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You have a very poor channel to publish. Please test your channel bandwith with iperf3 via UDP iperf3.exe -c yourserver -p 5201 -u The result will show you maximum bitrate you can publish without packets losses. So you should set up your local network (router etc), or relocate your server to get...

Hello. Is the issue reproduced with latest WCS build (5.2.25) with generate_av_for_ua=all setting?

Hello. We've checked your recording. It seems like packets are lost periodically. This confirms recommendation to either relocate your instance closer to publisher or to publish stream from US to US. You also can use WebRTC over TCP to publish/play a stream, it may help to escape packet loss but...

Hello. Please clarify the following: 1. What is your stream resolution? 2. What bitrate is shown in chrome://webrtc-internals, bweforvideo section, googTransmitBitrate parameter when your stream becomes poor? Attach screenshot if possible. 3. What is displayed in Stats graphs for bweforvideo...

Hello This means your H264 stream goes with payload type (pt) 119, but RTMP pull agent payload type for H264 is 95 by default. To fix it create file rtmp_agent.sdp with the following content v=0 o=- 1988962254 1988962254 IN IP4 0.0.0.0 c=IN IP4 0.0.0.0 t=0 0 a=sdplang:en m=video 0 RTP/AVP 119...

Hello Try to setup bitrate settings in WCS_HOME/conf/flashphoner.properties webrtc_cc_min_bitrate=1000000 webrtc_cc_max_bitrate=1500000

Hello The investigation will take a time and there is a chance it will be fixed. However to manage streams more efficiently you can use direct and raw Websocket API. https://docs.flashphoner.com/display/WCS52EN/Raw+WebSocket+API So you can manage streams using raw browser getUserMedia() WebRTC...

It is because you pass frameRate 1. Full HD with 1 FPS has low bitrate. Please check this manual related bitrate management https://docs.flashphoner.com/display/WCS52EN/Bitrate+management+when+capturing+WebRTC+stream+in+browser As you can see you can set constant bitrate using SDP hooks only and...

Hello You have server-side transcoding VP8 > H.264 Try to avoid transcoding enabling H.264 as priority codec. WCS_HOME/conf/flashphoner.properties Find codecs=opus,alaw,ulaw,g729,speex16,mpeg4-generic,g722,telephone-event,vp8,h264,flv Replace with...

Hello This is how you can analyze black streams (version 5.1.3762): 1. Find the black stream using REST API /rest-api/stream/find {"name":"89994"} You should see the stream with PUBLISHING status 2. Check sessionId from found stream: Example: "sessionId"...

Make sure you have open UDP and TCP ports in range [30000-32000] by default. For now your UDP ports look closed. You can test your UDP ports as described in the docs: https://docs.flashphoner.com/display/WCS52EN/Accessory+tools#Accessorytools-Portroutingchecking

Hello We have checked logs. Server's performance looks good and tuned properly. We also noticed by IP that your instance is located in United States Virginia region. It is default AWS region. However this might be a network problem if your location is not US or if location of 500 viewers is not...

You can't set custom stream name for pulled streams rest-api/rest-api/pull/rtmp/pull However you can re-publish pulled stream with a new name https://docs.flashphoner.com/display/WCS52EN/To+another+RTMP+server 1. Add in flashphoner.properties rtmp_transponder_full_url=true 2. Do /pull/push {...

It seems your pulled stream has empty audio track. If so try to add setting: rtp_activity_detecting=false to prevent stopping by timeout

Hello Please attach full logfile WCS_HOME/logs/server_logs/flashphoner.log

We'll try to reproduce it on our test servers. Please clarify the following: 1. What picture breaks, WebRTC (on WCS), RTMP (on another server), or both? 2. What RTMP server you use (Wowza, anothe WCS instance...)? Also, please collect logs, traffic dumps and configs as we asked in post above.

Hello. You cannot add the application with existing key. On screenshot you've attached there is an application 'defaultApp' with URL http://localhost/rest-hooks on first string. So it should work. Please collect publishing or playing logs and server configs as described here and send to...

Hello. You are publishing FullHD stream while sharing 1080p screen in fact. So, try to limit publishing bitrate as described here to optimize channel load, it may help to reduce the latency.

Hello. We raised internal ticket (WCS-1844) and let you know when investigate it.

Hello. Please check if WebRTC stream is smooth. If not, try to adjust the settings (values are shown for example only) webrtc_cc_min_bitrate=1000000 webrtc_cc_max_bitrate=1000000 to get smooth picture of stream published. If WebRTC stream is smooth, but RTMP stream on RTMP server is still...