sip/audio/rtp library

Minimal RFC 3550 RTP packet builder + parser — pure Dart.

Trimmed port of KellyKinyama/dart-rtp-packet (lib/src/rtp.dart), keeping only what an audio-only G.711 sender/receiver needs:

• fixed 12-byte header
• single-payload packet (no fragmentation; G.711 frames at 20ms are ~160 bytes, well under any MTU)
• sequence + timestamp + SSRC bookkeeping in RtpState

SRTP, RTCP, header extensions and FU-A fragmentation are intentionally omitted — they don't apply to plain SIP/G.711 calls.

Classes

RtpPacket

Decoded inbound RTP packet.

RtpState

Mutable per-stream state. One instance per outbound RTP stream.

Constants

rtpHeaderSize → const int

rtpVersion → const int

Functions

makeRtpPacket(RtpState state, Uint8List payload, int rtpTimestamp, {bool marker = false}) → Uint8List

Builds an RTP packet (RFC 3550 §5.1). Returns the wire bytes ready for RawDatagramSocket.send. Increments state.seq by one.

parseRtp(Uint8List data) → RtpPacket?

Parses an inbound RTP datagram. Returns null if the buffer is too short or the version isn't 2. Skips past CSRC list and any header extension.