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Android SDK две версии 1.0 и 1.1 1.1 для Android 8 и выше 1.0 для Android 7 и ниже

Каждый пользователь имеет websocket соединение с сервером с уникальным sessionId. sessionId передается в REST хуке "connect" при коннекте пользователя к WCS серверу. Каждый стрим имеет уникальный mediaSessionId. mediaSessionId передается в REST хуке playStream при попытке проиграть стрим с...

We tested again. The latest accounts you've provided are successfully registered on SIP server, but call cannot be established because WebRTC media ports (31001-32000/udp by default) of WCS server are not available from outside: Please check if the ports range is available from PC where you...

Добрый день. Приложение по умолчанию defaultApp работает для микшера 09:46:59,012 INFO RestClient - API-ASYNC-pool-13-thread-121 RECEIVED REST OBJECT <== URL:http://localhost:8081/apps/EchoApp/connect OBJECT: { "nodeId" : "62cjXxRa99MhToFv0RWUFWsoHwSE81W5@***.***.*.***", "appKey"...

We tested on latest WCS build 5.2.912 but did not found a viewable difference. Please try to increase opus encoder complexity from 10 by default to 20 for example opus.encoder.complexity=20 and check. If this does not help, please provide SSH access to your server and a test file (put it to...

Добрый день. В Web SDK можно отправить параметр token, который можно обработать на кастомном бэкенде connection = Flashphoner.roomApi.connect({urlServer: url, username: username, token: token}).on(SESSION_STATUS.FAILED, function(session){ ... }); Следует учесть, что при...

Добрый день. Мы создали тикет WCS-3095 и сообщим здесь о результатах

We just tried to register with both accounts you've provided. The SIP server breaks the connection after first SIP REGISTER packet sent to port 5061: but when we trying to register to the port 5060, the flow goes correctly (but server returns 403 Forbidden): Unfortunately, we cannot check...

Good day. There are a various ways to check server state and stream metrics on server side. On client side, you can check a channel quality and collect browser WebRTC statistics. To detect stream playback failure, you should use custom backend which should handle /StreamStatusEvent REST hook

If upgrade to latest build does not help, please provide SSH and publishing access to the server using this form, we will test.

Please clarify how much worse? Are there some clicks, pauses and so on in mixer output stream? Please check if you set opus.encoder.bitrate parameter as we recommended. Also, check server CPU load using top for example. If CPU load is constantly high, consider to upgrade the server to more CPU...

Поскольку мы оцениваем пропускную способность локальной сети, предположим, что среда передачи полностью прозрачна, и определим минимально необходимую пропускную способность адаптера. Как правило, физические сетевые адаптеры выпускаются со следующей максимальной пропускной способностью: 100...

Эти параметры не рекомендуется менять, они выбраны по умолчанию именно для случая стриминга 4K потоков и относятся к буферизации отправляемых и принимаемых пакетов на стороне сервера. Исходя из приведенных исходных данных, пропускная способность локальной сети должна быть не ниже 10 Гбит/с на...

With SIP credentials you've provided we receive 403 Forbidden response from SIP server, but REGITER flow looks normal. Please test a call. If this still cannot be established, please provide actual SIP credentials using this form, we will check.

There is Player example, please see its code on GitHub. You can build custom player based on the code. The stream should be published on WCS server before playing it, othewise client will receive an error.

First, all the files should be converted to MP4 with the same codec (AAC), channels count (stereo), sample rate (48 kHz to prevent excessive resampling) 1. Set the following parameter in WCS configuration file flashphponer.properties mixer_idle_timeout=10000 opus.encoder.bitrate=128000 2. When...

We finally checked your server. Seems like you've set a wrong IP address in ip and ip_local settings So WCS cannot bind SIP signaling port at all. When changing IP address settings, WCS binds SIP signaling port (30001 for first SIP registration by default) and sends SIP REGISTER to SIP server...

Hello! Streaming quality is often limited by network bandwidth rather than CPU and RAM. Network bandwidth at the rate of 1 stream with a bit rate of 1 Mbps takes about 2 Mbps of channel bandwidth. For high quality broadcasts without freezes, artifacts, etc. the channel should be loaded no...

Good day. We cannot reproduce the issue in latest build 5.2.912. Please check if the following parameter is set in flashphoner.properties file h264_strict_kframe_detect=true Pleas also check: - if keyframes are sent regularly from RTMP encoder - if WebRTC stream published from browser can be...

Good day. Unfortunately, no. You should concat the required files to the one using ffmpeg for example. Yes. You can play a VOD-live stream directly from browser as described here