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 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.
Chrome | Firefox | Safari 11 | Edge | |
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Windows | + | + | + | |
Mac OS | + | + | + | |
Android | + | + | ||
iOS | - | - | + |
Supported. Specify the name and password in the URL of the server, for example rtmp://name:password@server:1935/live
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:
Where:
REST-method | Example of REST query body | Example of response | Response statuses | Description | ||
---|---|---|---|---|---|---|
/push/startup |
|
| 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 |
|
| 404 - Transponder not found 500 - Internal error | Find transponders by a filter | ||
/push/find_all |
|
| 404 - Not found any transponder 500 - Internal error | Find all transponders | ||
/push/terminate |
|
| 404 - Not found transponder 500 - Internal error | Terminate operation of the transponder | ||
/push/mute |
| void | 404 - Not found transponder 500 - Internal error | Turn off audio | ||
/push/unmute |
| void | 404 - Not found transponder 500 - Internal error | Turn on audio | ||
/push/sound_on |
| 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 |
| void | 404 - Not found transponder 500 - Internal error | Stop inserting audio from the file |
Parameter name | Description | Example |
---|---|---|
streamName | Name of the republished stream | streamName |
rtmpUrl | URL of the server the stream is republished to | |
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 |
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 |
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} |
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.
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.
To send the REST query to the WCS server, use a REST-client.
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.
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.
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
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
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.
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.
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.
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, мс |
rtmp_out_buffer_initial_size | 2000 | Stream buffer initial size, мс |
rtmp_out_buffer_polling_time | 50 | Buffer polling timeout, мс |
rtmp_out_buffer_max_bufferings_allowed | -1 | Maximum stream bufferings allowed, unlimited by default |
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:
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(); |
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 |