Why is rtp used

RTP is a data transport protocol, whose mission is to move data between two endpoints as efficiently as possible under current conditions. Those conditions may be affected by everything from the underlying layers of the network stack to the physical network connection, the intervening networks, the performance of the remote endpoint, noise levels, traffic levels, and so forth. It isn't adequate for the purposes of fully managing users, memberships, permissions, and so forth, but provides the basics needed for an unrestricted multi-user communication session.

Some of the more noteworthy things RTP doesn't include:. Because the streams for an RTCPeerConnection are implemented using RTP and the interfaces above , you can take advantage of the access this gives you to the internals of streams to make adjustments.

Among the simplest things you can do is to implement a "hold" feature, wherein a participant in a call can click a button and turn off their microphone, begin sending music to the other peer instead, and stop accepting incoming audio. Note: This example makes use of modern JavaScript features including async functions and the await expression. This enormously simplifies and makes far more readable the code dealing with the promises returned by WebRTC methods.

In the examples below, we'll refer to the peer which is turning "hold" mode on and off as the local peer and the user being placed on hold as the remote peer. The Real Time Transport Protocol is able to code multimedia data streams such as audio or video, divide them into packets and transmit them over an IP network.

RTP allows data to be exchanged in Unicast as well as Multicast communication. Every RTP packet has a header which contains a variety of information related to the contents of the packet and its transfer. The header, for example, includes version and sequence numbers, the unique sender ID, time stamp and information about the data format.

RTP — short for Real-time Transport Protocol defines a standard packet format for delivering audio and video over the Internet. It is defined in RFC It was developed by the Audio Video Transport Working group and was first published in RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features.

While RTP carries the media streams e. RTP is originated and received on even port numbers and the associated RTCP communication uses the next higher odd port number. As its name implies, the design goal for RTP is the end-to-end streaming in real-time of media-related data. RTP includes mechanisms for jitter compensation, packet loss detection, as well as out-of-order data packet delivery, issues that are especially common in UDP User Datagram Protocol transmissions over IP.

As RTP enables data transfer to multiple destination end-points in parallel via IP multicast , it is the primary standard eployed for audio and video IP network transfers. The mechanisms for the associated profile and payload format, referenced in the design of the RTP architecture , are implemented on the level of the application layer, instead of the operating system layer.

Applications such as VoIP that need to employ real-time streaming of multimedia data, typically require the timely delivery of data, with varying tolerance in packet loss.