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This format of RTMP URL - rtmp://wowza:1935/app?user&pass - is currently not supported. Created internal issue WCS-1917 for the implementation, will notify when there is an update.

In your second report everything seem working, but you've changed WCS configuration and played RTSP stream from localhost. Is there a problem with the same configuration to play RTSP stream from another PC in the same network (192.168.1.*) using Player example...

Good day. Please update to the latest build from this page and check if the issue still persists. If yes, please prepare a report as described here and send to support@flashphoner.com, we will check.

Было исправлено в версии 5.2.61.

Does the callee send audio? Please send traffic dump and log for the RTMP_STREAM_WAIT case. Try publishing to WCS (RTMP URL - rtmp://localhost:1935/live) - does the stream get published?

Please see this doc: republishing a SIP call to an RTMP stream As to using SIP URI for callee, the format is 'sip:username@domain'.

Try rtmp://username:password@rtmp.myserver.com:1935/application

Good day. In logs you've sent we see 2 tries to establish a SIP call: 1. Your first call was not established because you probably set wrong authentication parameters, here's SIP signaling sequence from server log: --------------------> INVITE sip:callee@domain SIP/2.0 <--------------------...

Good day. We work on this issue, it took a time to reproduce the problem steadily. Please be patient.

Good day. We work on it and let you know results. Please be patient.

Good day. In your report client debug logs is missing, and traffic dump contains no WebRTC connection and RTSP connection sequences. So please collect the report again. Also collect report when you successfully play a stream, to compare. Preliminarly, it seems like IP address misconfiguration...

Это уже другая проблема, в Chrome Canary выставили более жесткий запрет на кроссдоменные запросы. Если обратиться непосредственно к примеру с демо сервера https://demo.flashphoner.com:8888/client2/examples/demo/streaming/two_way_streaming/two_way_streaming.html, публикация должна работать. По...

Good day. There is a decription of DTMF sending methods supported by WCS, with Asterisk PBX configuration examples, see details here. Another PBX should be configured similarly. Also please update to the latest WCS build from this page and increase server RAM to allocate more Java heap memory...

Good day. We are now testing dynamic RTSP interleaved channels setting to play RTSP stream from your camera without breaking another RTSP streams playback. Could you please provide us access to your stream again for at least 24 hours?

Please collect a report as described here and send to support@flashphoner.com. We will check.

Please configure firewall on your test server as described here or disable it, then check again: - start tcpdump tcpdump udp port 31000 - send a packet from another PC echo-n "hello"| nc -4u -w1 192.168.1.230 31000 If ports are opened, tcpdump should receive some data and log it to console

Добрый день. На сборках Chrome Canary 75.0.3757.0, 75.0.3759.0 проблема не воспроизводится, публикация и воспроизведение работают нормально, в том числе на нашем демо-сервере. Сообщение "The AudioContext was not allowed to start" работе не мешает, аудиоконтекст при воспроизведении создается...

Good day. Please check if media ports are not blocked when you try to access your server from a different PC using tools described here. Configure firewall on your server to access a mininum set of ports as described here.

Good day. Android SDK 1.0 supports Android from 4.4, older versions are not supported due to API requirements. Android SDK 1.1 supports Android 9 and newer. If you need to run your application on outdated Android versions, please implement web application to run it from Chrome browser. Note that...

Hello. Please update to latest WCS build from this page, version 5.0 is outdated. To avoid missing keyframe please set the following parameter in WCS_HOME/conf/flashphoner.properties file periodic_fir_request=true You can get stream recording file name by following ways: 1. Using WebSDK...