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I am able to view the mediaSessionId with "find_all". But when I search with "find" I can't find the information find_all Api URL: https://xxxxxxxxxxxxxxxxxxxx/rest-api/stream/find_all Return Info [ { "appKey": "defaultApp", "sessionId"...

I want to record a stream I received from the ip camera but I need to use mediaSessionId for recording. I can't start recording with rest api because I can't access mediaSessionId rest-api/rtsp/find_all { "uri": "rtsp://root:Sxxxxxxx@xxxxxxxx:555/live2.sdp"...

Problem solved . I updated the settings as you said. CPU usage dropped a lot and ram usage increased thanks

I deleted the initial -Xmx32g statement. java -XX:+UnlockExperimentalVMOptions -XX:+UseZGC -Xms64g -Xmx64g -Djava.net.preferIPv4Stack=true -Dcom.sun.management.jmxremote=true -Dcom.sun.management.jmxremote.local.only=false -Dcom.sun.management.jmxremote.ssl=false...

Hi post update code is below java -Xmx32g -XX:+UnlockExperimentalVMOptions -XX:+UseZGC -Xms64g -Xmx64g -Djava.net.preferIPv4Stack=true -Dcom.sun.management.jmxremote=true -Dcom.sun.management.jmxremote.local.only=false -Dcom.sun.management.jmxremote.ssl=false...

Hİ I have a 32 core server with an average of 400 people connecting to total bitrate 350kbit Although there was no problem on the stream side, the web service crashed and the embet player did not open. It worked when I restarted the webcallserver again. I see the following error numbers on...

After starting the broadcast via api, when nobody is watching, it stops after a while and it is necessary to start it again with the startup service. Even if nobody is watching the broadcast, I want it to continue to be shot over rtsp. How can I do that ?

this works great for me thank you. :)

I use the rest api system on flashphoner. After creating a new stream, I want to get the stream based instant viewer count via rest api. When I use these "/connection/find" and "/connection/find_all", they return me a lot of data. Thanks for your help

It works when I add it disable_rest_requests=true Well when I add this, am I unable to use the rest api service?

hi Max I use rest APIs i changed the port numbers again it is not working and gives the same error http.port=9081 https.port=9444 I'm getting this error maybe because I'm testing the trial version because although I changed port 8081 to 9081, I see that it is still processing 8081 port...

I changed the port so it doesn't conflict with other apps but it doesn't work on demo license. Flashphoner conf : # Config flashphoner.properties # To get more settings: # ssh -p 2001 admin@localhost # default password: admin # show node-settings # show node-settings | grep port #server...

I would like to provide rtmp re-publish function to my customer. When I send the video image as h264, CPU usage is working low and well, but when I send also the audio besides the image, then the CPU increases. I think webrtc acc kodec cannot be used. How can I send the file as acc? I don’t want...

I cannot upload log files over the address you provided. Cообщение <1594740686.5f0dcfce29dc3@flashphoner.com> для support@flashphoner.com не было доставлено через ApMailer\Smtp. Ошибка: 104: Не удалось считать данные из сокета. Ошибка: Undefined index: cert_private_key

application sharing is definitely not re-publishing. I made all the updates. re-publishing works for chrome tab but screen sharing does not work. I use nginx as an RTMP server and It works fine when I post from OBS program Crome TAB sharing is sending this to the server: [meta] => Array...

Can I fix the quality in screen sharing? var constraints = { audio: false, video: { width: parseInt(400), height: parseInt(280), minBitrate : parseInt(300), maxBitrate ...

I can't get the auto-generated video publish object so i can't transfer images into canvas mockVideoElementForID = document.getElementById("localVideoShare"); mockVideoElement_get = mockVideoElementForID.getElementsByTagName("video")[0].id; console.log("ID : ")...

Hi I observed some situations on the system Webcam Flashphoner : session.createStream({ name: googleID+"_teacher", display: localVideo, cacheLocalResources: true, receiveVideo: false, receiveAudio: false, rtmpUrl: googleServer...

I using rtmp-re-publish for screen share but i am losing data. FFMPEG OUT : Input #0, flv, from 'rtmp://******.com/live/rtmp_rtmp_333_share': Duration: N/A, start: 325.942000, bitrate: N/A Stream #0:0: Data: none Stream #0:1: Video: h264 (Constrained Baseline), yuv420p(progressive)...

I want to share screen with re-publish (rtmp-push) feature but only h264 codec can be used because the v8 cod requires extra core usage. How do I ensure that only h264 is used in webrtc inputs?