Codecs

rtpbridge supports three audio codecs with automatic transcoding between them.

Supported Codecs

Codec	PT	Sample Rate	RTP Clock Rate	Crate
PCMU (G.711 mu-law)	0	8 kHz	8000	xlaw (pure Rust)
G.722	9	16 kHz	8000*	ezk-g722 (pure Rust)
Opus	111 (dynamic)	48 kHz	48000	opus (libopus FFI)

* G.722 uses 8000 in SDP per RFC 3551 even though the actual audio sample rate is 16 kHz.

Transcoding

When endpoints in a session use different codecs, rtpbridge automatically transcodes:

Source codec → Decode to PCM i16 → Resample → Encode to destination codec

The resample step handles rate conversion between 8 kHz, 16 kHz, and 48 kHz using linear interpolation. Opus requires exact frame sizes (960 samples at 48 kHz for 20ms ptime), so the pipeline pads or truncates after resampling.

Passthrough

If source and destination use the same codec, packets are forwarded directly without transcoding.

DTMF Bypass

DTMF (telephone-event) packets are never transcoded. They bypass the transcode pipeline entirely and are forwarded as-is, with payload type remapping if the two endpoints negotiated different dynamic PTs.

Ptime

All codecs use 20ms ptime:

Codec	Samples per 20ms
PCMU	160
G.722	320
Opus	960

Telephone-Event

For WebRTC endpoints, RFC 4733 telephone-event uses PT 101. For plain RTP/SRTP endpoints, generated SDP includes telephone-event by default and answers retain the offered telephone-event payload type and clock rate. The a=fmtp:101 0-16 line supports digits 0-9, *, #, A-D, and flash. File, tone, bridge, and WebSocket audio endpoints do not negotiate telephone-event.