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Good day. We fixed the problem with recording stream from MS Edge when VP8 priority is higher then H264. Please update to build 5.2.68 and check.

Good day Please update WCS to latest version from this page and check if stream is displayed faster. If not, please reduce stream resolution and bitrate for stream playback. Also please clarify what do you use in your application: iOS SDK (for native application) or Web SDK (for React application)?

Good day. We implemented dynamic RTSP interleaved channels setting in build 5.2.67. Now you can remove rtsp_interleaved_channels parameter from config, and your stream will be played as so as other RTSP streams. Please update and check.

Good day. We have just checked your server, and we see license is activated and streaming is working. Browser can return Empty Response Error when you try to access HTTP port via HTTPS or vice versa. But now your site looks accessible via HTTPS with valid certificates. License key cannot be...

Good day. We fixed iOS Safari 12.1 issue, streaming is working in WebSDK build 0.5.28.2753.79. Please update WebSDK to this build.

Добрый день. Основные изменения перечислены здесь: https://docs.flashphoner.com/pages/viewpage.action?pageId=9241025 Немного поменялась архитектура. Убрали тяжеловесный модуль админки WCS manager, сервер стал быстрее стартовать. В остальном, все фиксы и улучшения из 5.1 были портированы в 5.2...

Hello Make sure UDP ports reachable https://docs.flashphoner.com/display/WCS52EN/Accessory+tools#Accessorytools-Portroutingchecking As an option you can try to switch to TCP and check ports using telnet...

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...