When you make a VoIP call or join a video meeting, something has to carry your voice and video across the internet in real time. That “something” is typically RTP. In this glossary entry, we’ll explain what RTP is, how it works, why it runs over UDP, and why it’s essential for technologies like VoIP, SIP trunking, and video conferencing.
What Is RTP (Real-Time Transport Protocol)?
RTP (Real-Time Transport Protocol) is a network protocol used to deliver real-time audio and video over IP networks. It plays a critical role in technologies like VoIP calls, video conferencing, and live media streaming, where data must arrive quickly to maintain natural communication. RTP prioritises low latency and smooth playback over perfect reliability, making it ideal for time-sensitive media transmission.
How Does RTP Work?
RTP works by taking live audio or video and breaking it into small packets that can be transmitted quickly across an IP network. The process is optimised for speed so that conversations and video streams feel natural and uninterrupted.
Here’s a simple view of how RTP operates in real time:
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Media is captured — A microphone captures voice or a camera captures video. The media is converted into digital data.
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RTP packetizes the media — The digital audio or video stream is divided into small data packets suitable for network transmission.
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Sequence numbers are added — Each RTP packet includes a sequence number so the receiving device can detect missing packets and reorder them correctly.
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Timestamps are added — RTP adds timestamps to help synchronise audio and video streams and ensure proper playback timing.
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Packets are sent over UDP — RTP typically runs over UDP to minimise delay. UDP does not retransmit lost packets, which helps maintain low latency.
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The receiver reorders and plays media — The receiving system uses the sequence numbers and timestamps to reorder packets, smooth out jitter, and play the media in real time.
Key Features of RTP
RTP is specifically designed to handle real-time media transmission. Here are its core features in simple terms:
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Packetisation of real-time media — RTP divides live audio and video into small packets so they can be transmitted efficiently across IP networks.
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Sequence numbers (packet ordering) — Each packet includes a sequence number, allowing the receiver to detect lost packets and restore the correct order during playback.
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Timestamps (synchronisation) — RTP adds timestamps to ensure audio and video streams stay synchronised and are played at the correct speed.
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Designed for low latency — RTP prioritises speed and smooth delivery, making it suitable for live communication like VoIP and video conferencing.
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Does not guarantee delivery — RTP does not retransmit lost packets. Instead, it tolerates minor packet loss to avoid delays that would disrupt real-time conversations.
RTP vs UDP — What’s the Difference?
RTP and UDP are closely related, but they are not the same thing. UDP is a transport protocol that moves data from one device to another without guaranteeing delivery. RTP runs on top of UDP and adds media-specific features needed for real-time audio and video transmission.
Here’s a simple comparison:
| Feature | RTP | UDP |
|---|---|---|
| Purpose | Media transport (audio/video) | General data transport |
| Reliability | No guarantee | No guarantee |
| Adds timestamps | Yes | No |
| Adds sequence numbers | Yes | No |
In short, UDP handles the delivery, while RTP structures and manages real-time media on top of UDP. UDP keeps things fast by avoiding retransmissions, and RTP adds the intelligence needed to keep voice and video synchronised and in the correct order.
RTP vs RTCP vs SRTP
RTP is often used alongside other related protocols. Understanding how they differ helps clarify how real-time communication systems work.
Here’s a simple comparison:
| Protocol | Purpose |
|---|---|
| RTP | Carries audio and video media |
| RTCP | Monitors quality and provides synchronisation feedback |
| SRTP | Encrypts and secures RTP traffic |
In a typical VoIP call, RTP carries the actual voice data between devices. At the same time, RTCP (RTP Control Protocol) sends quality reports, such as packet loss and jitter statistics, helping systems monitor performance and adjust if needed. When security is required, SRTP (Secure RTP) encrypts the media stream to protect conversations from interception. Together, these protocols ensure real-time media is delivered quickly, monitored for quality, and secured when necessary.
Why Is RTP Important for VoIP and SIP Trunking?
RTP is essential to VoIP and SIP trunking because it carries the actual voice data during a call. While SIP is responsible for signalling—setting up, managing, and ending the call—RTP is the protocol that transports the live audio between participants. Without RTP, a call could be established, but no sound would be transmitted.
In VoIP environments, RTP enables real-time communication by delivering voice packets quickly and in the correct order. It works alongside SIP to separate signalling from media, which improves flexibility and scalability in business phone systems.
In SIP trunking setups, RTP flows between the PBX (or cloud phone system) and the provider’s network, carrying inbound and outbound voice traffic. Performance factors like latency, jitter, and packet loss directly affect RTP streams, which is why network quality and proper configuration are critical for clear call audio.
