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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...

We let you know about ticket in this topic. Please be patient.

In logs we see a possible problem with RTSP sources: seems all the RTSP published sessions have no video: - server log: 05:00:53,189 WARN RtpActivityTimerTask - FScheduling-pool-1-thread-5 RTP ACTIVITY EVENT DETECTED! - client logs: 05:00:53,434 INFO PublishSEListener -...

This is not Embed Player problem, this is server side issue, we confirm it for REST API too. We raised the ticket WCS-3358 to fix it and let you know results here.

Добрый день. Пожалуйста, проверьте, что выполнен шаг 4 отсюда. Пути к SDK и NDK необходимо корректировать в файле export/output/local/properties. Также убедитесь, что build.gradle в Android Studio открыт из каталога export/output.

Good day. You can try to use Web SDK build 0.5.28.2753.153 which contains a fix for Safari 14.0.2 on Mac OS Bug Sur publishing issue. But if you want the code to work in iOS webview, you should use at least build 2.0.171 which contains latest fixes for that. So we recommend to update your...

Please collect a report as described here and send using this form.

You can pull external RTMP stream using REST API POST /rest-api/pull/rtmp/pull HTTP/1.1 Host: localhost:8081 Content-Type: application/json { "uri":"rtmp://externalhost:1935/live/streamName?username=un&password=pwd", "localStreamName":"rtmp_for embed_player" } Then, you can play it...

В данном случае речь идет о публикации из любого другого браузера и просмотре такого потока в Safari. Диапазон в 100 кбит/с выглядит приемлемо.

Добрый день. Для 360p битрейт 450 кбит/с должен давать хорошее качество. Максимальные и минимальный битрейты, как правило, лучше не делать одинаковыми. Если они одинаковые, происходит следующее: 1. Допустим, битрейт падает ниже minBitrate. Сервер посылает REMB клиенту, чтобы тот поднял битрейт...

Good day. By default, if you did not change anything, EC2 instance Id is used as admin password in AWS instance. But you can reset it as follows: 1. Open the file /usr/local/FlashphonerWebCallServer/conf/database.yml in editor sudo nano /usr/local/FlashphonerWebCallServer/conf/database.yml 2...