0% found this document useful (0 votes)
4 views

3.9 RTP Protocol architecture

The Real-time Transport Protocol (RTP) is designed for delivering audio and video over IP networks, primarily used in streaming media applications like telephony and video conferencing. RTP operates over UDP and works alongside the RTP Control Protocol (RTCP) to monitor quality of service and synchronize multiple streams. Key features of RTP include payload type identification, sequencing, timestamping, and error concealment, making it suitable for real-time multimedia communication.

Uploaded by

Surya Kanth
Copyright
© © All Rights Reserved
Available Formats
Download as PPTX, PDF, TXT or read online on Scribd
0% found this document useful (0 votes)
4 views

3.9 RTP Protocol architecture

The Real-time Transport Protocol (RTP) is designed for delivering audio and video over IP networks, primarily used in streaming media applications like telephony and video conferencing. RTP operates over UDP and works alongside the RTP Control Protocol (RTCP) to monitor quality of service and synchronize multiple streams. Key features of RTP include payload type identification, sequencing, timestamping, and error concealment, making it suitable for real-time multimedia communication.

Uploaded by

Surya Kanth
Copyright
© © All Rights Reserved
Available Formats
Download as PPTX, PDF, TXT or read online on Scribd
You are on page 1/ 24

DEPARTMENT OF ELECTRONICS AND

COMMUNICATION ENGINEERING

MODULE 3
TOPIC 3.8 : RTP PROTOCOL ARCHITECTURE
INTRODUCTION
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.
RTP was developed by the Audio-Video Transport Working
Group of the Internet Engineering Task Force (IETF) and first
published in 1996 as RFC 1889 which was then superseded
by RFC 3550 in 2003
INTRODUCTION
• RTP typically runs over User Datagram Protocol (UDP).
• RTP is used in conjunction with the RTP Control
Protocol (RTCP). While RTP carries the media streams (e.g.,
audio and video), RTCP is used to monitor transmission
statistics and quality of service (QoS) and
aids synchronization of multiple streams.
• RTP is one of the technical foundations of Voice over IP and in
this context is often used in conjunction with a signaling
protocol such as the Session Initiation Protocol (SIP) which
establishes connections across the network.
RTP:
TCP not suited to real time distributed application
Point to point so not suitable for multicast
Retransmitted segments arrive out of order
No way to associate timing with segments
UDP does not include timing information nor any support for
real time applications
Solution is real-time transport protocol RTP
GOALS OF RTP
To provide end-to-end delivery services for data with real-time
characteristics.
Primarily designed to satisfy the needs of multi- participant
multimedia conferences (using multicast distribution if provided
by the underlying network ).
Services include payload type identification, sequence
numbering, timestamping and delivery monitoring.
RTP DEFINED
RTP consists of two closely-linked parts:

 RTP- transport protocol that has real-time properties.

 RTCP – control protocol to monitor the quality of service


(QoS) and to convey session information.
RTP PROTOCOL ARCHITECTURE
RTP PROTOCOL ARCHITECTURE
Close coupling between protocol and application layer
functionality
Framework for application to implement single protocol
Application level framing
Integrated layer processing
Application Level Framing
Recovery of lost data done by application rather than transport layer
Application may accept less than perfect delivery
 Real time audio and video
 Inform source about quality of delivery rather than retransmit
 Source can switch to lower quality
Application may provide data for retransmission
 Sending application may recompute lost values rather than storing them
 Sending application can provide revised values
 Can send new data to “fix” consequences of loss
Lower layers deal with data in units provided by application
Application data units (ADU)
Integrated Layer Processing
Adjacent layers in protocol stack tightly coupled
Allows out of order or parallel functions from different layers
RTP Architecture Diagram
Multicast Support
Each RTP data unit includes:
Source identifier
Timestamp
Payload format
Relays
Intermediate system acting as receiver and transmitter for
given protocol layer
Mixers
Receives streams of RTP packets from one or more sources
Combines streams
Forwards new stream
Translators
Produce one or more outgoing RTP packets for each incoming packet
E.g. convert video to lower quality
RTP Header
RTP FEATURES
Streaming performance requirements
– Sequencing
– to report PDU loss
– to report PDU reordering
– to perform out-of-order decoding

– Time stamping and Buffering


– for play out
– for jitter and delay calculation

– Payload type identification


– for media interpretation

– Error concealment –covers up errors from lost PDU by using redundancy in most-
adjacent-frame

– Quality of Service (QoS) feedback – from receiver to sender for operation


adjustment

– Rate control –sender reduces sending rate adaptively to network congestion


Support from transport layers
 Why not only RTP?

RTP Does Not

 Assume the underlying network is reliable and delivers PDUs in sequence.


 Uses sequence number.
 Provide timely delivery or other QoS guarantees.
 However, it does provide necessary data to application to order packets and adjust signal quality.
 Relies on other protocols like RTCP and lower layers (e.g. UDP, TCP) to do so.
 Handle connection setups or tear-downs.
Need other protocols like SIP or H.323
 Handle resource reservation .
Need other protocols like RSVP
 Define media data formats or encodings.

Need media specific profiles.


 Why not TCP?

 TCP does retransmissions  unbounded delays due to (Acks, Flow control, windowing)
 No provision for time stamping
 TCP does not support multicast
 TCP congestion control (slow-start) unsuitable for real-time transport (A-V media)

 Why not UDP?

UDP offers datagram-like service


Connectionless , Unreliable , Unordered RTCP
No Flow , Error , Congestion control
Port numbers

RTP + UDP usually used for multimedia services


RTP Packets
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 transmission of each
medium constitutes a session.
RTP Synchronization Source

 synchronization source - each source of RTP PDUs


 Identified by a unique,randomly chosen 32-bit ID (the SSRC)
A host generating multiple streams within a single RTP must use a different SSRC per
stream
RTP Basics of Data Transmission
RTP PDUs
How does Sequence number and Timestamp help ?

Audio silence example:


Consider audio data
– What should the sender do during silence?
Not send anything

– Why might this cause problems? silence


Receiver cannot distinguish between loss and
silence

Solution:

– After receiving no PDUs for a while, next PDU received at the receiver
will reflect a big jump in timestamp, but have the correct next seq. no.
Thus, receiver knows what happened.
THANK YOU

You might also like