RTP and RTCP are protocols used for delivering audio and video over IP networks. RTP carries the media streams, while RTCP monitors transmission quality and aids synchronization. RTCP sends periodic reports containing statistics like packet loss to allow senders to adapt transmission rates. While TCP is not suitable due to retransmissions causing delay, RTP supports features needed for real-time multimedia like sequencing, timestamps, error concealment, and quality of service feedback.

1. RTP & RTCP

2. Need of RTP
• Real-time multimedia applications
– Video teleconferencing
– Internet Telephony (VoIP)
– Internet audio, video streaming

3. Real time multimedia application
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

4. Real time multimedia application
requirements
– Rate control –
– Sender reduces sending rate adaptively to network
congestion
– Quality of Service (QoS) feedback –
– From receiver to sender for operation adjustment

5. Limitations of TCP
• TCP is not used because-
• TCP does retransmissions  unbounded
delays
• No provision for time stampings(slow and
noisy walks )
• TCP does not support multicast
• TCP congestion control (slow-start) unsuitable
for real-time transport

6. RTP & RTCP Overview
• Standardized packet format for delivering audio and video
over IP networks.
• Used 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 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.

7. RTP & RTCP Overview
• RTP is originated and received on even port numbers
and the associated RTCP communication uses the next
higher odd port number.
• RTP is designed for end-to-end, real-time, transfer of
stream data. The protocol provides facility for jitter
compensation and detection of out of sequence arrival
in data, that are common during transmissions on an IP
network.
• RTP supports data transfer to multiple destinations
through IP multicast. RTP is regarded as the primary
standard for audio/video transport in IP networks and is
used with an associated profile and payload format.

8. Protocol stack for multimedia
services
RTSP
TCP
(till now)

9. 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)

10. RTP Packets
• Reasons are as follows :-
• It optimizes the payload/header overhead relationship
• 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 Ethernet
packet)
• ~100 bytes in wireless environments

11. RTP packet example
• DVD quality video transmission:-
• 30 frames/s, 720x480 resolution, 3 bytes per pixel
• 31,104,000 bytes/s raw rate
• 311,040 bytes/s compressed data rate (100x
compression)
• MTU = 1500 bytes:-
• 311,040/1460 = 213 packets/s > 319,500 bytes/s
(required throughput including overhead)
• MTU = 100 bytes:-
• 311,040/60 = 5184 packets/s -> 518,400 bytes/s
(required throughput)

12. RTP Header
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

13. RTP Header
RTP header contains the following:-
• Marker Bit
• Detecting the end of a group of related packets
• source identifiers
• Contributing and Synchronizing

14. SIP 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.

15. 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

16. RTP Basics of Data Transmission
• RTP PDUs

17. RTP PDU Header Incremented by one
Sampling instant of first data octet
for each RTP PDU:
multiple PDUs can have same
timestamp PDU loss
Payload
detection
not necessarily monotonic type
used to synchronize different
Restore PDU
sequence
media streams
Identifies synchronization (used by mixer)
source Identifies contributing sources

19. Example – Mixers and Translators
• Mixers:-
• RTP-level entity that receives streams of RTP data packets
from one or more sources and combines them into a single
stream.
• Mixer is like a new RTP-level source to the receivers.
• Mixer can re-synchronize the incoming stream and
generates its own timing info.
• Translator
• Forwards RTP packets from different sources separately.
• Translator is more transparent. Receivers can identify
individual sources even though packets pass through the
same translator and carry the translator’s network source
address.

20. Translators and Mixers
• The real distinction between mixers and translators:-
• SSRC identifier is not changed at a translator, but is changed
at a mixer.
• They both use a different transport address (network
address + port) at the output side.
• Multiple data packets can be combined into one.
• Uses of translators and mixers:-
• Go-through firewalls
• Transcoding for low-bandwidth links
• Adding or removing encryption
• Emulating multicast address with one or more unicast
addresses.

21. Mixers
A mixer will typically have to define synchronization relationships
between streams.
 Sources that are mixed together become contributing sources
(CSRC)
 Mixer itself appears as a new source having a new SSRC

22. Translators
An intermediate system that…
 Connects two or more networks
 Multicasting through a firewall
 Modifies stream encoding, changing the stream’s timing
 Transparent to participants
 SSRC’s remain intact
end system 1 from ES1: SSRC=6
from ES1: SSRC=6 from ES2: SSRC=23
transl.1 transl.2
from ES2: SSRC=23 authorized tunnel
end system 2 firewall from ES1: SSRC=6
from ES2: SSRC=23

23. RTCP
• The RTP control protocol (RTCP) is based on the periodic
transmission of control packets to all participants in the
session, using the same distribution mechanism as the data
packets.
• The underlying protocol must provide multiplexing of the data
and control packets, for example using separate port numbers
with UDP.
• It is recommended that the fraction of the session bandwidth
allocated to the RTCP is 5%.
• The primary function of this protocol is to provide feedback on the
quality of the data distribution.
• RTCP

24. RTP Control Protocol (RTCP)
 RTCP specifies report PDUs exchanged
between sources and destinations of
multimedia information
 receiver reception report
 sender report
 source description report
 Reports contain statistics such as the
number of RTP-PDUs sent, number of
RTP-PDUs lost, inter-arrival jitter
 Used by application to modify sender
transmission rates and for diagnostics
purposes

25. RTCP PACKETS
• SR – sender report, for transmission
and reception statistics from
participants that are active senders
• RR - Receiver report, for reception
statistics from participants that are
not active senders and in combination
with SR for active senders reporting
on more than 31 sources
• SDES - Source description
items, including CNAME (Canonical
Name – RTP source identifier)
• BYE - Indicates end of participation
• APP - Application specific functions

26. RTCP PACKETS

27. Conclusions
• RTP provides powerful instruments for adaptive video transmission.
• Potential applications include wireless links.
• Optimization can be done within the frames of the protocol
specification (loosely defined packet sizes and RTCP communication
• frequency)