Web Call Server 5.1 - EN
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To run online broadcasts you can use special hardware or software video capturing devices (Live Encoder). Such devices or programs capture a video stream and send it to the server via the RTMP protocol.
Web Call Server 5.1 can receive an RTMP video stream from such a device or software (Wirecast, ffmpeg, OBS Studio, FMLE etc.) encoded to H.264 + AAC or  Sorensen Spark H.263 + Speex and broadcast this video stream to browsers and mobile devices.

Overview

Technical specifications

  • Receiving incoming audio- and video streams via the RTMP protocol

  • Broadcasting of the received video stream to browsers and platforms: any among ones supported by WCS

  • Uses video stream playback technologies: any among ones supported by WCS

Codec support

  • Video H.264 + audio AAC
  • Video Sorensen Spark (H.263) + audio Speex 16 kHz

Operation flowchart


1. Live Encoder establishes a connection to the server via the RTMP protocol and sends the publish command.

2. Live Encoder sends the RTMP stream to the server.

3. The browser establishes a connection via Websocket and sends the play command.

4. The browser receives the WebRTC stream and plays that stream on the page.


Quick manual on testing

1. For test we use:

  • WCS server
  • OBS Studio
  • Player web application in Chrome browser to stream playback

2. Set up RTMP strteaming to the server address, for example rtmp://test1.flashphoner.com:1935/live/, set the stream key obsStream:

3. Start streaming in OBS Studio:

4. Open Player application. Set the stream key in 'Stream' field and press 'Start' button. The stream captured playback begins.

Call flow

Below is the call flow when an RTMP stream is broadcast from an external source (Live Encoder)to the WCS server

Parsing stream URL parameters

When RTMP stream is published or played on WCS, RTMP connection and stream parameters may be set in stream URL like this:

rtmp://host:1935/live?connectParam1=val1&connectParam2=val2/streamName?streamParam1=val1&streamParam2=val2

Where

  • host is WCS server hostname;
  • connectParam1, connectParam2 are RTMP connection parameters;
  • streamName is stream name on server;
  • streamParam1, streamParam2 are stteam parameters.

WCS server passes the parameters to backend server in REST hook in custom field, for example:

Connection parameters
URL:http://localhost:8081/apps/EchoApp/connect
OBJECT:
{
"nodeId" : "Qb3rAjf3lzoy6PEl1WZkUhRG1DsTykgj@192.168.1.1",
"appKey" : "flashStreamingApp",
"sessionId" : "/127.0.0.1:5643/192.168.1.1:1935",
"useWsTunnel" : false,
"useWsTunnelPacketization2" : false,
"useBase64BinaryEncoding" : false,
"keepAlive" : false,
"custom" : {
"connectParam1" : "val1",
"connectParam2" : "val2"
},
"login" : "rQq83sodiCPY0pJXCxGO"
}
Publishing parameters
URL:http://localhost:8081/apps/EchoApp/publishStream
OBJECT:
{
"nodeId" : "Qb3rAjf3lzoy6PEl1WZkUhRG1DsTykgj@192.168.1.1",
"appKey" : "flashStreamingApp",
"sessionId" : "/127.0.0.1:5643/192.168.1.1:1935",
"mediaSessionId" : "627990f9-8fe5-4e92-bb2a-863cc4eb43de",
"name" : "stream1",
"published" : true,
"hasVideo" : false,
"hasAudio" : true,
"status" : "NEW",
"record" : true,
"width" : 0,
"height" : 0,
"bitrate" : 0,
"minBitrate" : 0,
"maxBitrate" : 0,
"quality" : 0,
"mediaProvider" : "Flash",
"custom" : {
"streamParam1" : "val1",
"streamParam2" : "val2"
}
}
Playback parameters
URL:http://localhost:8081/apps/EchoApp/playStream
OBJECT:
{
"nodeId" : "Qb3rAjf3lzoy6PEl1WZkUhRG1DsTykgj@192.168.1.1",
"appKey" : "flashStreamingApp",
"sessionId" : "/127.0.0.1:5643/192.168.1.1:1935",
"mediaSessionId" : "stream1/127.0.0.1:5643/192.168.1.1:1935",
"name" : "stream1",
"published" : false,
"hasVideo" : true,
"hasAudio" : true,
"status" : "NEW",
"record" : false,
"width" : 0,
"height" : 0,
"bitrate" : 0,
"minBitrate" : 0,
"maxBitrate" : 0,
"quality" : 0,
"mediaProvider" : "Flash",
"custom" : {
"streamParam1" : "val1",
"streamParam2" : "val2"
}
}

This feature can be used for example to authenticate client on backend server while publishing or playing RTMP-stream on WCS server.


Setting a server application while RTMP stream publishing

While publishing RTMP stream to WCS server, a server application can be set that will be used to backend server interaction. It can be done with parameter in stream URL:

rtmp://host:1935/live?appKey=key1/streamName

Where

  • host is WCS server;
  • key1 is application key on WCS server;
  • streamName is stream name to publish

By default, if application key parameter is not set, the standard application flashStreamingApp will be used.

Besides, an application can be explicitly specified as stream URL part. To do this, the following parameter in flashphoner.properties file should be set

rtmp_appkey_source=app

Then application key must be set in stream URL as

rtmp://host:1935/key1/streamName

In this case, live is also an application name, therefore when stream is published with URL

rtmp://host:1935/live/streamName

live application must be defined on WCS server.

Sorenson Spark + Speex 16 kHz stream publishing

WCS server can capture RTMP stream encoded with Sorenson Spark + Speex 16kHz to FLV container. This stream can be published, for example, using ffmpeg as follows:

ffmpeg -re -i BigBuckBunny.flv -preset ultrafast -ar 16000 -ac 1 -acodec speex -vcodec flv -strict -2 -f flv rtmp://test1.flashphoner.com:1935/live/test

Known limits

1. To handle such stream including stream recording, the stream will be transcoded to H.264 + AAC.

2. Payload types 127 for video and 97 for audio should be set in SDP when publishing such stream, for example

v=0
o=- 1988962254 1988962254 IN IP4 0.0.0.0
c=IN IP4 0.0.0.0
t=0 0
a=sdplang:en
m=video 0 RTP/AVP 127
a=rtpmap:127 FLV/90000
a=sendonly
m=audio 0 RTP/AVP 97 8 0
a=rtpmap:97 SPEEX/16000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendonly

Known issues

1. A stream containing B-frames does not play or plays with artifacts (latencies, lags)

Symptoms:

  • a stream sent by the RTMP encoder does not play or plays with latencies or lags
  • warnings in the client log:
09:32:31,238 WARN 4BitstreamNormalizer - RTMP-pool-10-thread-5 It is B-frame!

Solution: change the encoder settings so, that B-frames were not used (lower encoding profile, specify in the command line etc).

2. AAC frames of type 0 are not supported by decoder and will be ignored while stream pulled playback

In this case, warnings will be displayed in the client log:

10:13:06,815 WARN AAC - AudioProcessor-c6c22de8-a129-43b2-bf67-1f433a814ba9 Dropping AAC frame that starts with 0, 119056e500

3. Some RTMP functions does not supported and will be ignored:

  • FCSubscribe
  • FCPublish
  • FCUnpublish
  • onStatus
  • onUpstreamBase
  • releaseStream

4. Some RTMP-encoders does not support KeepAlive.

Symptoms: disconnection occurs often while stream publishing with RTMP-encoder.

Solution: switch KeepAlive off for RTMP on the server using the following parameter in flashphoner.properties file

keep_alive.enabled=websocket,rtmfp
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