RTP: What is It and How Does It Work?

Standards Track Page 7 RFC 3550 RTP July 2003 Mixers and translators may be designed for a variety of purposes. The RTP header includes a means for mixers to identify the sources that contributed to a mixed packet so that correct talker indication can be provided at the receivers. The sequence number can also be used by the receiver to estimate how many packets are being lost. In these examples, RTP is carried on top of IP and UDP, and follows the conventions established by the profile for audio and video specified in the companion RFC 3551. A profile for audio and video data may be found in the companion RFC 3551 .

RTP Payload Types

In order to track loops of the participant’s own data packets, the implementation MUST also keep a separate list of source transport addresses (not identifiers) that have been found to be conflicting. Note that if two sources on the same host are transmitting with the same source identifier at the time a receiver begins operation, it would be possible that the first RTP packet received came from one of the sources while the first RTCP packet received came from the other. This problem can be avoided by keeping the source transport address fixed across restarts, but in any case will be resolved after a timeout at the receivers. (As explained below, this step is taken only once in case of a loop.) If a receiver discovers that two other sources are colliding, it MAY keep the packets from one and discard the packets from the other when this can be detected by different source transport addresses or CNAMEs.

Jitter Buffer

• The resolution of the clock MUST be sufficient for the desired synchronization accuracy and for measuring packet arrival jitter (one tick per video frame is typically not sufficient).
• RTP real-time protocol depends on its core features and processes for reliable and smooth real-time data transmission.
• Reverse reconsideration is also used to possibly shorten the delay before sending RTCP SR when transitioning from passive receiver to active sender mode.
• The disadvantage is that receivers on the output side don’t have any control over which sources are passed through or muted, unless some mechanism is implemented for remote control of the mixer.
• It is RECOMMENDED that stronger encryption algorithms such as Triple-DES be used in place of the default algorithm, and noted that the SRTP profile based on AES will be the correct choice in the future.
• The Payload Type field in the RTP header tells the receiver which codec was used to encode the media data.

Without a jitter buffer, this variation would produce choppy, uneven playback. The report interval scales with the number of participants, ensuring that luckygans casino RTCP traffic remains manageable even in large sessions. While RTP carries the media data, RTCP carries control information that enables quality monitoring, adaptive streaming, and synchronization. The Payload Type field in the RTP header tells the receiver which codec was used to encode the media data.

• If it can be assumed that packet loss is independent of packet size, then the number of packets received by a particular receiver times the average payload size (or the corresponding packet size) gives the apparent throughput available to that receiver.
• An end system can act as one or more synchronization sources in a particular RTP session, but typically only one.
• Since the SSRC identifier may change if a conflict is discovered or a program is restarted, receivers require the CNAME to keep track of each participant.
• Congestion Control All transport protocols used on the Internet need to address congestion control in some way .
• Where bandwidth is an issue and using a lower bitrate doesn’t help enough, SRT was designed to deliver low-latency video and other media across network conditions.
• O The interval between RTCP packets is varied randomly over the range 0.5,1.5 times the calculated interval to avoid unintended synchronization of all participants .

RTP Header Structure

Security Considerations RTP suffers from the same security liabilities as the underlying protocols. Those are the RTCP fraction of session bandwidth, the minimum report interval, and the bandwidth split between senders and receivers. A profile for audio and video applications may be found in the companion RFC 3551. Carrying several RTP packets in one network or transport packet reduces header overhead and may simplify synchronization between different streams. A profile MAY specify a framing method to be used even when RTP is carried in protocols that do provide framing in order to allow carrying several RTP packets in one lower-layer protocol data unit, such as a UDP packet.
RTP sessions are typically initiated between communicating peers using a signaling protocol, such as H.323, the Session Initiation Protocol (SIP), RTSP, or Jingle (XMPP). The control protocol, RTCP, is used for quality of service (QoS) feedback and synchronization between the media streams. Information provided by this protocol includes timestamps (for synchronization), sequence numbers (for packet loss and reordering detection) and the payload format, which indicates the encoded format of the data. RTP is used in conjunction with other protocols such as H.323 and RTSP. RTP is designed for end-to-end, real-time transfer of streaming media.

Methods for Ensuring QoS in RTP Streams

RTP is not an exception, but because the data transported over RTP is often inelastic (generated at a fixed or controlled rate), the means to control congestion in RTP may be quite different from those for other transport protocols such as TCP. Congestion Control All transport protocols used on the Internet need to address congestion control in some way . It is expected that authentication and integrity services will be provided by lower layer protocols.

A synchronization source may change its data format, e.g., audio encoding, over time. Examples of synchronization sources include the sender of a stream of packets derived from a signal source such as a microphone or a camera, or an RTP mixer (see below). If each participant sends RTCP feedback about data received from one other participant only back to that participant, then the conference is composed of three separate point-to-point RTP sessions. A participant distinguishes multiple RTP sessions by reception of different sessions using different pairs of destination transport addresses, where a pair of transport addresses comprises one network address plus a pair of ports for RTP and RTCP. A participant may be involved in multiple RTP sessions at the same time. Some underlying protocols may require an encapsulation of the RTP packet to be defined.

Can RTP stream both audio and video simultaneously?

The framework ensures the delivery of a smooth and synchronized audio or video stream using features like packetization, timestamping, and sequence numbering. The main purpose of RTP streaming is to provide a reliable framework for delivering real-time communication. That addition works alongside RTP, providing statistics and feedback about the quality of service of real-time sessions. It was initially intended to provide a standardized protocol for moving real-time audio and video over IP networks. So, the goal of QoS is to prioritize data packets and maximize the use of the available bandwidth without compromising the performance of critical applications. Which one you choose depends on the nature of your application and your preferred trade-off between streaming quality and playback continuity.
Standards Track Page 74 RFC 3550 RTP July 2003 Appendix A – Algorithms We provide examples of C code for aspects of RTP sender and receiver algorithms. Acknowledgments This memorandum is based on discussions within the IETF Audio/Video Transport working group chaired by Stephen Casner and Colin Perkins. These names are for use by higher-level control protocols, such as the Session Description Protocol (SDP), RFC 2327 , to refer to transport methods. Rightly or not, users may be more sensitive to privacy concerns with audio and video communication than they have been with more traditional forms of network communication . In addition, RTP may be sent via IP multicast, which provides no direct means for a sender to know all the receivers of the data sent and therefore no measure of privacy.
O Timing out a participant is to be based on inactivity for a number of RTCP report intervals calculated using the receiver RTCP bandwidth fraction even for active senders. The regeneration of synchronization information by mixers also means that receivers can’t do inter-media synchronization of the original streams. The interarrival jitter J is defined to be the mean deviation (smoothed absolute value) of the difference D in packet spacing at the receiver compared to the sender for a pair of packets. This correspondence may be used for intra- and inter-media synchronization for sources whose NTP timestamps are synchronized, and may be used by media-independent receivers to estimate the nominal RTP clock frequency. However, doing so may be appropriate for systems operating on unidirectional links or for sessions that don’t require feedback on the quality of reception or liveness of receivers and that have other means to avoid congestion.