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:
1. S. Casner (Packet Design)
2. V. Jacobson (Packet Design)
3. H. Schulzrinne (Columbia University)
4. 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[This 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 : 5th Sem CSE
Roll : CS18053
Topic : Real Time Transport Protocol