Simple definition
RTP (Real-time Transport Protocol) normally carries the real-time audio or video streams of a VoIP communication.
Technical definition
RTP encapsulates media samples with information such as sequence numbers, timestamps, and a source identifier. It commonly runs over UDP and uses dynamic ports negotiated during session setup; RTCP can provide additional quality and synchronization information.
How it works / role
After SIP/SDP negotiation, each endpoint sends RTP packets to the address and port advertised by the other side. Sequence numbers and timestamps help the receiver reorder packets and feed its jitter buffer.
What is it used for?
Carry voice, video, or other real-time media between endpoints, SBCs, PBXs, and gateways.
Practical example
During a call, SIP establishes the session and two UDP RTP streams can then carry voice in each direction between the phone and the remote system.
Common issues
- The RTP port range is blocked by the firewall
- NAT advertises the wrong address or causes one-way audio
- Packet loss, latency, or jitter is too high
- Codec or media parameters are incompatible
Key takeaway: When troubleshooting no-audio or one-way-audio calls, check SIP signaling and the negotiated RTP path separately.