip_rtmp=opus,g729,g722,mpeg4-generic,vp8,mpv,h265
codecs_exclude_sfu=alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,flv,mpv,h265

Stream captured may be played as WebRTC, RTMP, MSE, HLS with transcoding and as RTSP without transcoding

A first subscriber issue

Before WCS build 5.2.1760 RTSP streams may start to play with a huge delay for the first subscriber. This was due to key frames missing caused by subscriber thread starting after publisher thread. Since build 5.2.1760 the behavior was changed: publisher thread starts after subscriber thread. This may be reverted if necessary by the following parameter

agent_use_subscriber_listener=false

Stream timestamp fix

In some RTSP streams a frame timestamps may be in wrong order, for example two subsequent frames may have the same timestamp. Such stream may not be displayed or periodically display a gray square while playing via WebRTC. Since build 5.2.1794 the following parameter is availbel to fix a broken timestamps

jitter_buffer_attempt_to_correct_broken_timestamp=true

In this case RTSP capturing client log may contain a messages as follows

Non-monotonous timestamp in input stream; previous: 453424, current: 453424; changing to 453425. This may result in incorrect timestamps in the output

and the problem stream should play normally.

Known issues

5. Connection to the IP camera is lost on error in any track (audio or video)

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streams_synchronization=camera1/-21800;camera2/2079600704

Solution: in build before 5.2.1775 increase synchronization buffers for audio and video tracks

audio_incoming_buffer_size=100
video_incoming_buffer_size=100

since build 5.2.1775 increase force synchronization timeout for audio and video tracks

video_force_sync_timeout=1000
audio_force_sync_timeout=1000