RTP (Real-time Transport Protocol)
RTP, or Real-time Transport Protocol, is a network protocol that carries timestamped and sequenced audio or video packets so a receiver can reconstruct real-time media in the correct order. In internet telephony, RTP commonly carries the conversation audio after call signaling has established a session.
How RTP works
The sender encodes a short segment of audio, places it in an RTP packet, and adds metadata such as a sequence position, timestamp, and media type. The receiver uses that information to order packets and schedule playback. A jitter buffer can briefly hold arriving packets to smooth normal timing variation.
RTP is designed for timely delivery rather than guaranteed delivery. It is commonly carried over UDP, where a late packet may be discarded instead of retransmitted after its playback moment has passed. Packet loss can create gaps or distortion, while excessive jitter can force the receiver to add delay or drop late media.
A companion control protocol can report reception quality and timing information. Those reports help systems observe loss, delay, and synchronization, but they do not repair the underlying network. RTP itself also does not establish the call, select the participant, or guarantee bandwidth. SIP and WebRTC signaling handle session setup, while codec negotiation determines how the audio inside each packet is represented.
Plain RTP does not encrypt its media. Deployments that need media confidentiality and packet authentication use SRTP or another protected transport arrangement.
Why RTP matters for AI phone calls
An AI phone agent depends on a continuous stream of usable audio in both directions. Missing, late, or reordered RTP packets can make the caller's words harder to recognize and the agent's speech harder to understand. Media problems can resemble conversational problems: a clipped first syllable may look like poor turn-taking, and a burst of loss may look like an incorrect transcription.
Troubleshooting should therefore examine the media path alongside the agent. Packet timing, loss, jitter-buffer behavior, codec changes, and gateway transcoding can explain defects that prompt edits cannot fix. Testing from representative networks is especially useful because a clean local call does not prove that every carrier route will behave the same way.