Republishing to other RTMP server

Overview

Web Call Server may convert a WebRTC audio and video stream to RTMP and send it to the specified RTMP server on demand. This way you can run a broadcasting from a web page to Facebook, YouTube Live, Wowza, Azure 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	Edge
Windows	✅	✅	❌	✅
Mac OS	✅	✅	✅	✅
Android	✅	✅	❌	✅
iOS	✅	✅	✅	✅

Supported codecs

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

RTMP server authentication

It is 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 publishStream 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 API

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 responses

/push/startup

Create 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.

If rtmpUrl is not set, or is set incorrectly and cannot be resolved by DNS, 400 Bad request is returned.

Request example

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

{
    "streamName": "name",
    "rtmpUrl": "rtmp://localhost:1935/live",
    "rtmpTransponderFullUrl": false,
    "rtmpFlashVersion": "LNX 76.219.189.0",
    "options": {}
}

Response example

HTTP/1.1 200 OK
Access-Control-Allow-Origin: *
Content-Type: application/json

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

Return codes

Code	Reason
200	OK
409	Conflict
500	Internal error

/push/find

Find transponders by a filter

Request example

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

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

Response example

HTTP/1.1 200 OK
Access-Control-Allow-Origin: *
Content-Type: application/json

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

Return codes

Code	Reason
200	OK
404	Not found
500	Internal error

/push/find_all

Find all transponders

Request example

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

Response example

HTTP/1.1 200 OK
Access-Control-Allow-Origin: *
Content-Type: application/json

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

Return codes

Code	Reason
200	OK
404	Not found
500	Internal error

/push/terminate

Terminate the transponder

Request example

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

{
    "mediaSessionId": "eume87rjk3df1i9u14elffga6t"
}

Response example

HTTP/1.1 200 OK
Access-Control-Allow-Origin: *
Content-Type: application/json

Return codes

Code	Reason
200	OK
404	Not found
500	Internal error

/push/mute

Turn off audio

Request example

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

{
    "mediaSessionId": "eume87rjk3df1i9u14elffga6t"
}

Response example

HTTP/1.1 200 OK
Access-Control-Allow-Origin: *
Content-Type: application/json

Return codes

Code	Reason
200	OK
404	Not found
500	Internal error

/push/unmute

Turn on audio

Request example

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

{
    "mediaSessionId": "eume87rjk3df1i9u14elffga6t"
}

Response example

HTTP/1.1 200 OK
Access-Control-Allow-Origin: *
Content-Type: application/json

Return codes

Code	Reason
200	OK
404	Not found
500	Internal error

/push/sound_on

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

Request example

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

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

Response example

HTTP/1.1 200 OK
Access-Control-Allow-Origin: *
Content-Type: application/json

Return codes

Code	Reason
200	OK
404	Not found
500	Internal error

/push/sound_off

Stop inserting audio from the file

Request example

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

{
    "mediaSessionId": "eume87rjk3df1i9u14elffga6t"
}

Response example

HTTP/1.1 200 OK
Access-Control-Allow-Origin: *
Content-Type: application/json

Return codes

Code	Reason
200	OK
404	Not found
500	Internal error

Parameters

Parameter	Description	Example
streamName	Name of the republished stream	streamName
rtmpUrl	URL of the server the stream is republished to	rtmp://localhost:1935/live
rtmpFlashVersion	RTMP subscriber Flash version	LNX 76.219.189.0
options	Transponder options	{"action": "mute"}
mediaSessionId	Unique identifier of the transponder	eume87rjk3df1i9u14elffga6t
width	Image width	320
height	Image height	240
bitrate	Video bitrate, kbps	500
keyFrameInterval	Video keyframe interval	60
fps	Video framerate	30
muted	Is sound muted	true
soundEnabled	Is sound enabled	true
soundFile	Sound file	test.wav
loop	Loop playback	false
rtmpTransponderFullUrl	Take stream name to publish to RTMP server from RTMP URL	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

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. Since build 5.2.1043 bitrate parameter is added which also enables stream transcoding while republishing.

Therefore, stream will be transcoded while republishing with any of the following parameters:

{
    "streamName": "name",
    "rtmpUrl": "rtmp://localhost:1935/live",
    "height": 240,
    "keyFrameInterval": 60,
    "fps": 30,
    "bitrate": 500
}

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.

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

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 (Facebook, YouTube), 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.

RTMP metadata sending may be enabled by the following parameter

rtmp_transponder_send_metadata=true

Since build 5.2.1935 RTMP metadata are sent by default.

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 parameter rtmpUrl 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

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.

Warning

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

Since build 5.2.1110 it is possible to set authentication parameters

rtmp_push_auto_start_url=rtmp://user:password@rtmp.server.com:1935/live

or

rtmp_push_auto_start_url=rtmp://rtmp.server.com:1935/live?username=user&password=pwd

Parameters will be passed in RTMP connect command.

Known limits

Only one RTMP URL can be used for automatic republishing.

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

Parameter	Default value	Description
rtmp_out_buffer_start_size	300	Stream buffer start size, ms
rtmp_out_buffer_initial_size	2000	Stream buffer initial size, ms
rtmp_out_buffer_polling_time	50	Buffer polling timeout, ms
rtmp_out_buffer_max_bufferings_allowed	-1	Maximum 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.
   SESSION_STATUS.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
   STREAM_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:8081/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
    STREAM_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

Server 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 streams with big frame size, data packets to send may not fit to socket buffer

Symptoms

Artifacts occur while playing republished RTMP stream via good channel

Solution

Enable RTMP packets buffering with the parameter

rtmp.server_buffer_enabled=true