Search results

Good day. According your logs, you pull RTMP streams from some sources and play them via WebRTC. The errors that you marked probably mean a losses on channel between RTMP source and server. For more accurate diagnosis please collect a report as described here, including client debug logs, server...

Good day. You can try to solve the problem on SIP side: your SIP provider should allow SIP calls from your WCS servers IP addresses only. So if clients even can see SIP call details, they cannot make a call bypassing your servers. For WCS2 it is the only possible solution.

If you can not provide us access to RTSP stream, please provide SSH access to your server.

Good day. Unfortunately, the report contains no client debug logs, and RTSP session can not be extracted from traffic dump. Please provide us access to your RTSP stream to reproduce a problem on our test servers, and SSH access to your server for our engineers to check. The credentials should be...

With this setting, stream will stop after 60 seconds (this is 60,000 milliseconds). Change the transcoder_agent_activity_timer_timeout setting, specify the value you need, set this parameter in flashphoner.properties file. In the example there was a small typo, now it is fixed, try to...

You have to sign in as admin admin to be able to import certs.

Good day. Please collect a report as described here and send to support@flashphoner.com, we will check. You need to check the video with from different parameters (the Media Devices example can be used for the tests): https://{your server domain name or IP...

Good day. In step 10 ("Linked Frameworks and Libraries") you have been connected the "VideoToolbox.framework"? In additional, for more information provide full console output of the build script and attach for investigation screenshots step-by-step.

You need change the following parameter in flashphoner.properties file, that was listed in the previous post : transcoder_agent_activity_timer_timeout=60000 where 60000 - timeout in milliseconds the trascoder stops by inactivity (if no any viewers on your stream). The trial license may add...

Stream transcoding requires a lot of CPU resources. So if all you need is to rename pulled RTMP stream, you can do the following: 1. Set the following parameter in flashphoner.properties file rtmp_transponder_full_url=true and restart WCS 2. Pull stream /rest-api/pull/rtmp/pull { "uri"...

All the WCS main settings should be set in /usr/local/FlashphonerWebCallServer/conf/flashphoner.properties file.

Good day. We have checked report you sent by email and answer your questions: 1. "Overconstrained error" is Webkit issue. Safari can read min and max properties, but does not allow to stream in some resolution. Please check this test https://webrtchacks.github.io/WebRTC-Camera-Resolution/ If...

What exactly did not work? please provide more details. Unfortunately we didn't receive your report, it is not found even in spam. if you sent a large file (20 MB and more), please share this file as download link.

Good day. The default policy for calls is LOG when backend returns 403. So, we have to change the policy: 1. Put the file rest_client_config.json near the backend script. The file should contain at least the following: { "call" : { "clientExclude" : "", "restExclude" : ""...

Good day. You can use REST hook to authenticate user. Here is described detailed example to authorize users by domain, use this example as basis. If there is no any subscriber (viewer) for transcoder output stream, the trascoder stops by inactivity after 60 seonds by default. You can set this...

You should set your own server IP-address or domain name here: https://{your server domain name or IP address}:8444/client2/examples/demo/streaming/media_devices_manager/media_device_manager.html because this is Media Devices example URL on your server.

Use - rest-api/pull/rtmp/pull to pull the RTMP stream - rest-api/transcoder/startup { "uri": "rtmp://fms.105.net/live/rmc1", "localStreamName" :"5e0c" } to create a new stream with a local name rest-api/transcoder/startup can be used to create a transcoded version of a stream, but if the...

Try different resolution, bitrate and fps values both with UDP and TCP transport to see which would give better result. E.g., if bandwidth allows - increase bitrate and fps, and if bandwidth is low - decrease bitrate, fps and resolution. The Media Devices example can be used for the tests...

Hello, To establish Secure WebSocket connection, SSL certificates have to be imported. To use unsecure connection, use this URL: http://WCS:8081/admin/demo.html Also, the RTMP stream you are using for testing contains B-frames. There may be lags during playback.

Good day. If your WCS server and WebRTC stream source are in same datacenter, the stream quality is probably not a publishing but playback problem. So try to use WebRTC over TCP for playback, this helps to reduce channel lossess for viewers, but it may increase latency.