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Поправил настройку, влияющую на NOT_ENOUGH_BANDWIDTH webrtc_cc2_bitrate_overuse_event_threshold = 0.05 Начиная с билда 2146, мы отключили это поведение по-умолчанию, с помощью webrtc_cc2_сс = false Так как на тестах с хромом увидели ложные срабатывания. Вызвано потерями более 5% на канале...

В последних версиях такие настройки (даны значения по-умолчанию) 1. Включить вторую версиюуправления адаптивным битрейтом. webrtc_cc2 = true 2. Включить зависимость битрейта паблишера от зрителей. webrtc_cc2_сс = false 3. Инициировать событие NOT_ENOUGH_BANDWIDTH если превышен порог потерь в...

Please read the troubleshooting guide Make sure your WCS server is up and running WCS processes are running WCS logs flashphoner.log and flashphoner_manager.log are active.

flashphoner.xml is old config and no longer used. Please use our latest examples from Web SDK For example you can download latest build from here. You can test from dashboard of your WCS server: https://wcs5-eu.flashphoner.com/demo2/phone HTML / js code is located here...

Your servers: WCS server : 192.168.30.161 Asterisk PBX: 192.168.30.159 So you have to set ip=192.168.30.161 ip_local=192.168.30.161 In your flashphoner.properties config. Please do service webcallserver restart to apply changes Dynamic modification is not supported. You can use something like...

It seems your WCS configuration is incorrect. Please share your /usr/local/FlashphonerWebCallServer/conf/flashphoner.properties config

We have tested this with your server. It works fine. We tested on default samples: 1. SIP as RTMP /demo2/sip-as-rtmp 2. Flash Streaming /demo2/flash-streaming As you can see, WCS adds prefix rtmp_ for stream name stream123. And you have to play stream rtmp_stream123. You can disable this...

As I assume, you have three machines 1) WCS installed. 2) Asterisk installed. 3) Your PC. Please create three pcap files: 1) wcs.pcap 2) asterisk.pcap 3) yourpc.pcap On Linux or Mac: tcpdump udp -s 4096 -i any -w log.pcap On Windows or Mac Run Wireshark and capture / filter all traffic between...

Yes it looks like be a bug with new version of server. We will check.

Because SIP call is linked with RTMP stream. If RTMP stream is failed, associated SIP call will be terminated immediately. If your SIP call is alive after RTMP fail, it looks like a bug because SIP call should be terminated. However in logs I see that call is terminated with BYE request.

Hello Your configuration looks correct. According your logs, your RTMP server does not receive RTMP stream. We have checked your RTMP server with FMLE. As you can see, connection was failed. Please check your RTMP endpoint and make sure it is working.

Нет, это не будет работать. В параметр display нужно передать ссылку на div-элемент. Таб туда передать нельзя. var el = document.getElementById('mydiv'); session.createStream({name: "name", display: el}).publish(); Так будет.

Т.е. просто открываем браузер, не заходим ни на какие страницы и по клику на иконке отправляется стрим? Мы такие вещи не тестировали. Поэтому нельзя сказать будет это работать или нет. Тестировали стандартный скриншаринг, когда на странице расположена кнопка и JavaScript код, который...

Hello This is correct. Sorry, it is bit unclear. When you do session.createStream().publish () you don't open any additional pages. 1. First user presses 'publish'. His / her stream is published to server. 2. Second user presses 'play' on another page (Player). Could you clarify your...

PhoneJS is quite old and out of date. Please test dashboard Phone example. Not PhoneJS. Test default WCS phone sample with a softhphone through Asterisk If server was installed properly, you will be able to test default demo samples in the dashboard. It will look like this...

В последних сборках мы отладили работу с адаптивным битрейтом. webrtc_cc2=true Эта настройка включена по-умолчанию, начиная со сборки 2158. Нужно убедиться что в flashphoner.properties что она не переопределяется в false. Обрывы стримов могли происходить из-за того, что битрейт WebRTC потока...

This works out of the box. If user connected to WCS is in active call state, any incoming call to the same user will be rejected by WCS with 486 Busy Here SIP response. WCS works as a softphone device and has the same behavior. It just receives a SIP call or places a SIP call. You describe Call...

XMPP is not supported. Could you please describe this behavior in details. I'm not sure that I got it. WCS has JavaScript API (Web SDK) REST API REST Methods You can find all docs here.

WCS supports H.264 and VP8 video codecs. XMPP is not supported WCS works as a browser-sip softphone. All SIP events are translated to JavaScript layer. If you ask about JavaScript API, you can learn how our samples work. For example Phone sample.

Please attach crash log errorPID.log here PID is ID of crashed process