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# stream
  • d
    Is this the right place to report a typo in the api docs for stream?
  • r
    Here or a Pull Request into the docs also works!
  • d
    I don't see where to make a pr for api docs, but here https://api.cloudflare.com/#stream-videos-create-signed-url-tokens-for-videso
  • j
  • j
    I hope this will help
  • r
    Morning folks.. i have another question and a typo... 1) is there a way to add a username/passwd param when configuring a RTMP output, for an endpoint that requires it? 2) Text Typo on Outputs section - Live Video Inputs can be set up to forward any input to another input. (i think this should read any input to another output)
    b
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  • b
    Thanks! Will update
  • d
    Does anyone know of a web player for sub 1s latency?
  • d
    Or would a wasm of ffmpeg (https://github.com/ffmpegwasm/ffmpeg.wasm) work to get the same feature?
  • i
    The latency isn't in the player, it's in the encoding.
  • i
    Standard HLS takes about 10 to 30 seconds depending on settings.
  • i
    You can use WebRTC though, for sub-second latency.
  • i
  • d
    I couldn't figure out the player part, but I found it https://github.com/cloudflare/templates/blob/main/stream/webrtc/src/whep.html
  • d
    rtmps works fine with OBS, but trying the srt url errors out.
    Copy code
    accesslog
12:47:21.493: [obs-ffmpeg mpegts muxer: 'adv_stream'] info: Output format name and long_name: mpegts, MPEG-TS (MPEG-2 Transport Stream)
12:47:21.493: 
12:47:21.493: [obs-ffmpeg mpegts muxer / libsrt] : libsrt v.1.5.0 loaded.
12:47:21.493: 
12:47:21.552: [obs-ffmpeg mpegts muxer / libsrt] : Connection to srt://live.cloudflare.com:778?passphrase=<redacted>&streamId=<redacted> failed: I/O error
12:47:21.552: 
12:47:21.554: Couldn't open 'srt://live.cloudflare.com:778?passphrase=<redacted>&streamId=<redacted>', I/O error
12:47:21.554: [obs-ffmpeg mpegts muxer: 'adv_stream'] failed to open the url
12:47:21.554: Output 'adv_stream':  Reconnecting in 2.00 seconds..
    b
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  • d
    So based on my tests (1080p70 @ 12000kbps), using SRT input and HLS out has a delay of 45 seconds. Using SRT input with dash gets a delay of 22 seconds. But the problem with webrtc is the input needs to be webrtc too which tools like OBS can't
    b
    i
    z
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  • d
    I see the sub 1s latency examples but they use ffplay and I don't know of a way to get that in html5 (was looking at wasm but got lost in a rabbit hole)
  • e
  • d
    It only streams to 
    Millicast.com
    not any webrtc url
  • d
    Copy code
    <video src="https://customer- .cloudflarestream.com/ /manifest/video.m3u8" controls="">
be right back</video>
    is this tag correct for stream? firefox says unsupported format
  • s
    Streaming HLS videos (.m3u8 ext) using a video tag is unsupported in most browsers except Safari and Edge I guess. Better use the Stream Player instead https://developers.cloudflare.com/stream/viewing-videos/using-the-stream-player/
  • j
    is there a possibility to do : rtmps ingest out to Webrtc or do input type need to be the same output type ?
  • j
    I'm getting this error in chrome but not firefox for video with signed_urls, I assume this is because of some header that chrome sends but firefox doesn't?
  • j
    Copy code
    var signed_url_restrictions = {
            exp: Math.floor(Date.now() / 1000) + 3600,
            downloadable: false,
            accessRules:[{"type":"ip.src","ip":[DANGEROUS_SQL_IP],"action":"allow"}, {"type": "any","action": "block"}]
        };
  • j
    those are the rules I have
  • j
    oh nevermind lmao
  • j
    Cloudflare WARP just isn't proxying chrome lmao
  • j
    therefor it's a different IP
  • e
    That's not possible right now but it's on the teams radar to add in future
  • e
    Currently webrtc needs to be used for both in and out to work
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