
ffmpeg-streaming
by JosiahSiegel
SKILL.md
name: ffmpeg-streaming description: Complete live streaming and protocol system for FFmpeg 7.1 LTS and 8.0.1 (latest stable, released 2025-11-20). PROACTIVELY activate for: (1) RTMP streaming to Twitch/YouTube/Facebook, (2) HLS output and adaptive bitrate (ABR), (3) DASH streaming setup, (4) Low-latency streaming (LL-HLS, LL-DASH), (5) SRT protocol configuration, (6) WebRTC/WHIP sub-second latency (FFmpeg 8.0+), (7) Protocol conversion (RTMP to HLS), (8) Multi-destination streaming, (9) nginx-rtmp integration, (10) Docker streaming services. Provides: Platform-specific stream commands, ABR ladder examples, encryption setup, latency optimization, WHIP authentication, production patterns. Ensures: Reliable live streaming with optimal quality and latency.
CRITICAL GUIDELINES
Windows File Path Requirements
MANDATORY: Always Use Backslashes on Windows for File Paths
When using Edit or Write tools on Windows, you MUST use backslashes (\) in file paths, NOT forward slashes (/).
Quick Reference
| Protocol | Latency | Output Format | Best For |
|---|---|---|---|
| RTMP | 2-5s | -f flv rtmp://... | Twitch, YouTube ingest |
| HLS | 10-30s | -f hls output.m3u8 | Web playback, CDN |
| LL-HLS | 2-5s | -hls_flags low_latency | Low-latency web |
| SRT | <1s | -f mpegts srt://... | Reliable contribution |
| DASH | 10-30s | -f dash output.mpd | ABR streaming |
| Platform | Ingest URL |
|---|---|
| Twitch | rtmp://live.twitch.tv/app/{key} |
| YouTube | rtmp://a.rtmp.youtube.com/live2/{key} |
rtmps://live-api-s.facebook.com:443/rtmp/{key} |
When to Use This Skill
Use for live streaming workflows:
- Broadcasting to Twitch, YouTube, Facebook
- Setting up HLS/DASH servers for VOD
- Building ABR encoding ladders
- Low-latency streaming requirements
- Multi-destination restreaming
FFmpeg Streaming Guide (2025)
Complete guide to live streaming with FFmpeg covering RTMP, HLS, DASH, SRT, and WebRTC protocols.
Current Latest: FFmpeg 8.0.1 (released 2025-11-20) - Check with ffmpeg -version
Streaming Protocols Overview
Protocol Comparison (2025)
| Protocol | Latency | Compatibility | Security | Use Case |
|---|---|---|---|---|
| RTMP | 1-5s | Legacy, ingestion | TLS (RTMPS) | Stream ingestion |
| HLS | 6-30s | Universal | AES-128 | CDN delivery |
| DASH | 6-30s | Wide | DRM-ready | ABR streaming |
| LL-HLS | 2-4s | Apple/modern | AES-128 | Low-latency |
| LL-DASH | 2-4s | Modern browsers | DRM-ready | Low-latency |
| SRT | <1s | Growing | AES-128/256 | Contribution |
| WebRTC | <0.5s | Browsers | DTLS/SRTP | Real-time |
| WHIP | <1s | Emerging | TLS | WebRTC ingestion |
2025 Trends
- SRT replacing RTMP for contribution/ingest
- LL-HLS becoming default for Apple ecosystem
- WHIP standardizing WebRTC ingestion (FFmpeg 8.0+)
- QUIC/HTTP3 emerging for lower latency delivery
RTMP Streaming
Basic RTMP Stream
# Stream to RTMP server
ffmpeg -re -i input.mp4 \
-c:v libx264 -preset veryfast -b:v 3000k -maxrate 3000k -bufsize 6000k \
-c:a aac -b:a 128k \
-f flv rtmp://server/live/stream_key
Webcam/Screen to RTMP
# Linux webcam
ffmpeg -f v4l2 -i /dev/video0 \
-f alsa -i default \
-c:v libx264 -preset ultrafast -tune zerolatency -b:v 2500k \
-c:a aac -b:a 128k \
-f flv rtmp://server/live/stream_key
# macOS webcam
ffmpeg -f avfoundation -framerate 30 -i "0:0" \
-c:v libx264 -preset ultrafast -tune zerolatency -b:v 2500k \
-c:a aac -b:a 128k \
-f flv rtmp://server/live/stream_key
# Windows screen capture
ffmpeg -f gdigrab -framerate 30 -i desktop \
-c:v libx264 -preset ultrafast -tune zerolatency -b:v 3000k \
-c:a aac -b:a 128k \
-f flv rtmp://server/live/stream_key
RTMP with GPU Encoding
# NVIDIA NVENC
ffmpeg -re -i input.mp4 \
-c:v h264_nvenc -preset p3 -tune ll -zerolatency 1 -b:v 6000k \
-c:a aac -b:a 128k \
-f flv rtmp://server/live/stream_key
# Intel QSV
ffmpeg -re -init_hw_device qsv=hw -filter_hw_device hw \
-i input.mp4 \
-c:v h264_qsv -preset fast -b:v 6000k \
-c:a aac -b:a 128k \
-f flv rtmp://server/live/stream_key
RTMPS (Encrypted)
ffmpeg -re -i input.mp4 \
-c:v libx264 -preset veryfast -b:v 3000k \
-c:a aac -b:a 128k \
-f flv rtmps://secure-server/live/stream_key
Twitch/YouTube/Facebook
# Twitch
ffmpeg -re -i input.mp4 \
-c:v libx264 -preset veryfast -b:v 6000k -maxrate 6000k -bufsize 12000k \
-g 60 -keyint_min 60 \
-c:a aac -b:a 160k -ar 44100 \
-f flv rtmp://live.twitch.tv/app/YOUR_STREAM_KEY
# YouTube
ffmpeg -re -i input.mp4 \
-c:v libx264 -preset veryfast -b:v 4500k -maxrate 4500k -bufsize 9000k \
-g 60 -keyint_min 60 \
-c:a aac -b:a 128k -ar 44100 \
-f flv rtmp://a.rtmp.youtube.com/live2/YOUR_STREAM_KEY
# Facebook
ffmpeg -re -i input.mp4 \
-c:v libx264 -preset veryfast -b:v 4000k -maxrate 4000k -bufsize 8000k \
-g 60 \
-c:a aac -b:a 128k -ar 44100 \
-f flv rtmps://live-api-s.facebook.com:443/rtmp/YOUR_STREAM_KEY
HLS Streaming
Basic HLS Output
# Create HLS stream
ffmpeg -i input.mp4 \
-c:v libx264 -c:a aac \
-hls_time 6 \
-hls_list_size 0 \
-hls_segment_filename "segment_%03d.ts" \
playlist.m3u8
Live HLS Streaming
# Live HLS with rolling window
ffmpeg -re -i input.mp4 \
-c:v libx264 -preset veryfast -b:v 3000k \
-c:a aac -b:a 128k \
-hls_time 4 \
-hls_list_size 10 \
-hls_flags delete_segments \
-hls_segment_filename "live_%03d.ts" \
live.m3u8
Adaptive Bitrate HLS (ABR)
# Multi-bitrate HLS
ffmpeg -i input.mp4 \
-filter_complex "[0:v]split=3[v1][v2][v3]; \
[v1]scale=1920:1080[v1out]; \
[v2]scale=1280:720[v2out]; \
[v3]scale=854:480[v3out]" \
-map "[v1out]" -c:v:0 libx264 -b:v:0 5000k -maxrate:v:0 5350k -bufsize:v:0 7500k \
-map "[v2out]" -c:v:1 libx264 -b:v:1 2800k -maxrate:v:1 2996k -bufsize:v:1 4200k \
-map "[v3out]" -c:v:2 libx264 -b:v:2 1400k -maxrate:v:2 1498k -bufsize:v:2 2100k \
-map a:0 -c:a:0 aac -b:a:0 192k \
-map a:0 -c:a:1 aac -b:a:1 128k \
-map a:0 -c:a:2 aac -b:a:2 96k \
-f hls \
-hls_time 6 \
-hls_list_size 0 \
-var_stream_map "v:0,a:0 v:1,a:1 v:2,a:2" \
-master_pl_name master.m3u8 \
-hls_segment_filename "stream_%v/segment_%03d.ts" \
stream_%v/playlist.m3u8
HLS with AES Encryption
# Generate encryption key
openssl rand 16 > enc.key
echo "http://example.com/enc.key" > enc.keyinfo
echo "enc.key" >> enc.keyinfo
openssl rand -hex 16 >> enc.keyinfo
# Encrypted HLS
ffmpeg -i input.mp4 \
-c:v libx264 -c:a aac \
-hls_time 6 \
-hls_key_info_file enc.keyinfo \
-hls_segment_filename "encrypted_%03d.ts" \
encrypted.m3u8
Low-Latency HLS (LL-HLS)
# LL-HLS with partial segments
ffmpeg -re -i input.mp4 \
-c:v libx264 -preset veryfast -tune zerolatency \
-c:a aac \
-hls_time 4 \
-hls_list_size 6 \
-hls_flags independent_segments+delete_segments \
-hls_segment_type fmp4 \
-hls_fmp4_init_filename init.mp4 \
-hls_segment_filename "ll_%03d.m4s" \
ll_playlist.m3u8
DASH Streaming
Basic DASH Output
ffmpeg -i input.mp4 \
-c:v libx264 -c:a aac \
-f dash \
-seg_duration 4 \
-use_timeline 1 \
-use_template 1 \
output.mpd
Adaptive DASH
ffmpeg -i input.mp4 \
-map 0:v -map 0:v -map 0:v -map 0:a \
-c:v libx264 -c:a aac \
-b:v:0 5M -s:v:0 1920x1080 \
-b:v:1 3M -s:v:1 1280x720 \
-b:v:2 1M -s:v:2 640x360 \
-b:a 128k \
-f dash \
-seg_duration 4 \
-adaptation_sets "id=0,streams=v id=1,streams=a" \
manifest.mpd
Low-Latency DASH (LL-DASH)
ffmpeg -re -i input.mp4 \
-c:v libx264 -preset ultrafast -tune zerolatency \
-c:a aac \
-f dash \
-seg_duration 1 \
-frag_duration 0.1 \
-streaming 1 \
-ldash 1 \
-use_timeline 0 \
-use_template 1 \
-remove_at_exit 1 \
-window_size 5 \
ll_manifest.mpd
SRT Streaming (FFmpeg 4.0+)
SRT Listener (Server Mode)
# FFmpeg as SRT server
ffmpeg -re -i input.mp4 \
-c:v libx264 -preset veryfast -b:v 3000k \
-c:a aac -b:a 128k \
-f mpegts "srt://0.0.0.0:9000?mode=listener"
SRT Caller (Client Mode)
# Send to SRT server
ffmpeg -re -i input.mp4 \
-c:v libx264 -preset veryfast -b:v 3000k \
-c:a aac -b:a 128k \
-f mpegts "srt://server:9000?mode=caller"
SRT with Encryption
# AES-128 encrypted SRT
ffmpeg -re -i input.mp4 \
-c:v libx264 -c:a aac \
-f mpegts "srt://server:9000?passphrase=mySecretKey123&pbkeylen=16"
Receive SRT Stream
# Receive and save
ffmpeg -i "srt://0.0.0.0:9000?mode=listener" \
-c copy output.mp4
# Receive and transcode
ffmpeg -i "srt://server:9000?mode=caller" \
-c:v libx264 -c:a aac \
output.mp4
WebRTC/WHIP (FFmpeg 8.0+)
Overview
FFmpeg 8.0 introduces the WHIP (WebRTC-HTTP Ingestion Protocol) muxer, enabling sub-second latency streaming to WebRTC endpoints. WHIP is becoming the standard for WebRTC ingestion, replacing proprietary solutions.
WHIP Output
# Stream to WHIP endpoint
ffmpeg -re -i input.mp4 \
-c:v libx264 -preset ultrafast -tune zerolatency \
-c:a libopus \
-f whip \
"https://whip-server.example.com/publish/stream_id"
# Low-latency webcam to WHIP (Linux)
ffmpeg -f v4l2 -i /dev/video0 \
-f pulse -i default \
-c:v libx264 -preset ultrafast -tune zerolatency -b:v 2500k \
-c:a libopus -b:a 64k \
-f whip \
"https://whip-server.example.com/publish/stream_id"
# Low-latency webcam to WHIP (macOS)
ffmpeg -f avfoundation -framerate 30 -i "0:0" \
-c:v libx264 -preset ultrafast -tune zerolatency -b:v 2500k \
-c:a libopus -b:a 64k \
-f whip \
"https://whip-server.example.com/publish/stream_id"
# Low-latency screen capture to WHIP (Windows)
ffmpeg -f gdigrab -framerate 30 -i desktop \
-c:v libx264 -preset ultrafast -tune zerolatency -b:v 3000k \
-c:a libopus -b:a 64k \
-f whip \
"https://whip-server.example.com/publish/stream_id"
WHIP with Hardware Encoding
# NVIDIA NVENC WHIP streaming
ffmpeg -hwaccel cuda -hwaccel_output_format cuda \
-i input.mp4 \
-c:v h264_nvenc -preset p3 -tune ll -zerolatency 1 -b:v 3000k \
-c:a libopus -b:a 64k \
-f whip \
"https://whip-server.example.com/publish/stream_id"
# Intel QSV WHIP streaming
ffmpeg -init_hw_device qsv=hw -filter_hw_device hw \
-i input.mp4 \
-c:v h264_qsv -preset fast -b:v 3000k \
-c:a libopus -b:a 64k \
-f whip \
"https://whip-server.example.com/publish/stream_id"
WHIP Authentication
# With bearer token authentication
ffmpeg -re -i input.mp4 \
-c:v libx264 -preset ultrafast -tune zerolatency \
-c:a libopus \
-f whip \
-headers "Authorization: Bearer YOUR_TOKEN" \
"https://whip-server.example.com/publish/stream_id"
WHIP Servers/Platforms
- Cloudflare Stream - WebRTC ingest via WHIP
- Dolby.io - Interactive streaming
- Janus - Open-source WebRTC gateway
- mediasoup - Open-source SFU with WHIP support
- LiveKit - WebRTC platform with WHIP ingestion
Multicast Streaming
UDP Multicast
# Send multicast
ffmpeg -re -i input.mp4 \
-c:v libx264 -c:a aac \
-f mpegts "udp://239.0.0.1:1234?pkt_size=1316"
# Receive multicast
ffmpeg -i "udp://239.0.0.1:1234" -c copy output.mp4
Re-streaming (Protocol Conversion)
RTMP to HLS
# Receive RTMP, output HLS
ffmpeg -i rtmp://source/live/stream \
-c:v libx264 -preset veryfast \
-c:a aac \
-hls_time 4 \
-hls_list_size 10 \
-hls_flags delete_segments \
/var/www/html/hls/stream.m3u8
SRT to RTMP
ffmpeg -i "srt://0.0.0.0:9000?mode=listener" \
-c copy \
-f flv rtmp://destination/live/stream
RTMP to Multiple Destinations
ffmpeg -i rtmp://source/live/stream \
-c copy -f flv rtmp://youtube/live/key1 \
-c copy -f flv rtmp://twitch/app/key2 \
-c copy -f flv rtmp://facebook/rtmp/key3
Production Patterns
nginx-rtmp Integration
# nginx.conf
rtmp {
server {
listen 1935;
application live {
live on;
exec_push ffmpeg -i rtmp://localhost/live/$name
-c:v libx264 -preset veryfast -b:v 3000k
-c:a aac -b:a 128k
-hls_time 4
-hls_list_size 10
-hls_flags delete_segments
/var/www/html/hls/$name.m3u8;
}
}
}
Docker Streaming Setup
# docker-compose.yml
version: '3.8'
services:
rtmp:
image: tiangolo/nginx-rtmp
ports:
- "1935:1935"
- "8080:8080"
volumes:
- ./nginx.conf:/etc/nginx/nginx.conf
transcoder:
image: jrottenberg/ffmpeg:7.1-ubuntu2404
depends_on:
- rtmp
command: >
-i rtmp://rtmp:1935/live/stream
-c:v libx264 -preset veryfast
-c:a aac
-hls_time 4
-f hls /output/stream.m3u8
volumes:
- ./output:/output
Monitoring & Logging
# Enable detailed logging
ffmpeg -report -i input.mp4 -f flv rtmp://server/live/stream
# Real-time stats
ffmpeg -re -i input.mp4 \
-progress pipe:1 \
-c:v libx264 -c:a aac \
-f flv rtmp://server/live/stream
# Write stats to file
ffmpeg -re -i input.mp4 \
-progress stats.txt \
-c:v libx264 -c:a aac \
-f flv rtmp://server/live/stream
Troubleshooting
Common Issues
"Connection refused"
# Check server is reachable
nc -zv server 1935
# Test with simple stream
ffmpeg -re -f lavfi -i testsrc=size=1280x720:rate=30 \
-f flv rtmp://server/live/test
Buffer underrun/overrun
# Increase buffer size
ffmpeg -re -i input.mp4 \
-b:v 3000k -maxrate 3000k -bufsize 6000k \
-f flv rtmp://server/live/stream
High latency
# Reduce latency settings
ffmpeg -re -i input.mp4 \
-c:v libx264 -preset ultrafast -tune zerolatency \
-g 30 -keyint_min 30 \
-f flv rtmp://server/live/stream
Packet drops
# Increase receive buffer (SRT)
ffmpeg -i "srt://server:9000?rcvbuf=8192000" -c copy output.mp4
# Increase buffer for UDP
ffmpeg -buffer_size 8192000 -i udp://239.0.0.1:1234 -c copy output.mp4
Best Practices
- Use appropriate keyframe interval - 2 seconds for live streaming
- Match bitrate to bandwidth - Leave 20% headroom
- Use CBR for streaming - Consistent bandwidth
- Enable faststart for VOD -
-movflags +faststart - Monitor stream health - Check for drops/errors
- Use hardware encoding - NVENC/QSV for GPU offload
- Implement ABR - Multiple quality levels
- Secure your streams - Use RTMPS/encryption
- Test failover - Backup ingest points
- Log everything - Use
-reportfor debugging
This guide covers FFmpeg streaming fundamentals. For hardware acceleration, see the hardware acceleration skill.
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レビュー
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