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 | |
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} |
To send the REST query to the WCS server, use a REST-client.
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.
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.
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 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
WCS server can automatically republish all the streams published to a specified RTMP server. To activate this feature, set the next 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
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(); |