Codecs and protocols
Accepted input parameters of your live streams and videos
List of input parameters for transcoding original videos
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 | ||
| Audio codecs | AAC, MP3 | ||
| 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 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 | ||
| Audio sample rate | 44.1 / 48 kHz | ||
| Audio bitrate | 128 Kbps stereo | ||
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 |
Multiple Live Ingesting Points
For smoother and more reliable streaming, we offer entry servers in key regions including Luxembourg, Ashburn, Miami, and Singapore. By connecting your streaming equipment to the nearest upload server, you can minimize latency and improve performance before the stream is distributed globally through our CDN. This feature is especially valuable if your streams originate from multiple locations worldwide. You can specify preferred upload servers and the number of streams per region. Our team will then configure your account to match your streaming setup. Reach out to our support team or your account manager for setup assistance or additional details.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.
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 the 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 the protocol is rtmps://. |
| Your encoder doesn’t support RTMPS | Verify whether your encoder supports RTMPS. If not, consider switching to one that does. |