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    MCU / ROOM name

    Hello! What is the difference from MCU / Simple mixer? We notice that MCU audio & video is fluid and good quality. We notice that MIXER without #room on name - just a mixer adding inputs, has few troubles with image and sound quality. Is there anything related? Is the #room making a difference?
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    Mixer Layout

    Nice job team! Works perfect. Thanks a lot.
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    Mixer not work on build 709 - Java 12

    Works like a charm! Thank you!
  4. R

    How to clear edge routes

    Dear Max, How to clear edge routes? Our edges show a lot of refused logs due to oldest IPs from CDN. Thanks,
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    Transcoder on Edge servers

    Hello Max, cdn_origin_to_origin_route_propagation=false We need this to be set to true. About our system: ADMIN AREA: Has a monitor page, where can watch all streams. Can Publish and Remove from Mixer and MCU all participants - using API. SPEAKER AREA: Has a MCU to talk with others...
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    Mixer not work on build 709 - Java 12

    Hello Max, We have updated to build 710. Same issue. I sent the report generated file as requested. Thanks!
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    RTMP -> HLS - Audio Wihout Sync

    Hello Max! Bandwidht is Ok, No Video_Lost and iperf is OK. The problem is related to HLS player. Stream A -> WebRTC -> SYNC Stream A -> HLS Play -> Start on SYNC and lost after 30 - 40 minuts - not recover.
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    Transcoder on Edge servers

    Hello Max, Actually we have: 1 WCS -> Origin USA - Receive all RTMP streams and process outside MIXER to end users 1 WCS -> Origin BR - Receive all webcams streams (near to publishers) Propagation route between origins are set to true. 1 WCS -> StandAlone USA - Receive all PUSHED streams from...
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    AbstractStunSocket - FScheduling-pool-44-thread-2 Can not find local candidate

    No, not working. We were forced to return to 654 build. Its a production server. I will try to launch a clone machine on google to reproduce it outside our production side.
  10. R

    Can you please link ticket WCS-2753 to my account

    We are attempting to use Google API to monitor that instance and check if there is users connect to .. if not, shutdown from LB. Thanks Max!
  11. R

    RTMP -> HLS - Audio Wihout Sync

    Dear Max, We observe playing origin using Flashphoner Player - WebRTC, audio & video from rtmp is sync. After few minuts, we can see lost of audio and video sync but its recovery itself. Using HLS player - HLS.JS lost audio / video sync and not recover. Regards, Rafael
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    Mixer not work on build 709 - Java 12

    Hello! Our mixer stop works with build 709 - Java 12. Its working on build 654 - Java 8. # Config flashphoner.properties # To get more settings: # ssh -p 2001 admin@localhost # default password: admin # show node-settings # show node-settings | grep port #server ip ip...
  13. R

    Mixer Debug - Crash

    Hello! If debug is active and display streamname mixer does not start. WCS: 5.2.654 (with java 1.8 works) WCS: 5.2.709 (tested only with java 12) Java Version: 12.0.2+10 23:11:47,794 ERROR MixerAgent - MIXER-AGENT-mixer://room-51-cc11dbf6-48bc-4572-a446-21832d08b308 Mixer closed...
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    AbstractStunSocket - FScheduling-pool-44-thread-2 Can not find local candidate

    Hello! Once we attempt to add a participant to mixer / mcu, we got: 20:43:54,083 WARN AbstractStunSocket - FScheduling-pool-44-thread-2 Can not find local candidate for /X.X.X.X We are using Google Cloud.
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    RTMP -> HLS - Audio Wihout Sync

    Hello! We are streaming from VMIX to Youtube and FlashPhoner. On Youtube we got SYNC audio x video. On our structure, we got Audio and Video after some time, lost the sync and do not recover. We have an ORIGIN server and EDGE on Google Cloud. We can see the lost of SYNC on WebRTC and M3MU...
  16. R

    Call Duration

    Hello! There is an way to receive back from hooks or from API the duration of a CALL / ROOM? For example, Room A is started at: 12:00:00 and Room A is closed at: 12:35:00 -> Duration of this room: 35 minuts.
  17. R

    Can you please link ticket WCS-2753 to my account

    Today we experience an issue with Google Cloud. An instance has been "shutdown" from LB but users were connected to them. As work arround we edited the Instance Group and change to INCREASE ONLY. Decrease will be manually done until we find another way.
  18. R

    SIP Phone UI error

    UPDATE: Now is OK with this workarround. Was just local cache loading an oldest version of JS. Now is Ok and we hope a new release comes soon with the final solution. Thanks for all
  19. R

    SIP Phone UI error

    I revert back to 1963 and change the code. Error Displayed once we attempt to answer a call: WebRtcMediaConnection - onSetRemoteDescriptionErrorCallback(): error: OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: rtcpMuxPolicy is...
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    SIP Phone UI error

    Can you inform me what is the oldest working build?
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