RTP, or Real-time Transport Protocol, carries real-time media such as audio and video in packets. It adds information that helps a receiver identify, order, time, and monitor media, but it does not guarantee that packets arrive, arrive on time, or arrive in order. RTP is a transport protocol—not a codec or a complete streaming service.
What does RTP stand for?
RTP stands for Real-time Transport Protocol. The IETF’s RFC 3550 describes it as providing end-to-end transport functions for applications sending real-time data, including audio, video, and simulation data, over multicast or unicast network services. Despite the word “streaming,” RTP is a protocol used within media and communications systems, not a consumer streaming platform.
RTP can carry media produced with different codecs and payload formats. The protocol provides a common packet structure and transport information; separate profiles and payload-format specifications define how particular media payloads are represented and interpreted.
How does RTP streaming work?
At a high level, a sending application encodes media, divides it into chunks, and places each chunk in an RTP packet. The receiver uses the packet’s metadata and the applicable payload-format rules to put the media in context, decode it, and present it. The exact packetization, buffering, recovery, and playback behavior depends on the application and the RTP profile or extensions it uses.
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- The sender encodes and packetizes media. Audio or video is encoded by an application and split into manageable pieces.
- RTP adds a header. The header includes a payload type, a sequence number, a timestamp, and source-identifying information. These fields tell the receiver how to interpret and place the packet relative to other packets.
- Packets travel over a network. RTP is commonly carried over UDP, but it is designed to be independent of the underlying transport and network layers; it is not UDP-only.
- The receiver interprets and presents the media. Using the relevant profile and payload-format definition, the receiver can detect sequence gaps, place packets in order as appropriate, use timestamps for playback timing and synchronization, and decode the payload.
What the header information is for
- Payload type: identifies the payload format so the receiver can interpret the media. RTP itself does not define every codec.
- Sequence number: helps the receiver detect packet ordering and gaps. It provides information; it cannot make a network deliver missing packets.
- Timestamp: relates the packet to the media clock, helping the receiver time playback and synchronize media streams.
- Source information: identifies the contributing source in the RTP session.
What is the difference between RTP and RTCP?
RTP carries the media payload. RTCP, the companion control protocol, carries control information used for delivery monitoring and participant information. RTCP is not the audio or video payload itself. Together, RTP and RTCP let an application transport media while also exchanging information about the session and observed delivery.
What RTP does—and does not—guarantee
RTP provides metadata useful for handling real-time media; it does not guarantee delivery, in-order arrival, timely arrival, or quality of service. It also does not reserve network resources. Packets can be lost, delayed, or arrive out of sequence. Sequence numbers and timestamps help a receiver detect and manage these conditions, but the application, network, and any additional mechanisms determine what happens next.
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For that reason, it is inaccurate to say RTP by itself prevents buffering or packet loss. Buffering, retransmission, error concealment, congestion handling, and other recovery behavior depend on the application, profile, and extensions in use.
How RTP fits into WebRTC
WebRTC uses RTP as its required media transport protocol and requires RTCP as well. RFC 8834, published in January 2021, describes RTP use in WebRTC, including the profile’s RTCP feedback and secure RTP provisions. RFC 8835, also published in January 2021, covers WebRTC transport protocols and their interaction with intermediaries such as firewalls, relays, and NAT boxes.
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RTP is therefore a core media layer in WebRTC, not the whole WebRTC system. Signaling, security, and network traversal involve additional parts of the broader stack.
RTP is not RTSP
RTP and RTSP are different protocol names used in media contexts. Do not treat them as interchangeable: RTP’s role is transporting real-time media packets. The details of RTSP are outside this explanation.
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Quick Recap
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