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Good day. This is the most preferrable way if you can not provide direct access to RTSP stream.

And Case 3. Maybe you just need publish and play WebRTC stream? 1. Publish WebRTC https://demo.flashphoner.com/client2/examples/demo/streaming/two_way_streaming/two_way_streaming.html 2. Play WebRTC https://demo.flashphoner.com/client2/examples/demo/streaming/player/player.html This is the...

Case 2. Re-publishing to external server (your case). Are you sure you need RTMP re-publishing to external server? Because if Yes then you would need to install such an external server or CDN.

There are two different cases Case 1. WebRTC publish RTMP playback Here you don't need external server. How to test 1. Publish a stream. https://demo.flashphoner.com/client2/examples/demo/streaming/two_way_streaming/two_way_streaming.html 2. Play as RTMP in Flash player...

You can do that from CLI 1. Connect to ssh localhost ssh -p 2001 admin@localhost 2. List apps. show apps 3. Update FlashStreamingApp if you publish from OBS or live encoder update app -l "http://10.10.10.10/app" FlashStreamingApp or Update defaultApp if you publish via Wevsocket/WebRTC...

You should set up a backend server to handle /connect REST hook, username and passord will be passed to it. Please look at user authentication by domain example that can be used as basis for your custom backend setup.

Good day. Please collect a report including client debug logs as described here and send us using this link. Another option is to provide us SSH access to your server using this link, we will check stream recording in place.

Please check if target server for RTMP stream republishing is available, RTMP port ia not blocked on this server, and sofware to capture the stream (WCS, Wowza, Nimble etc) is already running

The record parameter affects stream publishing only, but not SIP calls. To record SIP call streams, you should rework backend: 1. Arrange conference on Asterisk 2. Clients makes SIP call to the conference using WCS Android SDK 3. Cilent side or backend initiates SIP as stream call from WCS to...

We just tried to reproduce the issue: 1. Published RTMP stream (because WebRTC publishing does not work due to closed media ports) to your server 2. Re-published stream as RTMP to our demo server 3. Played re-published stream on demo server So your server seems to be working. Please chaek...

Yes, in iOS stream is not available on client also.

Добрый день. Если Вы еще не покупали лицензию, нажмите кнопку Buy на этой странице и следуйте инструкциям на экране (язык можно переключить на русский). После того, как вы получите лицензионный ключ (это будет другой ключ, не триал, которым вы пользовались ранее), необходимо деактивировать на...

Good day. You can define custom video element and pass it using remoteVideo option while creating a stream to play. For example, to weak Two Way Streaming example: 1. Add video element to player div in two_way_streaming.html file ... <div class="col-sm-6"> <div class="text-center...

No, there's no any way to do this on client side. On server side, SIP call can be redirected to a stream, and this stream can be written to file. Another way is to republish SIP call as RTMP stream to another server and record this RTMP stream.

Good day. Screen sharing is supported in Android SDK 1.1 only due to libwebrtc implementatin limits.

Good day. We checked your server. Please update JDK 11 to JDK 12 or 14 (steps 1-4 from this page). Unfortunately, we cannot publish anything to your server because ports 8443 (wss) and 1935 (rtmp) seems to be rejected on router. Please allow streams publishing to the server or publish some...

This probably happens because you call finish() from main activity OnClick() rigth after service is starting. So, main activity finishes, and onDestroy works. Please take this example and replace stream publishing to voice call referring to Android Phone example.

Good day. Mixer looks excessive for the described case. You can implement the logic on frontend: 1. Client connects to the server before metting starts. 2. Frontend requests to play ads video. 3. When meeting begins, frontend stops playing ads video, publishes client local video and plays remote...

Good day. Unfortunally, voice call recording does not work on all Android devices in latest Android versions. Thus, recording usually works in 8 and 8.1, but is forbidden by Google since 9. Some vendors can bypass the restriction, but not all the vendors. That's why server side recording is the...

Good day. WCS WebSDK is intended to use in browser. React Native provides no DOM, so WebSDK needs to be reworked to integrate with React Native webrtc implementation. You can try to modify WebSDK source code to solve it. This seems like SSL certificate problem. Please import SSL certificate as...