rtmp republishing

Не могу найти конфигурацию для ретрансляции в Телеграм, пробовал на приложенной + вариант для youtube https://url/rest-api/push/startup { "streamName": "streamName", "rtmpUrl": "rtmp://dc4-1.rtmp.t.me/s/key", "rtmpTransponderFullUrl": true } Возможно задача решалась и есть...

Problems have arisen when I try to live-stream flash phone to YouTube. All other Rtmp outputs function normally; only YouTube is a problem child. moto x3m

Hi we have an issue to stream from flashphoner to YouTube. The other Rtmp output works fine just YouTube does not work.

Hello @Max How are you? We are your old customer. W had cancelled that service 3 years ago due to some other features, Now we want to again use that feature. We have installed web call server 5.2 on our server but its not going live to Facebook. Can you please check our server. I have emailed...

seems youtube rejected the multistream I dont know why after 30 seconds got closed. but via obs work normal ``` { "mediaSessionId": "", "streamName": "rtmp_73fe7866-d8a3-47df-a998-db60945006fd", "rtmpUrl": "rtmp://a.rtmp.youtube.com/live2/aaaa-aaaa-aaaa-aaaa", "rtmpFlashVersion"...

while publishing from obs failed when try add push startup with same name [ { "appKey": "flashStreamingApp", "sessionId": "", "mediaSessionId": "9a991c81-7b5f-4f51-aa5e-464f44e837c1", "name": "stream1", "published": true, "hasVideo": true...

I'm running the MCU client example: https://demo.flashphoner.com:8888/client2/examples/demo/streaming/mcu_client/mcu_client.html Seems to work nicely, but where / how can I see the output mosaic published through RTMP? I do not see any docs explaining where to find this on Flashphone's site...

I am running the simple screen sharing demo described here: https://flashphoner.com/how-to-embed-webrtc-screen-sharing-into-a-web-page-2/ In screen-sharing-min.js I pass an rtmpUrl to the session.CreateStream() function: function startStreaming() { var constraints = { video: {}...

Hello. I'm trying to do rtmp stream with 720p over phone, but I have some video glitches. Once I updated the codes the glitches are gone, but the quality is bad. codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,vp8,h264,flv,mpv The rest of my server config is...

Здравствуйте. Версия 5.2.969 Пробуем опубликовать микшированный поток на другой RTMP-сервер (https://gcorelabs.com/ru/support/articles/360014153118/) Запрос: { "streamName": "name", "rtmpUrl": "rtmp://vp-push-ix1.gvideo.co/in/130062?*key*/", "rtmpTransponderFullUrl": false, "options": {} }...

the RTMP client is connected, but I recieve logs: server part 1 validation failed for type 1, will try with type 2 ERROR ClientHandler - RTMP error server part 1 validation failed even for type: 2

Какой будет наиболее простой способ добавить надпись (и менять ее когда нужно) в обычный транслируемый с камеры поток? В этой теме советуют делать на клиенте через фильтры, в том числе чтобы уменьшить нагрузку на сервер. Это действительно самый простой способ для трансляций которые идут через...

We are setting up our application for live streaming using "Flashphoner WCS Server", during live stream we are able to find all stream using API Call (/rest-api/push/find_all) but when we are requesting mute with mediaSessionId on this endpoint (/rest-api/push/mute) we are getting Internal...

The config as below: Video Cam->RTSP->WCS->WebRTC->Browser, Is it possible to add some buffering of about 0.25s to 0.5s delay, so that the video can be smoother on network jittering, and i don't mine the delay within about 1 seconds. Thanks

взял код из примера, пытаюсь создать стрим в ютубе, через 2 мин после старта падает с ошибкой, стрим до ютуба так и не дошел {message: "notifyStreamStatusEvent", data: [{,…}]} data: [{,…}] 0: {,…} audioBitrate: 0 audioCodec: "opus" audioState: {muted: false} bitrate: 0 codecImpl: ""...

Hi guys, Can someone help me ? I am looking for a room with 2 (with cam) or more participants and with screen sharing and re-transmitting to RTMP. Thanks a lot,

I'm just starting with flashphonner. So, using server on AWS 1) to flashphoner.properties added 2 lines rtmp_transponder_stream_name_prefix= rtmp_transponder_full_url=true 2) created live stream on youtube 3) send POST request to `rtsp/startup` uri...

Can I capture video from web browser, passing through webrtc server and then streaming to rtmp Amazon ivs? (rtmp server url and stream key) thanks in advanced

exception "java.lang.NullPointerException" path "/rest-api/push/terminate" error "Internal Server Error" message null timestamp 1611119681055 status 500

I want to setup one to many live video broadcasting system for all web browser and all mobile browser in web. but here issue become in mobile browser broadcasting system. i am tired to connect webrtc to rtmp. here your demo link not working in mobile chrome ios and android...