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Overview

Upon request, Web Call Server converts a WebRTC audio and video stream to RTMP and sends it to the specified RTMP server. This way you can run a broadcasting from a web page to FacebookYouTube LiveWowzaAzure Media Services and other live video services.

Republishing of an RTMP stream can be made using REST queries or JavaScript API.

Supported platforms and browsers


Chrome

Firefox

Safari 11

Edge

Windows

+

+


+

Mac OS

+

+

+


Android

+

+



iOS

-

-

+



Supported codecs

  • Video: H.264
  • Audio: AAC, G.711, Speex 16

RTMP server authentication

Supported. Specify the name and password in the URL of the server, for example rtmp://name:password@server:1935/live

Operation flowchart

  1. The browser connects to the server via the Websocket protocol and sends the publish command.
  2. The browser captures the microphone and the camera and sends the WebRTC stream to the server.
  3. The REST client sends the /push/startup query from the browser.
  4. The WCS server publishes the RTMP stream on the RTMP server at the URL specified in the query.
  5. The WCS server sends the RTMP stream.

REST queries

Republishing a video stream to another server can be performed using REST queries.

A REST query must be an HTTP/HTTPS POST query in the following form:

  • HTTP: http://streaming.flashphoner.com:8081/rest-api/push/startup
  • HTTPS: https://streaming.flashphoner.com:8444/rest-api/push/startup

Where:

  • streaming.flashphoner.com - is the address of the WCS server
  • 8081 - is the standard REST / HTTP port of the WCS server
  • 8444 - is the standard HTTPS port
  • rest-api - is the required prefix
  • /push/startup - is the REST-method used

REST-methods and response statuses

REST-method

Example of REST query body

Example of response

Response statuses

Description

/push/startup

{
"streamName": "name",
"rtmpUrl": "rtmp://localhost:1935/live",
"options": {}
}
{
"mediaSessionId": "eume87rjk3df1i9u14elffga6t",
"streamName": "rtmp_name",
"rtmpUrl": "rtmp://localhost:1935/live",
"width": 320,
"height": 240,
"muted": false,
"soundEnabled": false,
"options": {}
}

409 - Conflict

500 - Internal error

Creates a transponder that subscribes to the given stream and sends media traffic to the specified rtmpUrl.


The name of the stream specified in the query can be the name of an already published stream or the name reserved when the SIP call was created (to send media traffic received from SIP).


If a transponder for the given stream and rtmpUrl already exists, 409 Conflict is returned.

/push/find

{
"streamName": "name",
"rtmpUrl": "rtmp://localhost:1935/live",
}
[{
"mediaSessionId": "eume87rjk3df1i9u14elffga6t",
"streamName": "rtmp_name",
"rtmpUrl": "rtmp://localhost:1935/live",
"width": 320,
"height": 240,
"muted": false,
"soundEnabled": false,
"options": {}
}]

404 - Transponder not found

500 - Internal error

Find transponders by a filter

/push/find_all

 

[{
"mediaSessionId": "eume87rjk3df1i9u14elffga6t",
"streamName": "rtmp_name",
"rtmpUrl": "rtmp://localhost:1935/live",
"width": 320,
"height": 240,
"muted": false,
"soundEnabled": false,
"options": {}
}]

404 - Not found any transponder

500 - Internal error

Find all transponders

/push/terminate

{
"mediaSessionId": "eume87rjk3df1i9u14elffga6t"
}

 

404 - Not found transponder

500 - Internal error

Terminate operation of the transponder

/push/mute

{
"mediaSessionId": "eume87rjk3df1i9u14elffga6t"
}

void

404 - Not found transponder

500 - Internal error

Turn off audio

/push/unmute

{
"mediaSessionId": "eume87rjk3df1i9u14elffga6t"
}

void

404 - Not found transponder

500 - Internal error

Turn on audio

/push/sound_on

{
"mediaSessionId": "eume87rjk3df1i9u14elffga6t"
"soundFile": "test.wav"
"loop": true
}

void

404 - Not found transponder

404 - No such file

500 - Internal error

Insert audio from a RIFF WAV file located in the /usr/local/ FlashphonerWebCallServer/media/ directory on the WCS server

/push/sound_off

{
"mediaSessionId": "eume87rjk3df1i9u14elffga6t"
}

void

404 - Not found transponder

500 - Internal error

Stop inserting audio from the file

Parameters

Parameter name

Description

Example

streamName

Name of the republished stream

streamName

rtmpUrl

URL of the server the stream is republished to

rtmp://localhost:1935/live

options

Transponder options

{"action": "mute"}

mediaSessionId

Unique identifier of the transponder

eume87rjk3df1i9u14elffga6t

width

Image width

320

height

Image height

240

muted

Is sound muted

true

soundEnabled

Is sound enabled

true

soundFile

Sound file

test.wav

loop

Loop playback

false

 

The options parameter can be used to turn off audio or insert audio from a file when creating a transponder.

Example,

"options": {"action": "mute"}
"options": {"action": "sound_on", "soundFile": "sound.wav", "loop": true}

Stream transcoding while republishing

If picture width and height are not set in /push/startup query parameters

{
 "streamName": "name",
 "rtmpUrl": "rtmp://localhost:1935/live"
}

or they are set to 0

{
 "streamName": "name",
 "rtmpUrl": "rtmp://localhost:1935/live",
 "width": 0,
 "height": 0
}

since build 5.2.560 transcoding will not be enabled for stream republishing.

If picture width and height are set explicitly (for example, if destination server does not accept streams below 720p)

{
 "streamName": "name",
 "rtmpUrl": "rtmp://localhost:1935/live",
 "width": 1280,
 "height": 720
}

the stream will be transcoded and pushed to destination server in resolution defined.

Sending the REST query to the WCS server

To send the REST query to the WCS server, use a REST-client.

JavaScript API

Using WebSDK you can republish a stream to an RTMP server upon creation, similar to the SIP as stream function. Usage example for this method is available in the WebRTC as RTMP web application.

webrtc-as-rtmp-republishing.html

webrtc-as-rtmp-republishing.js

1. When a stream is created, the method session.createStream() receives the parameter rtmpUrl that specifies the URL of the RTMP server that accepts the broadcast. The name of the stream is specified in compliance with rules of the RTMP server.

code:

function startStreaming(session) {
    var streamName = field("streamName");
    var rtmpUrl = field("rtmpUrl");
    session.createStream({
        name: streamName,
        display: localVideo,
        cacheLocalResources: true,
        receiveVideo: false,
        receiveAudio: false,
        rtmpUrl: rtmpUrl
        ...
    }).publish();
}

Republishing of the stream starts directly after it is successfully published on the WCS server.

Server configuration

When WCS creates an RTMP transponder it automatically adds a prefix to the republished stream as set in the flashphoner.properties file:

rtmp_transponder_stream_name_prefix=rtmp_

If the server the stream is republished to has certain requirements to the name (FacebookYoutube), this line must be commented out.

The option

rtmp_transponder_full_url=true

turns on a possibility to pass some request parameters to RTMP server.

A network interface to bind RTMP client for republishing may be set with the following parameter

rtmp_publisher_ip=127.0.0.1

In this case, RTMP will be republished to localhost only

Parameters passing in server URL

It is possible to pass some parameters to server. to which a stream should be republished. Parameters to pass are specified in server URL, e.g.

rtmp://myrtmpserver.com:1935/app_name/?user=user1&pass=pass1

or, if a stream supposed to be published to a specified instance of RTMP server application

rtmp://myrtmpserver.com:1935/app_name/app_instance/?user=user1&pass=pass1

Where

  • myrtmpserver.com is the RTMP server name
  • app_name is the application on the RTMP server name
  • app_instance is the instance name of the RTMP server application

Stream name is set in REST query /push/startup parameter 'streamName' or in corresponding stream creation option.

This is the example on RTMP connection establishing with query parameters passing

Stream name passing in server URL

In some cases, a stream publishing name shoukd be passed in the server URL. To do this, the following option must be set in flashphoner.properties file

rtmp_transponder_full_url=true

Then, the URL to publish should be set in REST query /push/startup 'rtmpUrl' parameter or in corresponding stream creation option like this:

rtmp://myrtmpserver.com:1935/app_name/stream_name

or, to publish to another application instance

rtmp://myrtmpserver.com:1935/app_name/app_instance/stream_name

In this case, 'streamName' parameter or REST query /push/startup or corresponding stream creation option is ignored.

Automatic republishing to a specified RTMP server (not for production)

WCS server can automatically republish all the published streams to a specified RTMP server. To activate this feature, set the following options in flashphoner.properties file:

rtmp_push_auto_start=true
rtmp_push_auto_start_url=rtmp://rtmp.server.com:1935/

where rtmp.server.com is RTMP server name to republish all streams from WCS.

This feature is supposed to be used for debug, not in production.

Automatic reconnection when channel is closed

When RTMP stream is published to another RTMP server, connection to this server may be interrupted and channel may be closed for some reasons (destination server restart, network problems etc). In this case automatic reconnection and RTMP stream republishsing can be enabled with the following parameter in flashphoner.properties file:

rtmp_push_restore=true

Reconnection attempts maxumum count and interval between attempts in milliseconds should also be set

rtmp_push_restore_attempts=3
rtmp_push_restore_interval_ms=5000

In this case, 3 attempts will be made to reconnect to RTMP server with 5 seconds interval. After that, reconnection stops.

Call flow

Below is the call flow when using the Two Way Streaming example to publish a stream and the REST client to send the /push/startup query:

two_way_streaming.html

two_way_streaming.js

1. Establishing a connection to the server.

Flashphoner.createSession(); code

    Flashphoner.createSession({urlServer: url}).on(SESSION_STATUS.ESTABLISHED, function (session) {
        setStatus("#connectStatus", session.status());
        onConnected(session);
    }).on(SESSION_STATUS.DISCONNECTED, function () {
        setStatus("#connectStatus", SESSION_STATUS.DISCONNECTED);
        onDisconnected();
    }).on(SESSION_STATUS.FAILED, function () {
        setStatus("#connectStatus", SESSION_STATUS.FAILED);
        onDisconnected();
    });


2. Receiving from the server an event confirming successful connection.

ConnectionStatusEvent ESTABLISHED code

    Flashphoner.createSession({urlServer: url}).on(SESSION_STATUS.ESTABLISHED, function (session) {
        setStatus("#connectStatus", session.status());
        onConnected(session);
    }).on(SESSION_STATUS.DISCONNECTED, function () {
        ...
    }).on(SESSION_STATUS.FAILED, function () {
        ...
    });


3. Publishing the stream.

stream.publish(); code

   session.createStream({
        name: streamName,
        display: localVideo,
        cacheLocalResources: true,
        receiveVideo: false,
        receiveAudio: false
        ...
    }).publish();


4. Receiving from the server and event confirming successful publishing of the stream.

StreamStatusEvent, status PUBLISHING code

   session.createStream({
        name: streamName,
        display: localVideo,
        cacheLocalResources: true,
        receiveVideo: false,
        receiveAudio: false
    }).on(STREAM_STATUS.PUBLISHING, function (stream) {
        setStatus("#publishStatus", STREAM_STATUS.PUBLISHING);
        onPublishing(stream);
    }).on(STREAM_STATUS.UNPUBLISHED, function () {
        ...
    }).on(STREAM_STATUS.FAILED, function () {
        ...
    }).publish();


5. Sending the audio-video stream via WebRTC

6. Sending the /push/startup query

http://demo.flashphoner.com:9091/rest-api/push/startup
{
 "streamName": "testStream",
 "rtmpUrl": "rtmp://demo.flashphoner.com:1935/live/testStream"
}



7. Establishing a connection via RTMP with the specified server, publishing the stream

8. Sending the audio-video stream via RTMP

9. Stopping publishing the stream.

stream.stop(); code

function onPublishing(stream) {
    $("#publishBtn").text("Stop").off('click').click(function () {
        $(this).prop('disabled', true);
        stream.stop();
    }).prop('disabled', false);
    $("#publishInfo").text("");
}


10. Receiving from the server an event confirming unpublishing of the stream.

StreamStatusEvent, status UNPUBLISHED code

   session.createStream({
        name: streamName,
        display: localVideo,
        cacheLocalResources: true,
        receiveVideo: false,
        receiveAudio: false
    }).on(STREAM_STATUS.PUBLISHING, function (stream) {
        ...
    }).on(STREAM_STATUS.UNPUBLISHED, function () {
        setStatus("#publishStatus", STREAM_STATUS.UNPUBLISHED);
        onUnpublished();
    }).on(STREAM_STATUS.FAILED, function () {
        ...
    }).publish();
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