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https://appr.tc/ тоже не запрашивает доступ к камере и микрофону? По-видимому, c Safari может быть проблема с доступом к устройствам (здесь аналогичный вопрос).

Hello, WCS build with fix for WCS-2780 - 5.2.685. The problem with SDP parsing was due to invalid value of profile-level-id (profile-level-id=0).

Good day. If you plan to just republish incoming WebRTC streams as RTMP to third party servers, without any subscribers, you should consider at least two servers for publishing. It should be at least 64 CPU cores and at lease 64 Gb RAM. You can also place server instances behind a load balancer...

Good day. Please make sure that main (external) VM IP is set to ip and ip_local. For example, we deployed test VM based on our DO marketplace image: ipv4: 138.*.*.* Floating IP: 104.*.*.* and publishing works perfectly without any errors in server log with default settings ip...

Good day. Yes Flashphoner SDK provides WebRTC streaming tools only. Even for SIP calls, it is WebRTC under the hood, SIP leg works on server side. So you can try to integrate TwoWayStreaming example with Call Kit as Skype developers do with their own code.

Please make sure you have provided stream name or mediaSessionId as query parameter, for example curl -H "Content-Type: application/json" -X POST http://wcs:8081/rest-api/stream/metrics -d '{"name":"test"}' Please also add the following settings mixer_display_stream_name=true...

On demo.flashphoner.com, the default settings are used: rtmp_transponder_stream_name_prefix =rtmp_ rtmp_transponder_full_url=false When you enable rtmp_transponder_full_url=true, the stream name is taken from publishing URL as is. That's the difference. If you're streaming audio only (that's...

By default, republished RTMP stream is named rtmp_<source stream name>. The prefix ("rtmp_") can be changed using setting rtmp_transponder_stream_name_prefix in flashphoner.properties.

If it means that the WebRTC stream is republished as RTMP, then there are two different streams with different names. So, which stream will be played would depend on the name specified for playback.

Hello, 1. Use mediaSessionId of a published (not played) session as parameter for startRecording and stopRecording. 2. If rotation is set in seconds, the value will be parsed as Long. 3. Remove setting encode_record_name=true,BASE64 from the config.

Good day. Would you try just stop stream and start to play it in another div? ... stream1.stop(); stream2.stop(); ... stream1 = session.createStream({ name: streamName1, display: div2 }).on(STREAM_STATUS.PENDING, function (stream) {...

Good day. Yes, if you publish RTMP stream and play it as WebRTC, it should add some delay. Yes, it should work.

Yes. These ports are used to pass media traffic, WebRTC publishing and playback will not work if the ports are unavailable for incoming connections from client.

If the Origin A is not responding when some CDN node is starting, this node could not connect to CDN and become invisible. When, however, Origin A responds again, this node connects to it and become visible. The only recovery way for entry point is to start it as soon as possible. However, the...

We tried to connect to RTSP URL provided, but all connection attempts are failed with exception 08:27:07,913 WARN annelUpstreamHandler - pool-45-thread-1 EXCEPTION, please implement com.flashphoner.rtsp.RtspResponseHandler.exceptionCaught() for proper handling. java.io.IOException: Connection...

Good day. The cdn_point_of_entry defines a server to which the node must connect for the first time to enter CDN. When some CDN node is started, it should receive current CDN nodes list to establish signaling connections with all of them. So, the first signaling connection will be established...

We just fixed the issue with Sectigo CA on our web server. Please check.

Good day. Yes, this is correct. When you're using TURN server to connect to WCS, at WCS point the TURN server is the ICE candidate, not a browser. You can check it by collecting a traffic dump on WCS side then parsing STUN traffic in Wireshark.

Good day. Please check the publisher channel quality: - is channel enough for publishers resolution/bitrate? - are there packet losses while publishing a stream? Publishers channel quality can be checked as decribed here. If channel is not enough, consider to use lower resolution/bitrate. Packet...

Yes. Please check Two Way Streaming example (demo). You should just combine this example with 2 players mentioned above. Some tips: 1. Websocket session can be one for all streams 2. For every stream, there should be a <div> element on the page.