RTP provides real-time transport of data such as audio and video by identifying payload types, sequencing packets, and including timestamps. It uses UDP and is augmented by RTCP to monitor delivery over multicast networks. RTP supports applications like audio/video conferencing by encoding and transmitting media in packets with RTP headers for reconstruction at receivers.
TCP & UDP Streaming Comparison and a Study on DCCP & SCTP ProtocolsPeter SHIN
As a graduate student work, I have compared the performance between TCP and UDP media streaming with empirical results. Also, I have researched on different attempts on UDP to be more reliable, but why its progress has not been as fast as possible
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.
Overview of SCTP (Stream Control Transmission Protocol)Peter R. Egli
Overview of SCTP (Stream Control Transmission Protocol), outlining the main features and capabilities of SCTP.
SCTP is a transport protocol that overcomes many of the shortcomings of TCP, namely head-of-line blocking and stream-oriented transmission.
SCTP supports multiple streams within a connection and preserves boundaries of application messages thus greatly simplifying communication.
Additionally, SCTP supports multi-homing which increases availability in applications with high reliability demands.
SCTP inherits much of the congestion, flow and error control mechanisms of TCP.
SCTP has its roots in telecom carrier networks for use in transitional voice over IP scenarios.
However, SCTP is generic so that it is applicable in many enterprise applications as well.
This slide contains fundamental concept about Quality of Service (QoS) technology and various types of Queuing Methods, according to the latest version of Cisco books (CCIE R&S and CCIE SP) and i taught it at IRAN TIC company.
Overview of SCTP (Stream Control Transmission Protocol)Peter R. Egli
Overview of SCTP (Stream Control Transmission Protocol), outlining the main features and capabilities of SCTP.
SCTP is a transport protocol that overcomes many of the shortcomings of TCP, namely head-of-line blocking and stream-oriented transmission.
SCTP supports multiple streams within a connection and preserves boundaries of application messages thus greatly simplifying communication.
Additionally, SCTP supports multi-homing which increases availability in applications with high reliability demands.
SCTP inherits much of the congestion, flow and error control mechanisms of TCP.
SCTP has its roots in telecom carrier networks for use in transitional voice over IP scenarios.
However, SCTP is generic so that it is applicable in many enterprise applications as well.
TCP & UDP Streaming Comparison and a Study on DCCP & SCTP ProtocolsPeter SHIN
As a graduate student work, I have compared the performance between TCP and UDP media streaming with empirical results. Also, I have researched on different attempts on UDP to be more reliable, but why its progress has not been as fast as possible
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.
Overview of SCTP (Stream Control Transmission Protocol)Peter R. Egli
Overview of SCTP (Stream Control Transmission Protocol), outlining the main features and capabilities of SCTP.
SCTP is a transport protocol that overcomes many of the shortcomings of TCP, namely head-of-line blocking and stream-oriented transmission.
SCTP supports multiple streams within a connection and preserves boundaries of application messages thus greatly simplifying communication.
Additionally, SCTP supports multi-homing which increases availability in applications with high reliability demands.
SCTP inherits much of the congestion, flow and error control mechanisms of TCP.
SCTP has its roots in telecom carrier networks for use in transitional voice over IP scenarios.
However, SCTP is generic so that it is applicable in many enterprise applications as well.
This slide contains fundamental concept about Quality of Service (QoS) technology and various types of Queuing Methods, according to the latest version of Cisco books (CCIE R&S and CCIE SP) and i taught it at IRAN TIC company.
Overview of SCTP (Stream Control Transmission Protocol)Peter R. Egli
Overview of SCTP (Stream Control Transmission Protocol), outlining the main features and capabilities of SCTP.
SCTP is a transport protocol that overcomes many of the shortcomings of TCP, namely head-of-line blocking and stream-oriented transmission.
SCTP supports multiple streams within a connection and preserves boundaries of application messages thus greatly simplifying communication.
Additionally, SCTP supports multi-homing which increases availability in applications with high reliability demands.
SCTP inherits much of the congestion, flow and error control mechanisms of TCP.
SCTP has its roots in telecom carrier networks for use in transitional voice over IP scenarios.
However, SCTP is generic so that it is applicable in many enterprise applications as well.
In the last few years, video streaming facilities over TCP or UDP, such as YouTube, Facetime, Daily-motion, Mobile video calling have become more and more popular. The important
challenge in streaming broadcasting over the Internet is to spread the uppermost potential quality,
observe to the broadcasting play out time limitation, and efficiently and equally share the offered
bandwidth with TCP or UDP, and additional traffic types. This work familiarizes the Streaming
Media Data Congestion Control protocol (SMDCC), a new adaptive broadcasting streaming
congestion management protocol in which the connection’s data packets transmission frequency is
adjusted allowing to the dynamic bandwidth share of connection using SMDCC, the bandwidth share
of a connection is projected using algorithms similar to those introduced in TCP Westwood. SMDCC
avoids the Slow Jump phase in TCP. As a result, SMDCC does not show the pronounced rate
alternations distinguishing of modern TCP, so providing congestion control that is more appropriate
for streaming broadcasting applications. Besides, SMDCC is fair, sharing the bandwidth equitably
among a set of SMDCC connections. Main benefit is robustness when packet harms are due to
indiscriminate errors, which is typical of wireless links and is becoming an increasing concern due to
the emergence of wireless Internet access. In the presence of indiscriminate errors, SMDCC is also
approachable to TCP Tahoe and Reno (TTR). We provide simulation results using the ns3 simulator
for our protocol running together with TCP Tahoe and Reno.
What is the role of sequence numbers in the rdt protocolWhat is t.pdfmonikajain201
What is the role of sequence numbers in the rdt protocol?
What is the role of timers in the rdt protocol?
Solution
The internet network layer provides only best effort service with no guarantee that packets arrive
at their destination. Also, since each packet is routed individually it is possible that packets are
received out of order. For connection-oriented service provided by TCP, it is necessary to have a
reliable data transfer (RDT) protocol to ensure delivery of all packets and to enable the receiver
to deliver the packets in order to its application layer.
A simple alternating bit RDT protocol can be designed using some basic tools. This protocol is
also known as a stop-and-wait protocol: after sending each packet the sender stops and waits for
feedback from the receiver indicating that the packet has been received.
Stop-and-wait RDT protocols have poor performance in a long-distance connection. At best, the
sender can only transmit one packet per round-trip time. For a 1000 mile connection this
amounts to approximately 1 packet (about 1500 bytes) every 20 ms. That results in a pathetic 75
KB per second rate.
To improve transmission rates, a realistic RDT protocol must use pipelining. This allows the
sender to have a large number of packets \"in the pipeline\". This phrase refers to packets that
have been sent but whose receipt has not yet verified by the receiver..
The performance of wireless ad hoc networks is impacted significantly by the way TCP reacts to lost packets. TCP was designed specifically for wired, reliable networks; thus, any packet loss is attributed to congestion in the network. This assumption does not hold in wireless networks as most packet loss is due to link failure. In our research we analyzed several implementations of TCP, including TCP Vegas, TCP Feedback, and SACK TCP, by measuring throughput, retransmissions, and duplicate acknowledgements through simulation with ns-2. We discovered that TCP throughput is related to the number of hops in the path, and thus depends on the performance of the underlying routing protocol, which was DSR in our research.
IMPACT OF CONTENTION WINDOW ON CONGESTION CONTROL ALGORITHMS FOR WIRELESS ADH...cscpconf
TCP congestion control mechanism is highly dependent on MAC layer Backoff algorithms that
predict the optimal Contention Window size to increase the TCP performance in wireless adhoc
network. This paper critically examines the impact of Contention Window in TCP congestion
control approaches. The modified TCP congestion control method gives the stability of
congestion window which provides higher throughput and shorter delay than the traditional TCP. Various Backoff algorithms that are used to adjust Contention Window are simulatedusing NS2 along with modified TCP and their performance are analyzed to depict the influence of Contention Window in TCP performance considering the metrics such as throughput, delay, packet loss and end-to-end delay
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1. RTP: A Transport Protocol for
Real-Time Applications
Provides end-to-end delivery services for data with real-time
characteristics, such as interactive audio and video.
Those services include payload type identification, sequence
numbering, timestamping and delivery monitoring.
Applications typically run RTP on top of UDP
2. RTCP
RTP is augmented by a control protocol (RTCP) to allow
monitoring of the data delivery in a manner scalable to large
multicast networks, and to provide minimal control and
identification functionality.
3. RTP Use Scenarios
Simple Multicast Audio Conference
The audio conferencing application used by each conference
participant sends audio data in small chunks of, say, 20 ms
duration.
Each chunk of audio data is preceded by an RTP header; RTP
header and data are in turn contained in a UDP packet.
The RTP header indicates what type of audio encoding (such
as PCM, ADPCM or LPC) is contained in each packet.
4. TP header contains timing information and a sequence
number that allow the receivers to reconstruct the timing
produced by the source.
The sequence number can also be used by the receiver to
estimate how many packets are being lost.
the audio application in the conference periodically multicasts
a reception report plus the name of its user on the RTCP port.
The reception report indicates how well the current speaker is
being received.
A site sends the RTCP BYE packet when it leaves the
conference.
5. Audio and Video Conference
Audio and video media are are transmitted as separate RTP session
and RTCP packets are transmitted for each medium using two
different UDP port pairs and/or multicast addresses.
There is no direct coupling at the RTP level between the audio and
video sessions, except that a user participating in both sessions
should use the same distinguished (canonical) name in the RTCP
packets for both so that the sessions can be associated.
Despite the separation, synchronized playback of a source's audio
and video can be achieved using timing information carried in the
RTP packets for both sessions.
6. MIXER
Receives streams of RTP data packets from one or more
sources, possibly changes the data format, combines the
streams in some manner and then forwards the
combined stream.
All data packets forwarded by a mixer will be marked
with the mixer's own SSRC identifier. In order to
preserve the identity of the original sources
contributing to the mixed packet
7. Translator
Forwards RTP packets with their SSRC identifier intact
May change the encoding of the data and thus the RTP
data payload type
8. RTP Header
Payload type Sequence number
Timestamp
SSRC identifier
9. RTCP
Is based on the periodic transmission of control packets to
all participants in the session and perform the following
functions:
provide feedback on the quality of the data distribution and
allows one who is observing problems to evaluate whether
those problems are local or global.
10. RTCP carries an identifier for an RTP source called the
canonical name or CNAME. Receivers use CNAME to associate
multiple data streams from a given participant in a set of related
RTP sessions, for example to synchronize audio and video.
11. RTCP Packet Format
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.
SDES: Source description items, including CNAME.
BYE: Indicates end of participation.
APP: Application specific functions.
12. RTCP Transmission Interval
RTP is designed to allow an application to scale automatically over
session sizes ranging from a few participants to thousands.
In an audio conference the data traffic is inherently self- limiting
because only one or two people will speak at a time, so with
multicast distribution the data rate on any given link remains
relatively constant independent of the number of participants.
However, the control traffic is not self-limiting. If the reception
reports from each participant were sent at a constant rate, the
control traffic would grow linearly with the number of
participants.
13. To maintain scalability, the average interval between packets from
a session participant should scale with the group size.
The control traffic should be limited to a small and known
fraction of the session bandwidth:
small so that the primary function of the transport protocol to carry data is
not impaired;
known so that each participant can independently calculate its share.
It is suggested that the fraction of the session bandwidth allocated
to RTCP be fixed at 5%
14. Receiver Report RTCP Packet
RC Type Length
SSRC of packet sender
SSRC of first source
Report block 1
Fraction lost Cumulative number of packet lost
Interarrival jitter
Last SR
Delay since last SR
Report block 2
15. SRC Identifier Allocation
The SSRC identifier carried in the RTP header and in various
fields of RTCP packets is a random 32-bit number that is
required to be globally unique within an RTP session.
All RTP implementations must be prepared to detect collisions
and take the appropriate actions to resolve them.
If a source discovers at any time that another source is using the
same SSRC identifier as its own, it must send an RTCP BYE
packet for the old identifier and choose another random one.
If a receiver discovers that two other sources are colliding, it
may keep the packets from one and discard the packets from the
other.