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RTSP-codecs

• Video: H.264, VP8, H265 (since build 5.2.1577)
• Audio: AAC, G.711, Speex

Operation flowchart

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1. The software player establishes a connection to the WCS server via RTSP.
2. The software player receives the media stream from WCS.

RTSP server configuration

By default, TCP port 554 is used to listen RTSP connections. This value can be set with the following parameter in flashphoner.properties file

rtsp.port=554

Since build 5.2.801, WCS is running from 'flashphoner' user for security reasons. Therefore RTSP server may not be launched because TCP ports in range 0-1000 are privileged and unavailable to non-root users. In this case RTSP port should be changed, for example

rtsp.port=5554

RTSP playback authentication via REST hook

RTSP playback authentication via REST hook acn  can be set up if necessary. To do this, the following parameter should be set in flashphoner.properties file:

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When RTSP connection is established, /playRTSP query will be sent to backend server application defaultApp

URL:http://localhost:8081/apps/EchoApp/playRTSP
OBJECT:
{
  "nodeId" : "NTk1tLorQ00llGbPJuFexrKceubGCR0k@192.168.1.5",
  "appKey" : "defaultApp",
  "sessionId" : "/192.168.1.100:59711/192.168.1.5:554",
  "mediaSessionId" : "29868390-73ee-4f49-ba92-78d717c53070-test-RTSP",
  "name" : "rtsp://p11.flashphoner.com:554/test",
  "mediaProvider" : "RTSP",
  "userAgent" : "LibVLC/3.0.4 (LIVE555 Streaming Media v2016.11.28)"
}

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

Custom access key and backend application usage for RTSP playback authentication

Since build 5.2.1008 it is possible to set ccustom authentication key (token) in RTSP URL, for example

rtsp://wcs:5554/streamName?aclAuth=1254789

The following REST hook /playRTSP content will be sent to defaultApp backend application

{
  "nodeId" : "XLepaP08Uyz9LqAjXHWnwuFxrEri0fCj@192.168.1.39",
  "appKey" : "testApp",
  "sessionId" : "/192.168.1.83:55195/192.168.1.39:5554",
  "mediaSessionId" : "71317dfc-0222-4acd-912e-57e67f2a272a-streamName-RTSP",
  "name" : "rtsp://wcs:5554/streamName?aclAuth=1254789",
  ...
  "mediaProvider" : "RTSP",
  "userAgent" : "LibVLC/3.0.8 (LIVE555 Streaming Media v2016.11.28)",
  "custom" : {
    "aclAuth" : "1254789"
  }
}

Authentication token name is set by the following parameter as like for HLS playback

client_acl_property_name=aclAuth

It is also possible to set custom backend application key

rtsp://wcs:5554/streamName?appKey=customAppKey&aclAuth=1254789

In this case REST hook /playHLS will be sent to backend application with defined key (customAppKey in the example above).

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.

Playing H265 without transcoding

Since build 5.2.1577 it is possible to play MPEG-TS H265 stream via RTSP. To do this, H265 codec must be set in rtsp_server.sdp file:

v=0
o=- 1988962254 1988962254 IN IP4 0.0.0.0
c=IN IP4 0.0.0.0
t=0 0
a=sdplang:en
a=range:npt=now-
a=control:*
m=audio 0 RTP/AVP 96
a=rtpmap:96 mpeg4-generic/48000/2
a=fmtp:96 profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexDeltaLength=3
a=control:audio
a=recvonly
m=video 0 RTP/AVP 119
a=rtpmap:119 H265/90000
a=control:video
a=recvonly

Dynamic codec detection

Since build 5.2.1592 codecs are detected dynamically for RTSP playback

v=0
o=- 1988962254 1988962254 IN IP4 0.0.0.0
c=IN IP4 0.0.0.0
t=0 0
a=sdplang:en
a=range:npt=now-
a=control:*
m=audio 0 RTP/AVP
m=video 0 RTP/AVP

In this case, if a stream is published as H264, VP8 or H265, and RTSP client supports the codec, the stream will be played via RTSP without transcoding. Audio codecs are detected by the same way.

Interleaved mode support

Before build 5.2.1609, WCS supported interleaved mode only, in this case both RTSP signaling and RTP traffic go via TCP, therefore some player applications (including VLC) may not play RTSP from WCS with default settings. Since build 5.2.1609, non-interleaved mode is also supported, in this case RTSP signaling goes via TCP, and RTP traffic flows via UDP. Note that non-interleaved mode is less packet loss proof.

Known issues

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

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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.