Real-time media and framing
The current Peer Lab transports actual PCM bytes between two independent localhost clients over WebSocket. It does not generate or receive SIP RTP and has no verified PSTN audio.
Binary packet (656 bytes)
| Byte offset | Size | Meaning |
|---|---|---|
| 0 | 4 | ASCII magic DT01 |
| 4 | 4 | Big-endian unsigned 32-bit per-sender sequence |
| 8 | 8 | Big-endian unsigned 64-bit timestamp in milliseconds |
| 16 | 640 | PCM16 little-endian, signed, mono, 16 kHz = 20 ms |
The server checks frame size, magic, strictly increasing sequence and per-role rate. A media frame is not a transcript and the timestamp comes from the sending client, not a trusted wall clock. PCM streams must be paced by the client. The local session duration is capped by the operator.
Text and control messages
The WebSocket also carries say (unverified agent assertion), dtmf (a symbolic message, not generated telephone tones), audio.end, audio.interrupt, hangup, and ping. Receiving an interruption requests model/TTS cancellation; a real PSTN implementation still needs measured barge-in handling, jitter buffering, interruption race control and end-to-end audio quality tests.
For full wire examples, see Peer Lab and the downloadable peer OpenAPI. For the experimental audio processing pipeline, see Audio-first Voice.
Frame example
import struct
# Exactly one 20-ms frame: 16 kHz, mono PCM16 little-endian.
samples = bytes(640) # silence; not a real voice recording
packet = struct.pack('!4sIQ', b'DT01', 1, 0) + samples
assert len(packet) == 656
The DT01 prefix is local Peer Lab framing. It is not an RTP, SIP or PSTN packet.