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Accepted initial parameters of your live streams and videos

Supported parameters

Gcore Video Streaming supports:

  • Receiving live streams from your server (PULL) or a dedicated publishing point (PUSH) using numerous protocols including RTMP(S) and SRT. All supported live protocols are listed in the table below. The stream is transcoded and sent with adaptive streaming via CDN in HLS/MPEG-DASH (CMAF low latency) formats.
  • Videos uploaded in almost any format, from standard MP4 to 4K HDR Video, are first transcoded to get videos of lower quality. Then, they're sent with adaptive streaming via CDN in HLS format.

We recommend the following parameters for streams:



Video bitrate and resolution





Video bitrate range



20,000–51,000 Kbps (60 fps), 

13,000–34,000 Kbps (30 fps)



9,000–18,000 Kbps (60 fps), 

6,000–13,000 Kbps (30 fps)



4,500–9,000 Kbps (60 fps), 

3,000–6,000 Kbps (30 fps)



2,250–6,000 Kbps (60 fps),

1,500–4,000 Kbps (30 fps) 



500–2,000 Kbps

Frame rate


Up to 60 fps

Audio codec


Video codec

H.264, H.265

Max original file size (Video Hosting)


30 GB

Container (Video Hosting)

3g2, 3gp, asf, avi, dif, dv, flv, f4v, m4v, mov, mp4, mpeg, mpg, mts, m2t, m2ts, qt, wmv, vob, mkv, ogv, webm, vob, ogg, mxf, quicktime, x-ms-wmv, mpeg-tts, vnd.dlna.mpeg-tts

Live protocols (Live Streaming) RTMP, RTMPS, SRT, RTSP, HLS

Keyframe frequency (Live Streaming)

2s (max 4s)

Bitrate encoding


Pixel aspect ratio


Audio sample rate

44.1 kHz 

Audio bitrate

128 Kbps stereo

If the recommended parameters do not suit your stream, write to us in the chat or via support@gcore.com, or contact your manager to find the solution.

RTMP, RTMPS, and SRT for live streaming

  • RTMP(S) and SRT are supported protocols for income live streams.
  • RTMP (Real-Time Messaging Protocol) is a protocol for transmitting audio, video, and data over the Internet between a player and a server, supporting low-latency communication for real-time streaming. RTMPS is a variation of RTMP but incorporates SSL usage.
  • SRT (Secure Reliable Transport) is an open-source video transport protocol for delivering high-quality, secure, low-latency video across unreliable networks.

RTMP(S) troubleshooting

Error Cause Solution
SSL issues You used
but in the encoder
is specified
Check the protocol in your encoder. Follow step 3 of the guide.
You used a port (80) unsuitable for secure data transfer Manually add a correct port (443) to the server link, e.g.: rtmp://vp-push-ed1.gvideo.co:443/in/
No transcoding or image degradation when using web cameras with custom video codecs The video codec H264+ extension has an over-increased keyframe Check the outgoing live stream parameters of the web camera:
  • Streams must be encoded by H264. No hacks/codecs like H264+ maybe be used.
  • “Smart” mode codecs must be turned off.
Web camera codec settings
“Connection timed out” The server URL is incorrect Check the server URL in the encoder settings. Ensure that protocol is
Your encoder doesn’t support RTMPS Check if there is RTMPS support, change encoder if required

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