Supported Protocols

Overview

bilbycast-edge is a pure-Rust media gateway supporting multiple transport protocols for professional broadcast and streaming workflows. All protocol implementations are native Rust with no C library dependencies.

Input Protocols

RTP (SMPTE ST 2022-2)

• Direction: Input
• Transport: UDP unicast or multicast, IPv4 and IPv6
• Payload: MPEG-2 Transport Stream in RTP per SMPTE ST 2022-2
• Features:
  • SMPTE 2022-1 FEC decode (column + row parity)
  • IGMP/MLD multicast group join with interface selection
  • Source IP allow-list filtering (RP 2129 C5)
  • RTP payload type filtering (RP 2129 U4)
  • Per-flow ingress rate limiting (RP 2129 C7)
  • DSCP/QoS marking on egress (RP 2129 C10)
  • VSF TR-07 mode: auto-detects JPEG XS streams

UDP (Raw MPEG-TS)

• Direction: Input
• Transport: UDP unicast or multicast, IPv4 and IPv6
• Payload: Raw MPEG-TS datagrams (no RTP headers required)
• Features:
  • Accepts any UDP datagram (no RTP v2 header filtering)
  • IGMP/MLD multicast group join with interface selection
  • Compatible with OBS, ffmpeg udp:// output, srt-live-transmit, VLC

SRT (Secure Reliable Transport)

• Direction: Input
• Transport: UDP with ARQ retransmission
• Modes: Caller, Listener, Rendezvous
• Features:
  • AES-128/192/256 encryption (AES-CTR default, AES-GCM authenticated encryption selectable via crypto_mode)
  • Stream ID access control (stream_id, max 512 chars per SRT spec; supports #!::r=name,m=mode,u=user structured format)
  • FEC (Forward Error Correction) via packet_filter — XOR-based row/column parity, staircase layout, ARQ integration modes (always/onreq/never), wire-compatible with libsrt v1.5.5
  • Configurable latency buffer (symmetric or asymmetric receiver/sender latency)
  • Retransmission bandwidth capping (Token Bucket shaper via max_rexmit_bw)
  • SMPTE 2022-7 hitless redundancy merge (dual-leg input)
  • Automatic reconnection

Output Protocols

RTP (SMPTE ST 2022-2)

• Direction: Output
• Transport: UDP unicast or multicast
• Payload: RTP-wrapped MPEG-TS packets with RTP headers
• Features:
  • SMPTE 2022-1 FEC encode
  • DSCP/QoS marking (default: 46 Expedited Forwarding)
  • Multicast with interface selection

UDP (Raw MPEG-TS)

• Direction: Output
• Transport: UDP unicast or multicast
• Payload: Raw MPEG-TS datagrams (7×188 = 1316 bytes, no RTP headers)
• Features:
  • Strips RTP headers when input is RTP-wrapped
  • TS sync detection and packet alignment
  • DSCP/QoS marking
  • Compatible with ffplay, VLC, multicast distribution

SRT (Secure Reliable Transport)

• Direction: Output
• Modes: Caller, Listener, Rendezvous
• Features:
  • AES encryption (AES-CTR or AES-GCM via crypto_mode)
  • Stream ID access control (stream_id, max 512 chars; callers send during handshake, listeners filter)
  • FEC (Forward Error Correction) via packet_filter — same capabilities as SRT input
  • Asymmetric latency support (independent receiver/sender latency)
  • Retransmission bandwidth capping (Token Bucket shaper)
  • SMPTE 2022-7 hitless redundancy duplication (dual-leg output)
  • Non-blocking mpsc bridge prevents TCP backpressure from affecting other outputs
  • Automatic reconnection

RTMP/RTMPS

• Direction: Output only (publish)
• Transport: TCP (RTMP) or TLS over TCP (RTMPS)
• Use case: Delivering to Twitch, YouTube Live, Facebook Live
• Features:
  • Pure Rust RTMP protocol implementation (handshake, chunking, AMF0)
  • Demuxes H.264 and AAC from MPEG-2 TS, muxes into FLV
  • Automatic AVC sequence header (SPS/PPS) and AAC config generation
  • Reconnection with configurable delay and max attempts
  • Non-blocking: uses mpsc bridge pattern, never blocks other outputs
• Limitations:
  • Output only. RTMP input (ingest from OBS etc.) is not implemented.
  • Only H.264 video and AAC audio. HEVC/VP9 not supported via RTMP.
  • RTMPS (TLS) uses the tls feature (enabled by default).

HLS Ingest

• Direction: Output only
• Transport: HTTP PUT/POST over TCP
• Use case: YouTube HLS ingest (supports HEVC/HDR content)
• Features:
  • Segments MPEG-2 TS data into time-bounded chunks
  • Generates rolling M3U8 playlist
  • Configurable segment duration (0.5-10 seconds)
  • Optional Bearer token authentication
  • Async HTTP upload, non-blocking to other outputs
• Limitations:
  • Output only. Segment-based transport inherently adds 1-4 seconds of latency.
  • Uses a minimal built-in HTTP client (not a full HTTP/2 client).

RTSP

• Direction: Input only
• Transport: TCP (interleaved) or UDP
• Client library: retina (pure Rust)
• Features:
  • Pull H.264 or H.265/HEVC video and AAC audio from IP cameras and media servers
  • Digest and Basic authentication support
  • TCP interleaved (default, works through firewalls) or UDP transport
  • Automatic reconnection with configurable delay
  • Received media is muxed into MPEG-TS with proper PAT/PMT program tables
  • Audio-only streams supported (PAT/PMT emitted even without video)
• Limitations:
  • Input only (no RTSP server mode)
  • AAC audio is passed through as ADTS in MPEG-TS

WebRTC (WHIP/WHEP)

• Direction: Input and Output
• Transport: UDP (ICE-lite/DTLS/SRTP) via str0m pure-Rust WebRTC stack
• Status: Fully implemented. The webrtc feature is enabled by default.
• Four modes:
  • WHIP input (server): Accept contributions from OBS, browsers — endpoint at /api/v1/flows/{id}/whip
  • WHIP output (client): Push media to external WHIP endpoints (CDN, cloud)
  • WHEP output (server): Serve browser viewers — endpoint at /api/v1/flows/{id}/whep
  • WHEP input (client): Pull media from external WHEP servers
• Video: H.264 only (RFC 6184 RTP packetization/depacketization)
• Audio: Opus passthrough. Opus flows natively on WebRTC paths and gets muxed into MPEG-TS for SRT/RTP/UDP outputs. AAC sources going to WebRTC output automatically fall back to video-only (no C-library transcoding).
• Interoperability: Compatible with OBS, browsers, Cloudflare, LiveKit, and other standard WHIP/WHEP implementations.
• Security: Bearer token authentication on WHIP/WHEP endpoints, DTLS/SRTP encryption, ICE-lite for server modes.
• NAT traversal: Configurable public_ip and optional stun_server for ICE candidate advertisement.

Monitoring and Analysis

TR-101290 Transport Stream Analysis

• Priority 1: Sync byte, continuity counter, PAT/PMT presence
• Priority 2: TEI, PCR discontinuity, PCR accuracy
• Runs as independent broadcast subscriber (zero jitter impact)

VSF TR-07 Awareness

• Auto-detects JPEG XS (stream type 0x61) in PMT
• Reports TR-07 compliance status via API and dashboard
• Enable with tr07_mode: true in RTP input config

SMPTE Trust Boundary Metrics (RP 2129)

• Inter-arrival time (IAT): min/max/avg per reporting window
• Packet delay variation (PDV/jitter): RFC 3550 exponential moving average
• Filtered PDV (CMAX): peak-to-peak filtered metric
• Source IP filtering (C5), payload type filtering (U4), rate limiting (C7)
• DSCP/QoS marking (C10)
• Flow health derivation (M6): Healthy/Warning/Error/Critical

Cargo Features

Feature	Description	Default
tls	Enable RTMPS (RTMP over TLS) via rustls/tokio-rustls	Yes
webrtc	Enable WebRTC WHIP/WHEP input and output via str0m	Yes

All features are enabled by default. A plain cargo build --release includes everything.

Configuration Examples

See the config_examples/ directory for JSON configuration examples for each protocol type.

Architecture Notes

• All outputs subscribe to a shared broadcast channel independently
• Slow outputs receive Lagged errors and drop packets — they never block the input or other outputs
• TCP-based outputs (RTMP, HLS) use an mpsc bridge pattern to keep TCP operations off the broadcast receive path
• All monitoring (TR-101290, IAT/PDV, TR-07) runs on independent analyzer tasks with zero impact on the data plane