RTP (Real-time Transport Protocol) is a standard protocol for delivering audio and video streams across IP networks. It’s used in everything from VoIP calls and video conferencing to web-based push-to-talk, IP TV, and WebRTC services.

It is defined in RFC 1889.by the IETF Audio Video Transport working group, RTP is a key component of modern network protocols for real-time communication. It’s often paired with RTCP (RTP Control Protocol) and SIP (Session Initiation Protocol) to establish, deliver, and monitor live media sessions.

The RTP protocol is used in conjunction with the RTP Control Protocol (RTCP) and Session Initiation Protocol (SIP). While RTP carries the media streams (e.g., audio and video), RTCP packets are used to monitor transmission statistics and quality of service (QoS) data such as jitter, packet loss (identified using a sequence number), and round-trip time. RTCP also aids in the synchronization of multiple streams. RTP sessions originate and are received on even port numbers and the associated RTCP communication uses the next higher odd port number. Any port number can be used for RTP traffic although, in general terms, ports used can range between 1024 and 65535. RTP is one of the foundations of VoIP and it is used in conjunction with SIP which assists in setting up the connections across the network.

RTP was designed for time-sensitive delivery of media. Its key features include:

• Timestamping and sequence numbers for playback control
• Support for multicast and unicast transmission
• Packet loss detection using sequence tracking
• Adaptability to other transport protocols, especially UDP
• Application-layer design for broad compatibility

These features make RTP the standard protocol for transmitting media across diverse networks.

While RTP carries the actual media (e.g., audio or video), RTCP provides status updates and performance stats:

• Jitter monitoring
• Packet loss detection
• Round-trip time calculation
• Stream synchronization

Together, RTP and RTCP enable efficient and stable media delivery, especially for voice over internet protocol and conferencing tools.

• RTP runs primarily over UDP, not TCP, to minimize delay
• Each RTP session uses an even-numbered port, with RTCP using the next odd-numbered port
• Common port ranges: 1024–65535
• The RTP payload can be audio, video, or text

TCP and Stream Control Transmission Protocol (SCTP) are supported but rarely used due to higher latency.

RTP includes timestamping to support synchronization of different streams. For example, if audio and video are sent separately, the RTP time markers ensure they stay in sync during playback.

RTCP helps fine-tune this by exchanging control information without impacting the RTP stream itself. The result is smoother, uninterrupted playback, even in unstable networks.

Unlike TCP, RTP does not guarantee delivery. Instead, it prioritizes speed over accuracy, compensating for lost or delayed packets to maintain a fluid experience.

• Jitter buffers smooth out timing gaps
• Sequence numbers help identify missing packets
• Loss of a few milliseconds of media is acceptable in most real-time apps

This trade-off is what makes RTP suitable for VoIP, WebRTC, and live streaming.

VoIP services rely on RTP to deliver voice reliably. Even if a few packets are lost, error compensation algorithms can make up for it, keeping voice quality high.

When paired with SIP and Session Description Protocol (SDP), RTP becomes the engine behind most voice over internet protocol services.

RTP by default is not encrypted. SRTP (Secure Real-time Transport Protocol) adds encryption and message authentication. It’s used in privacy-focused apps like 3CX, Zoom, and Teams.

SRTP:

• Encrypts the RTP payload (e.g., audio data)
• Adds integrity checks
• Works alongside TLS for full end-to-end security

RTSP (Real-Time Streaming Protocol) is often confused with RTP. RTSP controls playback (pause, seek, etc.), but does not transmit media. RTP is what actually streams the media data.

In a typical setup:

• RTSP sets up the session
• SDP defines the codecs and media types
• RTP carries the data
• RTCP monitors the connection

RTP is codec-agnostic. It can transport:

• G.711, G.729 for audio
• H.264, VP8 for video
• Custom codecs for niche use cases

The RTP payload type is negotiated at the start of a session using SDP.

You’ll find RTP in most real-time and low-latency applications:

• VoIP platforms like 3CX, Zoom, Cisco, and Asterisk
• Video conferencing tools
• IP-based radio and TV systems
• Security camera feeds
• WebRTC-based browsers and apps

RTP is codec-agnostic. It can transport:

• G.711, G.729 for audio
• H.264, VP8 for video
• Custom codecs for niche use cases

The RTP payload type is negotiated at the start of a session using SDP.