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According these stats, you have 55 websocket connections in total 2 incoming webrtc streams (875960cc, 875960ccAudio) 47 outbound webrtc streams Stream 875960cc has 46 subscribers Stream 875960ccAudio has 46 subscribers too So all seems correct. No. You have 55 connections. 2 publishing...

Make sure that following UDP ports are open [30000-32000] on your server. You can temporary stop firewall to avoid any firewall issues.

What is your IP address? Are you in the same LAN with the server 172.31.1.100 ?

Yes. This is correct. When you do var session = Flashphoner.createSession({...}); Websocket connection is established. When you do session.createStream({...}).publish(); WebRTC stream is established. So you can have, i.e. 100 websocket connections and 0 streams.

These settings should be correct ip=172.31.1.100 ip_local=172.31.1.100 Please remove duplicated 'codecs' setting. First setting is correct: codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv Restart WCS server to apply changes. service webcallserver restart

Hello Most likely you have configuration error in WCS_HOME/conf/flashphoner.properties Example: ip_local=192.168.1.5 Here 192.168.1.5 is your LAN IP address. Please share 1) Your flashphoner.properties config 2) Your ifconfig output. ifconfig

Проблемы конкретно с этой камерой (камера RVI). Другие video-only rtsp-стримы отыгрываются нормально. Результаты тестирования показывают, что стрим идет с кодеком H.264H (High profile) Stream #0:0, 28, 1/90000: Video: h264 (High), yuv420p, 352x288, 1/180000, 6 tbr, 90k tbn, 180k tbc Этот профиль...

WCS works with browsers via WebRTC and uses ICE and DTLS. WCS works with Asterisk as plain SIP / RTP and does not use ICE and DTLS or WebRTC in communication with Asterisk.

To get this working with WCS you have to use Asterisk peers without DTLS, like this: [1061] ; This will be the legacy SIP client type=friend username=1061 host=dynamic secret=password context=default disallow=all allow=alaw qualify=no ;rtp.conf ICE and DTLS should be disabled for the Asterisk...

I would list four basic steps: Test your Asterisk First of all you have to test your Asterisk server without WCS. Just make a test between two softphones like bria, xlite or linphone through Asterisk. Let's say you have configured accounts (peers) 101 and 102 on Asterisk server. You should be...

Hello Web Call Server communicates with Asterisk as a legacy SIP peer and operates with browser as WebRTC peer. Browser <-- WebRTC --> WCS <-- legacy SIP/RTP --> Asterisk PBX So you have to configure peers for WCS as legacy SIP peers on your Asterisk server. Please create a new topic in the Web...

Если пользователь неожиданно теряет коннект к серверу, то включаются keep alive. Если поднимать их значение, например 10 проб по 6 секунд, то сервер будет держать пользователя 60 секунд в памяти. Если уменьшить их значение, например 2 пробы по 2 секунды, то сервер будет держать пользователя 4...

Проверили. Проблема, похоже именно с этой камерой. Т.е. WCS скорее всего неправильно депакетизирует H.264 поток, который приходит с камеры. Пожалуйста, не отключайте камеру и не меняйте настройки, чтобы мы могли ее протестировать и сделать фикс.

Добрый день SSL сертификат обязательно использовать только в том случае, если вы используете камеру или микрофон в браузере Google Chrome. Если у вас работает только на просмотр, то сертификат не нужен совсем и можно работать полностью через http / ws, а не через https / wss. Тогда...

Hello We don't have ready-to-use solution for pure flash-sip client. Though we can implement flash client as a custom request. Please contact sales@flashphoner.com for more details.

There is no package for apt-get. Installation: 1. Download and unzip tar.gz. 2. ./install.sh https://flashphoner.com/download

Добрый день. Общая техподдержка WCS версии 3 на текущий момент завершена. Если у вас есть лицензия и активный план техподдержки, пожалуйста обратитесь по почте helpdesk@flashphoner.com для решения этих вопросов и персональной техподдержки.

Hello All examples are located in the folder: /usr/local/FlashphonerWebCallServer/client2 So you can put your code into this folder. However, by default this folder is invisible. You have to enable demo user to see this folder over http / https Edit config...

Hello In latest versions we have disabled bandwidth detection by default because it was generated ton of error logs on some systems. So it is ok because it is currently disabled. global_bandwidth_in=0.000 global_bandwidth_out=0.000 Regarding If you really do not have such number of connections...

Does this help?