...

URL:http://localhost:8081/apps/EchoApp/playRTSP
OBJECT:
{
  "nodeId" : "Qb3rAjf3lzoy6PEl1WZkUhRG1DsTykgj@192NTk1tLorQ00llGbPJuFexrKceubGCR0k@192.168.1.5",
  "appKey" : "defaultApp",
  "sessionId" : "/192.168.1.1100:3531659711/192.168.1.5:554",
  "mediaSessionId" : "29868390-73ee-4f49-ba92-78d717c53070-test-RTSP",
  "rtspUrlname" : "rtsp://testp11.flashphoner.com:554/test",
  "User-AgentmediaProvider" : "RTSP",
  "userAgent" : "LibVLC/3.0.4 (LIVE555 Streaming Media v2016.11.28)"
}

...

Thus, RTSP client can be authenticated by RTSP stream URL, User-Agent, client and server IP address and port.

Adjusting RTSP playback parameters

To adjust RTSP playback parameters, for example, to change audio or video codec, SDP setting file rtsp_server.sdp should be used. Note that this file should contain WCS server IP address.

Known issues

1. Frame loss and picture artefacts can occur when HD stream is played via RTSP 

Symptoms: some artefacts are observed, and player log contains lost frame reports when HD stream is played via RTSP.

Solution: switch player to RTSP interleaved mode, for example, in VLC settings tab Input/Codecs set radiobutton Live 555 stream transport to RTP over RTSP (TCP)

2. Freezes can occur when WebRTC stream is played via RTSP, if player recieves no key frame

Symptoms: freezes when WebRTC stream is played as RTSP in VLC player

Solution: enable the following setting in flashphoner.properties file

periodic_fir_request=true

3. When stream is played as RTSP in VLC on Windows, audio samplerate and bitrate can be displayed incorrectly due to VLC known bug.