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What is your stream source? Could you provide more information.

Добрый день. Для этого необходимо отключить транскодинг, который Вы включаете, указывая явно разрешение при воспроизведении. Если играть поток с тем же разрешением, с которым он опубликован, транскодинга не будет Вы можете контролировать качество канала при публикации. Обратите внимание. что...

This is probable log files. You can stop WCS systemctl stop webcallserver then clean logs folder and recording folder if there is one rm -rf /usr/local/FlashphonerWebCallServer/logs/* rm -rf /usr/local/FlashphonerWebCallServer/records/*

We also cannot access the server by SSH credentials you've sent Seems like something broken on router/NAT, please check.

We've tested RTMP republishing using your flow (capturing RTSP, then republish as RTMP) with your server. There is the following message in server log when trying to publish RTMP stream to Youtube: 03:00:30,253 ERROR RtmpPublisher - Thread-144 error creating client connection: Connection...

Unfortunately, SIP traffic cannot be identified in pcap dump. Perhaps you've started dump collection after call establishing, but it must be started before this. On your server we saw the following messages in server logs: 13:13:08,673 WARN AbstractStunSocket - STUN-UDP-pool-40-thread-1 Can...

You can detect browser and device by checking navigator.userAgent, please see example here. Then, define tha constraints depending on device, for example function publish() { ... var constraints = getConstraints(); publishStream = session.createStream({ name: streamName...

Your screenshot is missing two details 1. No /rest-api/rtsp/startup 2. No rtmpTransponderFullUrl parameter for /rest-api/push/startup request

This works on your server as described here step by step.

If instance has been stopped, IP address will be changed on next launch. Know you public IP address and correct config /usr/local/FlashphonerWebCallServer/conf/flashphoner.properties Then restart WCS server. service webcallserver restart Check private IP address too and correct if needed.

Please test the call again and gather the traffic dump tcpdump -i eno8 udp -s 4096 -w log.pcap If you open this dump in Wireshark you should be able to see voice traffic. Send us this dump for analyzing either via form or directly into the forum thread. Note. The test call should be started...

It seems port 8081 is not reachable on your server. Please check if port is open and not bound by other applications. As an option you can try https port 8444 https://cibunet.com:8444/rest-api/push/startup { "streamName": "rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov"...

Hello Server 192.168.10.12 must be located in DMZ zone. At least TCP UDP ports in range [30000-32000] for SIP, voice, video traffic must be reachable. For example, your host has public static IP address 9.9.9.9 In such a case if UDP or TCP packet received on 9.9.9.9 port i.e. 31500, then host...

Ah yes https port is 8444 curl -X POST "https://127.0.0.1:8444/rest-api/stream/find" -H "Content-Type:application/json" -d '{"name":"100000000000003","published":"true","display":"metrics"}' --insecure http port is 8081 curl -X POST "http://127.0.0.1:8081/rest-api/stream/find" -H...

Try to pass valid json curl -X POST "https://127.0.0.1:8081/rest-api/stream/find" -H "Content-Type:application/json" -d '{"name":"100000000000003","published":"true","display":"metrics"}' --insecure

1. Restart WCS server to stop all push sessions 2. Open RTSP stream manually to see in the player. Example: https://demo.flashphoner.com/client2/examples/demo/streaming/player/player.html 3. Try to push this stream to localhost: curl -H "Content-Type: application/json" -X POST...

It is normal. MCU stream name without participants audio is formed as {mixerName}-{incomingStreamName} No. Two streams with the same name cannot be published on server.

This means your SSL certificate is expitred. Please buy a new certificate for your domain and import it to WCS as described here. You can also use HTTP port 8081 for REST API queries, in this case SSL certificate is not required.

WebRTC may publish lower resolution than set in constraints depending on channel quality. So video contraints strictly define aspect ratio only. You can check actual publishing resolution in stream metrics: POST /rest-api/stream/find HTTP/1.1 Host: localhost:8081 Content-Type: application/json...

We checked your server again. You've not set rtmp_flash_ver_subscriber as we recommended above rtmp_flash_ver_subscriber=LNX 76.219.189.0 Now, Youtube returns Connection refused when you're trying to republish strem to it. So, set the parameter as recommended. To republish RTSP stream to...