RTP REAL TIME TRANSPORT PROTOCOL
BY RAGAVENDRA
A(1731310019)
PRESENTATION OUTLINE
Overview
Introduction What is RTP
Protocol structure RTP generalities RTP packets
Working of RTP RTCP
OVERVIEW
The Real-time Transport Protocol defines a
standardized packet format for delivering audio and video over networks
->
It provides transparent transfer of data between end usersUSAGE:
Faster
Avoid Starvation
Monitoring services
Independent of network protocol
INTRODUCTION
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(instant messaging) features.
ORIGIN:
Developed by the Audio-Video Transport Working Group of
the IETF(Internet Engg Task Force)
H. Schulzrinne (Columbia University)
S. Casner (Packet Design)
R. Frederick (Blue Coat Systems Inc.)
V. Jacobson (Packet Design)
First published as RFC 1889 in the year 1996
WHAT IS RTP
RTP consists of two components a data and a control part. The latter is called RTCP.
The data part of RTP is a thin protocol providing support
for applications with real-time properties such as
continuous media (e.g., audio and video), timing reconstruction,
loss detection,
security and content identification.
RTCP provides support for real-time conferencing of
groups of any size within an internet. This support includes:
Source identification and support for gateways like
audio and video bridges
PROTOCOL STRUCTURE
Application layer protocol
Typically used on top of UDP
Applications that use RTP are:
Less sensitive to packet loss Very sensitive to packet delays
UDP provides key services:
RTP GENERALITIES
carry data that has real-time properties
Simple Multicast Audio Conference Audio and Video Conference
Scalable: unicast, multicast, from 2 to
Timing:
intra-media synchronization: remove jitter with playout buffers
GENERAL SCENARIO
One-to-one One-to-many Many-to-many Local
transmission
(access within
one machine)
RTCP (Sender
and
RTP PACKETS
Consist of RTP header, optional payload headers and the
payload itself
RTP overhead = 12 bytes
IP+UDP+RTP overhead = 20+8+12 = 40 bytes
It is advisable to keep coded slice sizes as close to, but never
bigger than, the MTU size (largest size of a packet that can be transmitted without being split/recombined on the transport and network layer), because:
1. It optimizes the payload/header overhead relationship
2. Minimizes the loss probability of a (fragmented) coded slice due to the loss of a single fragment on the network/transport layer and the resulting discarding of all other fragments belonging to the coded
slice
MTU sizes: ~1500 bytes for wireline IP links (max. size of an
RTP PACKET FORMAT
V) Version; 2 bits (P) Padding; 1 bit.
(X) Extension; 1 bit. (CC) CSRC Count; 4 bits.
(M) Marker; 1 bit. (PT) Payload Type; 7 bits.
Sequence Number; 16 bits. Time Stamp; 32 bits.
PACKET HEADER INFO:
RTP header contains the following:
sequence number (used for packet-loss
detection),
timestamp (timing information,
synchronization of media streams),
Payload type (identifies the media codec of the
payload),
marker bit (detecting the end of a group of
related packets),
source identifiers(contributing and
HOW RTP WORKS
Ip header->udp header->RTP header ->RTP Video Payload Ip header->udp header->RTP header
->RTP audio payload
Video and audio payloads are sent separately
Uses sequence number to synchronise audio and
RTP SESSION
RTP session is sending and receiving of RTP data by a group of participants
For each participant, a session is a pair of transport addresses used to
communicate with the group
If multiple media types are communicated by the group, the
RTP BASIC DATA TRANSMISSION
Same synchronous source Different seq no
Different payloads and timestamp
End user
RTCP
Real-time Transport Control Protocol (RTCP) for transmitting
control information.
It consists of a data and control (RTCP) component that work
together.
It provides support for streaming data, Timing reconstruction, loss
detection, etc.
This is the control part of RTP, and provides the following functions:
Data delivery monitoring
Source identification
CONTD…
Which port numbers do they use?
RTP or RTCP are not assigned any well-known port number.
The port numbers are assigned on demand.
Restriction:
For RTP, port number must be even.2p For RTCP, port number must be odd.2p+1
The control protocol RTCP is used to specify quality of service
(QoS) feedback and synchronization between the media streams.
The bandwidth of RTCP traffic compared to RTP is small,
EG: SIMPLE MULTICAST AUDIO
CONFERENCE
1. Call connection established
2. Audio sampled at 20ms durations
3. Each data chunk is packaged with an RTP header
4. RTP packet is wrapped around UDP packet 5. Sent through network
6. Receiver receives and parses RTP header
MIXER AND TRANSLATOR
Accommodate participant network resources
->Mixer – Low Bandwidth
->Mixer – Combining media streams
PAYLOAD FORMATS
Media types:
Application -> smpte336m Application -> H.261
Audio -> ac3
Audio ->attrac3 (44100-clockrate) Text -> rtx
Text -> ulpfec
Video ->h263 (90000-clockrate) Video ->nv ……..etc
CONCLUSION
RTP provides powerful instruments for
adaptive video transmission
Potential applications include wireless links Optimization can be done within the frames of