Search results

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    1080p RTMP

    Dear Max, We are running well with 720p -> Input RTMP and Output is HLS edge. On this scenario, if we watch WEBRTC stream from source server (rtmp input) and output HLS edge, the quality is the same. If we attempt to move to 1080p, including this on cdn_profiles, we have GOOD quality on...
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    Watch a VOD video - API

    Hello Max! Is there a a way to get notified by API once a VOD is played in totally - from begining to the end of file?
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    Canvas - Capture iFrame instead a video

    Hello Max, Checking Canvas DOCs, we notice that is loading a video example file. We would like to capture an iframe div or specific div on page. How to?
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    Feature Request - Mixer Caption & Voice Activity Color Picker

    Hello Max, We would like to request a feature: Actually we know is possible to display stream name on mixer, but we use stream name as ID to identify the user. Due some "special chars" we preffer do not use stream name as username. We would like to add a CAPTION text for each stream on...
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    FAILED Failed by ICE timeout

    Hello Max, We are testing another datacenter, and we got FAILED Failed by ICE timeout If we remove the LOCAL IP to WAN IP .. it works.
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    HLS Stream Quality

    Hello Max, We are confused about HLS. How to define 720p quality to HLS on edge? We start the HLS by API, but how to call an specific transcoder to them?
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    Get RTMP from another server like wowza or red5

    Hello! How can I push/pull a stream from another server ... example, we have a RTMP stream from Wowza, and would like to add it to our ROOM ... is there an API to this?
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    Vmix vs OBS - RTMP

    Hello! If we stream from VMIX, we got audio first .. later image. If we stream from OBS, audio & video are sync. Both streams are 0 losts and we monitor it 4 hours. Everytime SYNC AUDIO from VMIX is in front of video and OBS audio & video sync are floating ... sometimes audio is first...
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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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    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, 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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    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 # To get more settings: # ssh -p 2001 admin@localhost # default password: admin # show node-settings # show node-settings | grep port #server ip ip...
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    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...
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    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.
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    SIP Phone UI error

    Starting few days ago our users got this message once 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 'require', but...