RTP
Real-time Transport ProtocolRTP carries media that is worth less the later it arrives. It rides on UDP and adds only what live audio and video need: a timestamp so playback can be paced, a sequence number so loss and reordering are visible, and a payload type saying which codec produced the bytes.
RTP adds timing and sequence to a UDP stream — enough to play media smoothly, and deliberately not enough to retransmit it.
How it works
The sequence number shows what went missing and what arrived out of order; the timestamp says when it should be played, not when it turned up. A jitter buffer uses both to turn a bumpy network into steady playback.
What RTP does, step by step
SIP or WebRTC negotiates the addresses, ports and codecs in SDP before any media flows.
The encoder’s output is split into packets, each stamped with a sequence number and a media timestamp.
Packets go out at the media’s own pace — a late packet is worse than a missing one, so nothing is retransmitted.
The receiver holds a small jitter buffer, reorders by sequence, and conceals whatever never arrived.
In cybersecurity
Plain RTP is an unencrypted, unauthenticated media stream — anyone on the path can record the call, or inject audio into it.
The secure profile encrypts and authenticates each packet; WebRTC refuses to send anything else.
Captured RTP replays as audio with ordinary tools — no cryptanalysis involved.
Without authentication, a forged packet with a plausible sequence number is simply played.
SRTP is only as good as DTLS-SRTP or MIKEY delivering the keys; SDES in clear SDP is not enough.
Media uses a wide dynamic UDP range — open it narrowly and pin it to the signalling.
Open relays and turn servers left unauthenticated get used to bounce traffic at someone else.