Real time transport protocol

The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.

1. Real Time Transport Protocol

2. Concept Of RTP A protocol is designed to handle real-time traffic (like audio and video) of the Internet, is known as  Real Time Transport Protocol (RTP) . 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 . RTP supports different formats of files like MPEG and MJPEG. It is very sensitive to packet delays and less sensitive to packet loss.

3. History of RTP This protocol is developed by Internet Engineering Task Force (IETF) of four members: S. Casner (Packet Design) V. Jacobson (Packet Design) H. Schulzrinne (Columbia University) R. Frederick (Blue Coat Systems Inc.) RTP is first time published in 1996 and known as  RFC 1889 . And next it published in 2003 with name of  RFC 3550 .

4. RTP Format

5. Applications of RTP 1. Simple Multicast Audio Conference Initially the Host of the conference through some allocation mechanism obtains a multicast group address and pair of ports. One port is used for audio data, and the other is used for control (RTCP) packets. This address and port information is distributed to the intended participants. If privacy is desired, the data and control packets may be encrypted, in which case an encryption key must also be generated and distributed. Each participant sends the audio data in small chunks (say 20ms) or packets.

6. Applications of RTP 2. Audio and Video Conference If both audio and video media are used in a conference, they are transmitted as separate RTP sessions RTCP packets are transmitted for each medium using two different UDP port pairs and/or multicast addresses. The canonical name or CNAME of individual participants are used to match the audio and video sessions. For Example : Ongoing Presentation, Google Meet

7. Applications of RTP 3 . Mixers and Translators So far, we have assumed that all sites want to receive media data in the same format. However, this may not always be appropriate. For users having connections of different bandwidth or those working behind a firewall which won't allow IP packets to pass will need some extra processing. This is done in the form of  mixers [ T his mixer resynchronizes incoming audio packets to reconstruct the constant 20 ms spacing generated by the sender, mixes these reconstructed audio streams into a single stream, translates the audio encoding to a lower-bandwidth one and forwards the lower-bandwidth packet stream across the low-speed link]   and  translators .

8. Data Communication Self Learning Name : Swaroop Sorte Class : 5 th Sem CSE Roll : CS18053 Topic : Real Time Transport Protocol