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Путь в конфигурации запуска должен выглядеть как ...wcs_android_sdk_samples/export/output/build.gradle Попробуйте установить по этой инструкции.

If the call itself is successfully established, this seems like SIP codecs negotiation issue. SDP may be cut during negotiation when using SIP UDP signaling because it does not fit to one standard network packet (1400 bytes). Please set SIP signaging to use TCP if your PBX supports this...

Please update to the latest NPM package build 2.0.203 @flashphoner/websdk If this does not help, go to English section of the forum and create a topic about it, our engineers will answer.

На первом скриншоте (screen.png) артефакты транскодирования, характерные для потерь на канале паблишера. В таких случаях рекомендуем бороться с потерями: использование TCP, снижение максимального битрейта публикации, смена канала. На втором скриншоте (screen2.png) похоже на сброс битрейта, но и...

Ориентировочно через месяц.

Уточните, пожалуйста, какого вида артефакты наблюдаются? Если макроблоки и акварель, то это выглядит как сброс битрейта. В этом случае, если проблема наблюдается у какого-то конкретного клиента, можно рекомендовать снизить границы битрейта, например 300-400 кбит/с, либо попробовать не...

Добрый день. К сожалению, на предоставленном скриншоте трудно разглядеть артефакты. Если артефактов нет при проигрывании исходного потока, но есть при проигрывании потока в микшере, это может означать проблемы с самим потоком (скачки битрейта), которые браузер компенсирует за счет jitter...

Good day. Please do not dub topics. Our engineers will answer to all the questions in working hours (Mon - Fri 10:00-20:00 GMT +3).

Good day. Yes, WCS REST API supports request body in JSON form only. Please read details here You should also use header Content-Type: application/json This is the working example of /push/startup REST API query using curl curl -H "Content-Type: application/json" -X POST...

Добрый день. Если речь идет о публикующем клиенте, достаточно использовать функцию Stream.muteVideo(). Если же речь о зрителе, то для отключения видео нужно переподписаться на поток с констрейнтами constraints { video: false, audio: true } Вы также можете попробовать новые функции SFU...

If rtmpUrl is specified when the stream is created, and the push fails (e.g., if the RTMP server is unavailable), then the WebRTC publishing also fails. Please verify that the server rtmpUrl and stream key are valid, and streaming not from the WCS is working. With the rtmpUrl from the example...

Good day. Yes. But demo server uses default parameters for all examples to work: - stream published name is always used as a part of stream key - stream key is always prefixed by rtmp_ to allow republishing to localhost So the example above publishes a stream as...

Good day. Screen Sharing example uses a much of dependencies which should be placed by relative paths. This is the reason to deploy Web SDK bundle on webserver and open via HTTPS (or HTTP only if the server is on the same host and can be opened by localhost address) You can also try a minimal...

Receiving incoming call using push notifications relates to the function described in this post. In ticket WCS-3361 we will check if this still working, and fix if not, for Web SDK. Then, we raise corresponding tickets for mobuile SDKs if needed. Please reproduce this and collect a report on...

In Web SDK, it is possible to re-connect to existion SIP session using a special auth token: 1. Open a page, establish a WCS session, register on SIP server, remember a token 2. Close the page. SIP session should exist during 1 hour by default 3. Open a page, connect with token. 4. Receive an...

Only webrtc connection playing that stream will be closed if no video traffic. But RTSP connection will persist. So the only way to check if video is stopped is stream metrics REST API request: POST /rest-api/stream/find HTTP/1.1 Host: localhost:8081 Content-Length: 57 Content-Type...

Good day. The media ports (31001-32000/udp by default) should be available to connect from publisher to server. Please check your server settings media_port_from and media_port_to, then check ports availability from this range using this method. The ports may be blocked on server side (firewall...

When app is closed, websocket session is closed too. So, WCS closes SIP connection with PBX. May be you should consider SIP call to mobile phone via PSTN gateway (see details here)? Please try to test a call between iOS Phone application (you should build it from sources) or Android Phone Min...

No, this issue is not related to server setup. We mentioned it for your information only. Perhaps you should check those cameras. You should enable video RTP activity checking if disabled rtp_activity_video=true In this case, connection to RTSP camera should be closed by RTP activity.

Good day. SIP connection is always establishing between WCS and SIP PBX. Only websocket signaling messages and WebRTC traffic goes between client and WCS. Please read details here. So you should keep SIP connection between WCS and SIP PBX and, therefore, Websocket session between client and WCS...