Changelog#

All notable changes to go-rtmp are documented here.

Unreleased#

SRT Encryption — Full AES-CTR encryption with passphrase-based key exchange (KMREQ/KMRSP), supporting AES-128/192/256 with PBKDF2 key derivation
SRT Key Rotation — Hitless even/odd key rekeying via post-handshake KMREQ control packets for long-running encrypted streams
SRT Reconnection Fix — Second SRT connection with same stream key no longer fails after first disconnects

SRT (Secure Reliable Transport) Ingest: Accept SRT streams over UDP alongside RTMP
- SRT v5 handshake with SYN cookie exchange and extension negotiation
- Stream ID parsing: simple, prefixed, and structured formats
- TSBPD jitter buffer with configurable latency
- ACK/NAK reliability with RTT measurement and retransmission
- Optional AES encryption (128/192/256-bit) with PBKDF2 key derivation
MPEG-TS Demuxer: Full transport stream parser with PAT/PMT table decoding, PES reassembly
Codec Converters: H.264/H.265 Annex B→AVCC and AAC ADTS→raw converters for SRT-to-RTMP bridge
SRT-to-RTMP Bridge: End-to-end pipeline converting SRT data packets through TS demuxing and codec conversion
Codec-Aware Recording: Automatic container selection — FLV for H.264, MP4 for H.265/HEVC
FLV onMetaData Script Tag: FLV recordings include onMetaData with video dimensions, codec IDs, audio properties, and patched duration/filesize on close
- MP4 recorder streams to disk in real-time (zero memory buffering)
- Lazy recorder initialization for correct codec detection
Ingress Abstraction: Protocol-agnostic publish lifecycle manager shared by RTMP and SRT
Comprehensive E2E Test Suite: 25+ tests in e2e-tests/ covering all major features
SRT CLI Flags: -srt-listen, -srt-latency, -srt-passphrase, -srt-pbkeylen
SRT Metrics: 6 new expvar counters for SRT connections, bytes, packets, retransmits, drops
Comprehensive package documentation and developer guide

internal/srt/ — Full SRT protocol implementation (packet, circular, crypto, handshake, conn)
internal/ts/ — MPEG-TS demuxer
internal/codec/ — Video/audio codec converters (H.264, H.265, AAC)
internal/ingress/ — Publish lifecycle abstraction

Enhanced RTMP (E-RTMP v2): H.265/HEVC, AV1, VP9 codec support via FourCC signaling
- Compatible with FFmpeg 6.1+, OBS 29.1+, SRS 6.0+
- Automatic codec detection — no configuration needed
- connect command negotiation with fourCcList echo
- Sequence header caching for all enhanced codecs (late-join support)
Enhanced audio signaling: Opus, FLAC, AC-3, E-AC-3 via E-RTMP FourCC
27 new unit tests for enhanced video/audio parsing
E2E test scripts: scripts/test-enhanced-rtmp.sh and .ps1

RTMPS (TLS) support — encrypted RTMP connections via TLS termination at the transport layer
- New CLI flags: -tls-listen, -tls-cert, -tls-key
- Dual-listener architecture: plain RTMP and RTMPS simultaneously
- rtmps:// URL support in the Go client and relay destinations
- Minimum TLS 1.2 enforced; TLS startup failure is fatal (no silent fallback)
- 4 TLS integration tests with self-signed certificate helper
Cross-platform E2E testing scripts — comprehensive test suite in scripts/
- 12 scripts (6 Bash + 6 PowerShell pairs) for Linux/macOS/Windows
- 7 E2E test cases: RTMP, RTMPS, HLS hooks, authentication (allowed + rejected), combined TLS + auth
- Helper scripts: dependency checker, TLS cert generator, parameterized server launcher, HLS hook
Cross-platform build scripts — scripts/build.sh and scripts/build.ps1 for local compilation
Hugo documentation site — full docs with GitHub Pages deployment, Hugo-book theme

Shell hook Windows compatibility (powershell.exe detection instead of hardcoded /bin/bash)
Docs workflow Hugo version bump for theme compatibility
GitHub Pages deployment configuration

Expvar metrics — live counters exposed via HTTP /debug/vars endpoint (-metrics-addr flag)
Disconnect handlers — proper cleanup of publisher/subscriber registrations on connection close
TCP deadline enforcement — read 90s / write 30s deadlines for zombie detection, reset on each I/O operation
Lifecycle hook events — new events: connection_close, publish_stop, play_stop, subscriber_count

Performance optimizations:
- AMF0 decode: reduced allocations in object/array parsing
- Chunk writer: buffer reuse to avoid repeated allocation
- RPC: lazy initialization of command dispatcher
Dead code removal: removed unused bufpool package, ErrForbidden sentinel, and unused Session type

Token-based authentication with 4 backends:
- token — static stream key/token pairs via CLI flags
- file — JSON file with stream key/token mappings
- callback — webhook URL for external auth validation
- none — open access (default)
URL query parameter parsing — tokens extracted from ?token=value in stream URLs
EventAuthFailed hook event — fires when authentication fails

First feature-complete release of go-rtmp.

Full RTMP v3 protocol implementation:
- Handshake (C0/C1/C2 ↔ S0/S1/S2) with FSM and timeout enforcement
- Chunk streaming with FMT 0-3, extended timestamps, configurable chunk size
- AMF0 codec (Number, Boolean, String, Object, Null, Strict Array)
- Command parsing (connect, createStream, releaseStream, FCPublish, publish, play)
- Control messages (Set Chunk Size, Abort, Acknowledgement, Window Ack Size, Set Peer Bandwidth)
Live relay — transparent pub/sub forwarding to unlimited subscribers
Late-join support — H.264 SPS/PPS and AAC AudioSpecificConfig caching
FLV recording — automatic recording with timestamped filenames
Multi-destination relay — forward streams to external RTMP servers (-relay-to)
Event hooks:
- Webhook — HTTP POST to configurable URLs
- Shell — execute scripts on lifecycle events
- Stdio — structured output (JSON or env format) to stdout
Structured logging — log/slog with configurable level and context fields
Integration tests — end-to-end pub/sub validation through the full protocol stack
Golden binary vectors — exact wire-format test data for handshake, chunks, AMF0, and control messages