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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

+

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iOS

-

-

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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",
"rtmpTransponderFullUrl": false
"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

rtmpFlashVersionRTMP subscriber Flash versionLNX 76.219.189.0

options

Transponder options

{"action": "mute"}

mediaSessionId

Unique identifier of the transponder

eume87rjk3df1i9u14elffga6t

width

Image width

320

height

Image height

240

keyFrameIntervalVideo keyframe interval60
fpsVideo framerate30

muted

Is sound muted

true

soundEnabled

Is sound enabled

true

soundFile

Sound file

test.wav

loop

Loop playback

false

rtmpTransponderFullUrlTake stream name to publish  to RTMP server from RTMP URLfalse

Parameters added since build 5.2.785: rtmpFlashVersion, keyFrameInterval and fps.

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

Since build 5.2.560, 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
}

transcoding will not be enabled for stream republishing.

If picture height is 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 defined resolution.

Specified width is applied only if picture aspect ratio preserving is disabled, and height is also specified. If only width parameter is passed - without height - it is not applied, and the stream is not transcoded.

Since build 5.2.785, there are two more parameters enabling transcoding: keyFrameInterval and fps. Stream will be transcoded if either of them, or height is specified.

Set stream name to publish to RTMP server

By default, a stream will be published to RTMP server with the same name as it is publishing on WCS, and the prefix rtmp_, for example rtmp_test. This behaviour can be changed by the following parameters

rtmp_transponder_full_url=true
rtmp_transponder_stream_name_prefix=

But, these settings are applyed to all the republishings, and require server restart. That's why since build 5.2.860 the /push/startup query parameter is added to allow to define full RTMP URL, including stream name on RTMP server, regardless of server settings

POST /rest-api/push/startup HTTP/1.1
Host: localhost:8081
Content-Type: application/json

{
 "streamName":"stream1",
 "rtmpUrl":"rtmp://rtmp.flashphoner.com:1935/live/test",
 "rtmpTransponderFullUrl":true
}

In this case, the stream will be published to RTMP server with the name defined in RTMP URL even with default WCS settings.

Sending the REST query to the WCS server

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

JavaScript API

Using Web SDK 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 should 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 republishing 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.

RTMP outgoing stream buffering

Since build 5.2.700 outgoing RTMP stream can be buffered. This icreases translation latency, but allows to play the stream more smooth from destination RTMP server. Bufferization is enabled with the following parameter

rtmp_out_buffer_enabled=true

The following bufferization parameters can be tuned

ParameterDefault valueDescription

rtmp_out_buffer_start_size

300Stream buffer start size, мс

rtmp_out_buffer_initial_size

2000Stream buffer initial size, мс

rtmp_out_buffer_polling_time

50Buffer polling timeout, мс

rtmp_out_buffer_max_bufferings_allowed

-1Maximum stream bufferings allowed, unlimited by default

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();

Known issues

1. When stream is republished to RTMP server and is played from this server in JWPlayer, stream picture aspect ration can be distorted

Symptoms: playing stream aspect ratio in JWPlayer differs from published one

Solution: enable metadata sending while stream republishing as RTMP

rtmp_transponder_send_metadata=true

2. Republishing may fail if RTMP destination server requires specific Flash version

Symptoms: RTMP handshake fails, the channel is closed with RTMP error in WCS server log

Solution: specify RTMP subscriber Flash version, either using rtmp_flash_ver_subscriber setting in flashphoner.properties, or rtmpFlashVersion parameter in republishing REST request

For example, for republishing to Periscope:

rtmp_flash_ver_subscriber = LNX 76.219.189.0

3. RTMP destination server may require specific stream parameters: bitrate, keyframe interval, or framerate

Symptoms: e.g., Periscope displays warnings about not corresponding to the recommended settings

Solution: set specific constraints to the source stream (e.g., for audio bitrate) and specify required parameters in republishing REST request (keyFrameInterval and fps)

4. When republishing FullHD, 2K, 4K streams with big frame size, data packets to send may not fit to socket buffer, this leads to artifacts in some players

Symptoms: artifacts occur while playing republished RTMP stream via good channel

Solution: enable RTMP packets buffering with the parameter

rtmp.server_buffer_enabled=true