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  Sorenson Spark + 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 Sorenson Spark + audio Speex 16 kHz

Operation flowchart

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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.

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Call flow

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

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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:

...

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"
}

...

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"
}
}

...

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.

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Using RTMP connection timeouts

In some cases, if RTMP encoder does not support Keep Alive packets sending, or Keep Alives are disabled due to another reason with the following parameter

keep_alive.algorithm=NONE

it is necessary to control RTMP connection and close it when no data was transmitted for a long time. To do this, use the following parameters.

Read timeout

Read timeout is set with the following settings in flashphoner.properties file:

rtmp.server_read_socket_timeout=120

In this case RTMP connection will be closed if no data was received in last 120 seconds.

Write timeout

Write timeout is set with the following setting

rtmp.server_write_socket_timeout=120

In this case RTMP connection will be closed if no data was sent in last 120 seconds.

Read and write timeout

Read and write timeout is set with the following setting

rtmp.server_socket_timeout=120

In this case RTMP connection will be closed if no data was received and sent in last 120 seconds.

Known issues

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

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

5. 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

7. When stream published with RTMP encoder is played as HLS, freezes may occur if GOP is not multiple of FPS of file published

Symptoms: freezes occur when RTMP stream is played as HLS

Solution: in RTMP encoder settings, assign GOP to value equal or multiple of FPS of file published. For example, when publishing file with FPS 25 set GOP to 50.