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Good day. Please clarify your case: 1. You're trying to play some file as VOD (vod://file.mp4) 2. If there's no such file in WCS_HOME/media, no error returns Also please clarify what WCS version and WebSDK version are used, and what browser. Please try to reproduce the issue in Player example...

Hello As you can see you have a lot of connections to websocket port (8443 by default) So you have to protect TCP connections to this port using - netstat - iptables - tcpdump Using netstat you can find established connections netstat -nlp | grep java Using iptables you can reject unwanted IPs...

Please clarify: did you try using the Android SDK Media Devices example? What did not work, and on what device?

Hello, To import Objective-C files (the framework files) into Swift code, Objective-C bridging header file has to be used. Path to the file should be specified in Build Settings > Swift Compiler - General > Objective-C Bridging Header (e.g., as path to Info.plist is specified). Framework header...

Посмотрите, пожалуйста, как получение метрик реализовано в примере Media Devices. Показывает packets, bytes received и bitrate. Нужно убедиться, что битрейт не ниже требуемого.

Эти скрипты нужны для работы WSPlayer в iOS-браузерах, которые не поддерживают WebRTC.

Those methods for muting audio and video are supposed to be used for a published stream - as in the Android SDK Media Devices example. For player, there is volume control in the example. Use - Android SDK 1.1 for Android 9 and higher - Android SDK 1.0 for Android 8 and lower

Добрый день, На вашем сервере версия WCS 5.2.56 - рекомендуем обновить на более новую и перепроверить. На demo-сервере тоже была не новейшая версия (позднее был фикс для воспроизведения в Edge) - обновили. Проверили в Edge и IE 11 c Two-way Streaming на вашем сервере - публикация и...

Hello, Please try it with this setting in WCS-HOME/conf/flashphoner.properties: periodic_fir_request=true Restart WCS to apply the setting. Also, try using RTP over RTSP (TCP) - interleaved mode in the player used for playing the RTSP stream. If that doesn't have effect, please clarify the...

Hello, The issue has been fixed in WCS version 5.2.265. rtsp_interleaved_mode setting is not required and can be set back to default (removed from the config).

Hello We do not have Cisco account. To proceed please 1. Request a quote for SIP enterprise license at sales@flashphoner.com from your corporate email. 2. Provide Cisco account and step-by-step instructions how to make a call using Cisco API. Then we will be able to arrange a demo server and...

Good day. So, the sync issue was fixed some builds ago. If another issue occur on latest build, please let us know.

Hello No progress for now. Anyone can fork our Web SDK and port to React Native if it is portable at all: https://github.com/flashphoner/flashphoner_client Another way is using raw Websocket API + WebRTC opensource lib https://docs.flashphoner.com/display/WCS52EN/Raw+WebSocket+API

Hello Generic endpoint should be enough. It should be configured exactly as production endpoint otherwise you may encounter issues during migration to production Cisco endpoint. For example if your production endpoint is configured with SIP/TCP and codecs H.264+Opus, the generic endpoint must...

Good day. Have you tested latest builds 5.2.259 or 5.2.263? If yes, and if problem still reproduced, we need full error report to check. Not separately sent logs and dumps, but one archive with logs and dumps collected simultaneously. If you're having any troubles to collect report, please...

Please try playing the "unstable video" (low fps) RTSP on WCS 5.2 with this setting in WCS-HOME/conf/flashphoner.properties: rtsp_interleaved_mode=false Restart WCS to apply the setting.

Please correct if I'm wrong. 1. Cisco places a call to WCS. 2. WCS does auto-reply using hosted mp4 file as audio+video source. 3. Cisco is gathering statistics from the received stream (streamed mp4 file). This is what works out of the box: 1. Receiving of incoming call by WCS and auto-reply...

Good day. We checked logs you've set. It seems like the issue was fixed in build 5.2.130. Please update to latest build 5.2.259. If problem still persists please collect a full report as described here, since build 5.2.241 there is a script for it. Also collect traffic dump for the problem stream.

Good day. Please update WebSDK to latest build from this page. Also please check WebRTC streaming known issues, point 8, this problem can occur in Safari on old iOS versions.

Unfortunately, there's not enough information to diagnose a problem in logs you have sent. Please collect a full report as described here including client debug logs (mandatory) and traffic dump if the problem is still reproducing.