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Hello We did not test playback with iOS 11 beta, Safari. Once we have any results, I will update this thread.

Hi Yes we can implement such filter. Please contact sales@flashphoner.com to get a quote

Да, можете. Точнее, транскодинг H.264 + G.711 > VP8 + Opus будет выполнен автоматически, если на сервере задан приоритет кодеков в настройке flashphoner.properties: codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,vp8,h264,flv,mpv Здесь Opus имеет приоритет над G.711. VP8...

Hello To decrease CPU usage you have to 1. Set H.264 codec as priority in flashphoner.properties codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv 2. For WebRTC, make sure you DO NOT set resolution when you play stream. For example...

У нас не воспроизводится описанная проблема. Возможно ошибка измерения битрейта, или другие настройки. Мы тестируем последний билд 2325. Настройки кодеков default (flashphoner.properties): codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv Т.е. Chrome...

It is 2-3 seconds for me. Our demo server is located n Europe / Frankfurt. You can have such initial delay when: 1) H.264 codec is used. 2) Windows 8 is used 3) There is a packet lost on the network between server and you As an option you can switch to VP8 codec flashphoner.properties...

We have added CORS headers for API urls like: /rest-api/stream etc So it should work cross-domain now. You can update to the latest build 2323 service webcallserver update

Do you have the same issue with our demo page https://wcs5-eu.flashphoner.com/demo2/two-way-streaming Test 1. Publish stream left. 2. Play stream right. I have about 2 seconds delay when I press Play. Do you have 5-6 seconds delay here?

If you mean ffmpeg, you can try to set a short GOP (K-frame interval). For example: 48 means K frame every 48-th frame. If FPS is 30 (30 frames per second), the K frame interval should be equal 1.5 seconds. Shorter K frame interval should provide shorter initial delay.

We can add a simple regexp filter for stream name. For example, name:stream* To select all streams stream1, stream2, etc. It should help to reduce selection weight.

As you can see from docs, https://flashphoner.com/docs/wcs5/wcs_docs/html/en/wcs-rest-api/index.html?return_objects.htm Stream has following filters: name mediaSessionId status published name is name of stream, for examples room-alice-b7c6 (for stream participating in the room) published -...

iOS does not support WebRTC in Chrome and Safari browsers. You can't publish stream from Chrome or Safari on iOS. Just from a native app.

It works for me if I enable Adobe Flash Player plugin in my Safari browser on Mac Sierra. Settings - Security - Plugins - Enable Adobe Flash Player When I do this, I'm able to publish stream from your page: https://buckworm.com/customers/newTestPublisher.php?stramName=buckwormTest Then I'm able...

iOS Safari does not support video publishing. Apple announces WebRTC support for iOS Safari this year. So with iOS Safari you can just play a stream. Not publish. If you want to publish from iOS, you have to consider native iOS app and iOS SDK.

We still need step-by-step instruction how to reproduce this. Example: 1. Make video call (Call1) from Android Phone Video sample to Softphone (Bria4). 2. Make audio call (Call2) to mobile phone from Android Phone samle. 3. Pick up mobile phone. 4. Hear audio from Call2 in Call1. Please describe...

Hello We have fixed similar behavior for WebRTC only (Chrome, Firefox). Build 2317. You can do: service webcallserver update The similar means we have created a model emulating your streams. Please let us know if you still have the issue in your production environment with latest build. For...

Sorry, it is not enough clear. Please describe in detail what behavior do you have and what behavior do you expect to see? For now I can assume you mean echo effect?

Исправили в 2319. Сборка доступна на демо-сервере: https://wcs5-eu.flashphoner.com/demo2/video-chat Видео есть с двух сторон.

It was a regression. Should be fixed in 2319. Please check. service webcallserver update

You can play a stream via Room API if it is a stream of a joined participant. ffmpeg with its RTMP stream can't join as a participant. Therefore you can't play ffmpeg's stream inside the room. As a workaround, you can create a new session as in the normal player example...