Live and VOD
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 containers. Video is transcoded to ABR, and then is sent with adaptive streaming via CDN.
Recommended Video 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 | ||
| Video codecs* | See below | ||
| VOD max original file size | Up to 30GB | ||
| VOD containers | 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 ingest protocols | RTMP, RTMPS, SRT, RTSP, HLS PULL, WebRTC | ||
| LIVE keyframe frequency | 1–2s | ||
| Bitrate encoding | CBR | ||
| Pixel aspect ratio | Square | ||
| Chroma subsampling | 4:2:0 | ||
| Audio codecs* | AAC | ||
| Audio bitrate | 128 Kbps stereo | ||
| Audio sample rate | 44.1 / 48 kHz | ||
Supported Input Video Codecs
To ensure successful transcoding, your source video must use one of the supported input codecs listed below. If a video is uploaded or streamed with an unsupported codec, the transcoding process will fail, and an error will be returned. These input codecs are validated during the initial processing stage before any adaptive bitrate (ABR) renditions are created.| Codec | Sampling Depth | BPP | Format | Max Resolution |
|---|---|---|---|---|
| MPEG2 | 4:2:0 | 8 | Progressive, Interlaced | FHD |
| AVC | 4:2:0 | 8 | Progressive, Interlaced | 4K |
| HEVC | 4:2:0 | 8, 10, 12 | Progressive | 8K |
| HEVC | 4:2:2 | 8, 10, 12 | Progressive | 8K |
| HEVC | 4:4:4 | 8, 10, 12 | Progressive | 5K |
| VP9 | 4:2:0 | 8, 10, 12 | Progressive | 8K |
| VP9 | 4:4:4 | 8, 10, 12 | Progressive | 5K |
| AV1 | 4:2:0 | 8, 10 | Progressive | 8K |
LIVE stream ingest protocols
We support next procotols:- RTMP/S (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.
- WebRTC (Web Real-Time Communication) is a protocol designed for low-latency audio and video delivery over browsers and apps. It is supported only as an ingest format for bridge WebRTC to HLS/DASH. PUSH only.
- HLS PULL is a protocol of pulling (downloading) already prepared HLS stream from your encoder publicly open in the Internet, passing or transmuxing into another necessary protocol of delivery over CDN. PULL only.
- 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.