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.
Recommended stream parameters
We recommend the following parameters for streams:
|
Parameters |
Value |
||
|
Video bitrate and resolution |
Quality |
Resolution
|
Video bitrate range |
|
4k |
3840x2160 |
20,000–51,000 Kbps (60 fps), 13,000–34,000 Kbps (30 fps) |
|
|
1440 |
2560x1440 |
9,000–18,000 Kbps (60 fps), 6,000–13,000 Kbps (30 fps) |
|
|
1080 |
1920x1080 |
4,500–9,000 Kbps (60 fps), 3,000–6,000 Kbps (30 fps) |
|
|
720 |
1280x720 |
2,250–6,000 Kbps (60 fps), 1,500–4,000 Kbps (30 fps) |
|
|
480 |
854x480 |
500–2,000 Kbps |
|
|
Frame rate
|
Up to 60 fps |
||
|
Audio codec |
AAC, MP3 |
||
|
Video codec |
H.264, H.265, AV1 |
||
|
Max original file size (VOD)
|
up to 30 GB |
||
|
Container (VOD) |
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) | RTMP, RTMPS, SRT, RTSP, HLS, WebRTC | ||
|
Keyframe frequency (Live) |
1-2s |
||
|
Bitrate encoding |
CBR |
||
|
Pixel aspect ratio |
Square |
||
|
Chroma subsampling |
[4:2:0](https://en.wikipedia.org/wiki/Chroma_subsampling) |
||
|
Audio sample rate |
44.1 kHz |
||
|
Audio bitrate |
128 Kbps stereo |
||
If the recommended parameters do not suit your stream (codecs, custom FPS, ProRes, High 4:4:4, Enhanced RTMP, etc.), write to us in the chat, send an email to support@gcore.com, or contact your manager to find the solution.
RTMP, RTMPS, and SRT for live streaming
- 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.
- RTSP (Real Time Streaming Protocol) is a communication protocol used to control servers that stream media content. RTSP uses the Real-time Transport Protocol (RTP) with Real-time Control Protocol (RTCP) to deliver media streams.
- SRT (Secure Reliable Transport) is an open-source video transport protocol for delivering high-quality, secure, low-latency video across unreliable networks.
You can use Push or Pull methods where applicable.
RTMP(S) troubleshooting
| Error | Cause | Solution |
| SSL issues | You used rtmps://but in the encoder rtmp://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:
|
| “Connection timed out” | The server URL is incorrect | Check the server URL in the encoder settings. Ensure that protocol is rtmps://. |
| Your encoder doesn’t support RTMPS | Check if there is RTMPS support, change encoder if required |
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