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.
RTP is a protocol for delivering audio and video over IP networks. It works on top of UDP to provide real-time media streaming. RTP provides mechanisms for sequencing packets, time stamping media, and detecting packet loss to enable real-time playback of media. RTSP is a control protocol that establishes and controls media sessions between end points, allowing features like play, pause, and teardown of RTP streams. Together, RTP and RTSP enable real-time streaming applications like video calls and media streaming.
The document discusses Real-Time Transport Protocol (RTP) which is an Internet standard for delivering audio and video over IP networks. RTP is designed for real-time streaming applications that require low latency. It provides mechanisms for synchronization, transport of timing information and mechanisms to detect packet loss for media transport. The document outlines the fundamental design philosophies, standard elements and packet format of RTP as well as its companion protocol, RTCP, which is used for quality of service feedback.
Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) are used for streaming real-time multimedia such as video calls and audio/video streaming. RTP handles the transfer of data and sequencing while RTCP monitors transmission quality and allows endpoints to synchronize. Real-Time Streaming Protocol (RTSP) provides the control channel for multimedia streaming using RTP, allowing clients to connect, control delivery, and terminate sessions.
An overview of TCP (Transmission Control Protocol)Ammad Marwat
This document provides an overview of the Transmission Control Protocol (TCP). It discusses TCP's key characteristics of being connection-oriented and reliable. It describes how TCP establishes connections using a 3-way handshake and provides flow control, error control, and congestion control to ensure reliable data delivery. The document also explains sliding window protocols and how they help make transmission more efficient while preventing receiver overflow and network congestion.
This document discusses the Transmission Control Protocol (TCP) which provides reliable, connection-oriented data transmission over the internet. TCP establishes a virtual connection between endpoints, ensuring reliable delivery through mechanisms like positive acknowledgement and retransmission. It uses a sliding window algorithm to guarantee reliable and in-order delivery while enforcing flow control between sender and receiver. Key aspects of TCP include connection establishment and termination, port numbers, segments, headers, and addressing end-to-end issues over heterogeneous networks.
This document provides an overview of TCP congestion control algorithms. It describes the basic additive increase/multiplicative decrease approach and key mechanisms like slow start, fast retransmit, and fast recovery. It also discusses algorithms for setting the retransmission timeout value and adaptations made in protocols like New Reno and Cubic.
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.
RTP is a protocol for delivering audio and video over IP networks. It works on top of UDP to provide real-time media streaming. RTP provides mechanisms for sequencing packets, time stamping media, and detecting packet loss to enable real-time playback of media. RTSP is a control protocol that establishes and controls media sessions between end points, allowing features like play, pause, and teardown of RTP streams. Together, RTP and RTSP enable real-time streaming applications like video calls and media streaming.
The document discusses Real-Time Transport Protocol (RTP) which is an Internet standard for delivering audio and video over IP networks. RTP is designed for real-time streaming applications that require low latency. It provides mechanisms for synchronization, transport of timing information and mechanisms to detect packet loss for media transport. The document outlines the fundamental design philosophies, standard elements and packet format of RTP as well as its companion protocol, RTCP, which is used for quality of service feedback.
Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) are used for streaming real-time multimedia such as video calls and audio/video streaming. RTP handles the transfer of data and sequencing while RTCP monitors transmission quality and allows endpoints to synchronize. Real-Time Streaming Protocol (RTSP) provides the control channel for multimedia streaming using RTP, allowing clients to connect, control delivery, and terminate sessions.
An overview of TCP (Transmission Control Protocol)Ammad Marwat
This document provides an overview of the Transmission Control Protocol (TCP). It discusses TCP's key characteristics of being connection-oriented and reliable. It describes how TCP establishes connections using a 3-way handshake and provides flow control, error control, and congestion control to ensure reliable data delivery. The document also explains sliding window protocols and how they help make transmission more efficient while preventing receiver overflow and network congestion.
This document discusses the Transmission Control Protocol (TCP) which provides reliable, connection-oriented data transmission over the internet. TCP establishes a virtual connection between endpoints, ensuring reliable delivery through mechanisms like positive acknowledgement and retransmission. It uses a sliding window algorithm to guarantee reliable and in-order delivery while enforcing flow control between sender and receiver. Key aspects of TCP include connection establishment and termination, port numbers, segments, headers, and addressing end-to-end issues over heterogeneous networks.
This document provides an overview of TCP congestion control algorithms. It describes the basic additive increase/multiplicative decrease approach and key mechanisms like slow start, fast retransmit, and fast recovery. It also discusses algorithms for setting the retransmission timeout value and adaptations made in protocols like New Reno and Cubic.
The document discusses several application layer protocols used in TCP/IP including HTTP, HTTPS, FTP, and Telnet. HTTP is used to access resources on the world wide web over port 80 and is stateless. HTTPS is a secure version of HTTP that encrypts communications over port 443. FTP is used to transfer files between hosts but sends data and passwords in clear text. Telnet allows users to access programs on remote computers.
This document discusses Go-Back-N ARQ, a method for error control in data transmission that uses pipelining and sliding windows. It introduces the concept of using sequence numbers, timers, acknowledgements, and resending frames to improve efficiency over Stop-and-Wait ARQ. Go-Back-N ARQ allows the sender to send multiple frames before receiving ACKs, but requires resending all frames after the lost one. It has higher efficiency than Stop-and-Wait but also has disadvantages related to buffering requirements and wasted bandwidth from resending uncorrupted frames.
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.
The document discusses transport layer protocols TCP and UDP. It provides an overview of process-to-process communication using transport layer protocols. It describes the roles, services, requirements, addressing, encapsulation, multiplexing, and error control functions of the transport layer. It specifically examines TCP and UDP, comparing their connection-oriented and connectionless services, typical applications, and segment/datagram formats.
The document discusses the Transport layer protocols TCP and UDP. It describes TCP as a connection-oriented protocol that provides reliable, ordered delivery of streams of data through mechanisms like sequencing, acknowledgment, flow control and error checking. UDP is described as a simpler connectionless protocol that provides best-effort delivery without checking for errors or lost packets. The key concepts of ports, sockets, multiplexing and demultiplexing are also covered, as well as the header formats and functions of TCP and UDP.
1) Distributed deadlock detection approaches include prevention, avoidance, detection, and resolution. Detection is the primary focus of the chapter.
2) Distributed deadlock detection modifies the bipartite graph strategy to use a wait-for graph (WFG) to model processes requesting resources from other processes. A deadlock exists if there is a directed cycle in the global WFG.
3) Distributed detection requires progress such that all deadlocks are found within a finite time, and safety such that no false detections occur due to network latencies. Approaches include centralized, distributed, and hierarchical control frameworks.
IGMP (Internet Group Management Protocol) allows multicast routers to track group memberships across multicast networks. It has three message types - query, membership report, and leave report. Upon receiving a query, hosts send membership reports to the router to join or leave groups. The router uses these reports to maintain a list of members for each multicast group on that network segment. IGMP messages are encapsulated in IP datagrams and Ethernet frames for transmission.
Distance Vector & Link state Routing AlgorithmMOHIT AGARWAL
1) Each router maintains a routing table containing the outgoing link and distance to reach each destination node. 2) Routers periodically share their routing tables with neighbors so each can update its own table. 3) This allows routers to continuously determine the shortest paths to all destinations as network conditions change.
Socket programming uses a client-server model where the client initiates contact with the server to request a service. It uses sockets to allow two processes to communicate by sending and receiving data through the socket. The socket API provides functions to create, bind, listen for, accept, and communicate over sockets. It defines sockets as endpoints for communication between processes running on the same or different devices on a network.
This document discusses unicasting and multicasting in computer networks. It provides details on:
- The key differences between unicasting (one-to-one communication) and multicasting (one-to-many communication), including how routers handle forwarding for each.
- Common applications that use multicasting like audio/video distribution, file sharing, and conferencing.
- Approaches to multicast routing including source-based trees, group-shared trees, and protocols like PIM, CBT, and MBONE tunneling to connect isolated multicast networks.
- Mechanisms used in multicast routing protocols like RPF, pruning/grafting, and IGMP to discover multicast group members
TCP and UDP are transport layer protocols used for data transfer in the OSI model. TCP is connection-oriented, requiring a three-way handshake to establish a connection that maintains data integrity. It guarantees data will reach its destination without duplication but is slower than UDP. UDP is connectionless and used for applications requiring fast transmission like video calls, but does not ensure packet delivery and order. Both protocols add headers to packets with TCP focused on reliability and UDP on speed.
This document discusses IP addressing and classful addressing in TCP/IP networking. It covers the following key points:
- IP addresses are 32-bit addresses that uniquely identify devices on the Internet. They are organized into classes A, B, C, D and E based on the binary pattern of the address.
- Classful addressing allocates address blocks to organizations based on these classes. However, this led to inefficient address usage and rapid depletion of available addresses.
- Subnetting and supernetting were introduced to allow better allocation of addresses within the original classful blocks through the use of subnet and supernet masks. However, classful addressing is now mostly obsolete.
The document discusses the Internet Control Message Protocol (ICMP). ICMP provides error reporting, congestion reporting, and first-hop router redirection. It uses IP to carry its data end-to-end and is considered an integral part of IP. ICMP messages are encapsulated in IP datagrams and are used to report errors in IP datagrams, though some errors may still result in datagrams being dropped without a report. ICMP defines various message types including error messages like destination unreachable and informational messages like echo request and reply.
RTSP is a protocol for controlling streaming media. It uses RTP for data delivery and allows VCR-like controls like play, pause, etc. RTSP sits on top of RTP and handles session setup and control, while RTP handles actual media data transport. RTSP messages are similar to HTTP but maintain state between requests and responses unlike stateless HTTP. RTSP enables streaming media playback by requesting media from a server and receiving it as a continuous stream.
This document discusses the stop-and-wait protocol. It provides flow control by allowing only one frame to be transmitted at a time before waiting for an acknowledgment. However, it does not provide error control. The key aspects are:
1. It is used for unidirectional data transmission over noiseless channels.
2. It only allows one frame to be transmitted at a time before waiting for an acknowledgment, providing flow control but no error control.
3. A disadvantage is that if a frame is lost or corrupted, both the sender and receiver will be stuck indefinitely waiting.
This document discusses several key design issues that occur across multiple layers in computer networks, including addressing, error control, flow control, multiplexing, and routing. Addressing refers to the need for each layer to identify senders and receivers. Error control handles imperfect physical circuits using error detection and correction codes agreed upon by both ends. Flow control deals with assembling and reassembling messages as they are transmitted. Routing selects a path when multiple options exist between source and destination. Multiplexing and demultiplexing improve network systems by combining and separating multiple communication signals.
The selective repeat protocol allows the receiver to accept and buffer frames following a damaged or lost one. Both the sender and receiver maintain a window of outstanding and acceptable sequence numbers. The receiver has a buffer for each sequence number within its fixed window. Whenever a frame arrives within the receiver's window, it is accepted and stored without regard to expected sequence. This protocol has fewer retransmissions than go-back-n but is more complex, as each frame must be acknowledged individually and the receiver may receive frames out of order.
RIP is an interior gateway protocol that uses distance vector routing and the Bellman-Ford algorithm to dynamically adapt to network changes. It works by having each router calculate the distances to reachable networks and share these distances with neighboring routers. However, RIP has issues with slow convergence and count-to-infinity problems when network failures occur. Several techniques are used to address these issues, including hold downs, split horizon, poison reverse updates, and triggered updates.
This document provides an overview of the Transmission Control Protocol (TCP). It discusses TCP services like reliable data delivery and connection-oriented communication. The document explains TCP features such as flow control, error control, and congestion control. It describes TCP segments, the three-way handshake for connection establishment, and the TCP state transition diagram. Examples are provided to illustrate TCP windows, acknowledgments, retransmissions, and timers.
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.
RTP is used for real-time multimedia streaming applications that require timely delivery over UDP. It provides timestamps, sequence numbers, and payload format identification to enable synchronization and detection of packet loss or reordering. The companion protocol RTCP is used for quality of service feedback and synchronization between media streams. Common uses of RTP include simple multicast audio conferences where each participant's audio is sent in chunks with RTP headers, and audio/video conferences where separate RTP sessions are used for each medium.
The document discusses several application layer protocols used in TCP/IP including HTTP, HTTPS, FTP, and Telnet. HTTP is used to access resources on the world wide web over port 80 and is stateless. HTTPS is a secure version of HTTP that encrypts communications over port 443. FTP is used to transfer files between hosts but sends data and passwords in clear text. Telnet allows users to access programs on remote computers.
This document discusses Go-Back-N ARQ, a method for error control in data transmission that uses pipelining and sliding windows. It introduces the concept of using sequence numbers, timers, acknowledgements, and resending frames to improve efficiency over Stop-and-Wait ARQ. Go-Back-N ARQ allows the sender to send multiple frames before receiving ACKs, but requires resending all frames after the lost one. It has higher efficiency than Stop-and-Wait but also has disadvantages related to buffering requirements and wasted bandwidth from resending uncorrupted frames.
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.
The document discusses transport layer protocols TCP and UDP. It provides an overview of process-to-process communication using transport layer protocols. It describes the roles, services, requirements, addressing, encapsulation, multiplexing, and error control functions of the transport layer. It specifically examines TCP and UDP, comparing their connection-oriented and connectionless services, typical applications, and segment/datagram formats.
The document discusses the Transport layer protocols TCP and UDP. It describes TCP as a connection-oriented protocol that provides reliable, ordered delivery of streams of data through mechanisms like sequencing, acknowledgment, flow control and error checking. UDP is described as a simpler connectionless protocol that provides best-effort delivery without checking for errors or lost packets. The key concepts of ports, sockets, multiplexing and demultiplexing are also covered, as well as the header formats and functions of TCP and UDP.
1) Distributed deadlock detection approaches include prevention, avoidance, detection, and resolution. Detection is the primary focus of the chapter.
2) Distributed deadlock detection modifies the bipartite graph strategy to use a wait-for graph (WFG) to model processes requesting resources from other processes. A deadlock exists if there is a directed cycle in the global WFG.
3) Distributed detection requires progress such that all deadlocks are found within a finite time, and safety such that no false detections occur due to network latencies. Approaches include centralized, distributed, and hierarchical control frameworks.
IGMP (Internet Group Management Protocol) allows multicast routers to track group memberships across multicast networks. It has three message types - query, membership report, and leave report. Upon receiving a query, hosts send membership reports to the router to join or leave groups. The router uses these reports to maintain a list of members for each multicast group on that network segment. IGMP messages are encapsulated in IP datagrams and Ethernet frames for transmission.
Distance Vector & Link state Routing AlgorithmMOHIT AGARWAL
1) Each router maintains a routing table containing the outgoing link and distance to reach each destination node. 2) Routers periodically share their routing tables with neighbors so each can update its own table. 3) This allows routers to continuously determine the shortest paths to all destinations as network conditions change.
Socket programming uses a client-server model where the client initiates contact with the server to request a service. It uses sockets to allow two processes to communicate by sending and receiving data through the socket. The socket API provides functions to create, bind, listen for, accept, and communicate over sockets. It defines sockets as endpoints for communication between processes running on the same or different devices on a network.
This document discusses unicasting and multicasting in computer networks. It provides details on:
- The key differences between unicasting (one-to-one communication) and multicasting (one-to-many communication), including how routers handle forwarding for each.
- Common applications that use multicasting like audio/video distribution, file sharing, and conferencing.
- Approaches to multicast routing including source-based trees, group-shared trees, and protocols like PIM, CBT, and MBONE tunneling to connect isolated multicast networks.
- Mechanisms used in multicast routing protocols like RPF, pruning/grafting, and IGMP to discover multicast group members
TCP and UDP are transport layer protocols used for data transfer in the OSI model. TCP is connection-oriented, requiring a three-way handshake to establish a connection that maintains data integrity. It guarantees data will reach its destination without duplication but is slower than UDP. UDP is connectionless and used for applications requiring fast transmission like video calls, but does not ensure packet delivery and order. Both protocols add headers to packets with TCP focused on reliability and UDP on speed.
This document discusses IP addressing and classful addressing in TCP/IP networking. It covers the following key points:
- IP addresses are 32-bit addresses that uniquely identify devices on the Internet. They are organized into classes A, B, C, D and E based on the binary pattern of the address.
- Classful addressing allocates address blocks to organizations based on these classes. However, this led to inefficient address usage and rapid depletion of available addresses.
- Subnetting and supernetting were introduced to allow better allocation of addresses within the original classful blocks through the use of subnet and supernet masks. However, classful addressing is now mostly obsolete.
The document discusses the Internet Control Message Protocol (ICMP). ICMP provides error reporting, congestion reporting, and first-hop router redirection. It uses IP to carry its data end-to-end and is considered an integral part of IP. ICMP messages are encapsulated in IP datagrams and are used to report errors in IP datagrams, though some errors may still result in datagrams being dropped without a report. ICMP defines various message types including error messages like destination unreachable and informational messages like echo request and reply.
RTSP is a protocol for controlling streaming media. It uses RTP for data delivery and allows VCR-like controls like play, pause, etc. RTSP sits on top of RTP and handles session setup and control, while RTP handles actual media data transport. RTSP messages are similar to HTTP but maintain state between requests and responses unlike stateless HTTP. RTSP enables streaming media playback by requesting media from a server and receiving it as a continuous stream.
This document discusses the stop-and-wait protocol. It provides flow control by allowing only one frame to be transmitted at a time before waiting for an acknowledgment. However, it does not provide error control. The key aspects are:
1. It is used for unidirectional data transmission over noiseless channels.
2. It only allows one frame to be transmitted at a time before waiting for an acknowledgment, providing flow control but no error control.
3. A disadvantage is that if a frame is lost or corrupted, both the sender and receiver will be stuck indefinitely waiting.
This document discusses several key design issues that occur across multiple layers in computer networks, including addressing, error control, flow control, multiplexing, and routing. Addressing refers to the need for each layer to identify senders and receivers. Error control handles imperfect physical circuits using error detection and correction codes agreed upon by both ends. Flow control deals with assembling and reassembling messages as they are transmitted. Routing selects a path when multiple options exist between source and destination. Multiplexing and demultiplexing improve network systems by combining and separating multiple communication signals.
The selective repeat protocol allows the receiver to accept and buffer frames following a damaged or lost one. Both the sender and receiver maintain a window of outstanding and acceptable sequence numbers. The receiver has a buffer for each sequence number within its fixed window. Whenever a frame arrives within the receiver's window, it is accepted and stored without regard to expected sequence. This protocol has fewer retransmissions than go-back-n but is more complex, as each frame must be acknowledged individually and the receiver may receive frames out of order.
RIP is an interior gateway protocol that uses distance vector routing and the Bellman-Ford algorithm to dynamically adapt to network changes. It works by having each router calculate the distances to reachable networks and share these distances with neighboring routers. However, RIP has issues with slow convergence and count-to-infinity problems when network failures occur. Several techniques are used to address these issues, including hold downs, split horizon, poison reverse updates, and triggered updates.
This document provides an overview of the Transmission Control Protocol (TCP). It discusses TCP services like reliable data delivery and connection-oriented communication. The document explains TCP features such as flow control, error control, and congestion control. It describes TCP segments, the three-way handshake for connection establishment, and the TCP state transition diagram. Examples are provided to illustrate TCP windows, acknowledgments, retransmissions, and timers.
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.
RTP is used for real-time multimedia streaming applications that require timely delivery over UDP. It provides timestamps, sequence numbers, and payload format identification to enable synchronization and detection of packet loss or reordering. The companion protocol RTCP is used for quality of service feedback and synchronization between media streams. Common uses of RTP include simple multicast audio conferences where each participant's audio is sent in chunks with RTP headers, and audio/video conferences where separate RTP sessions are used for each medium.
The document discusses the Real-Time Transport Protocol (RTP) which provides end-to-end delivery services for delivering audio and video over IP networks. RTP is used with the RTCP to monitor transmission quality. RTP allows synchronization of multiple streams through timestamping and sequencing of packets. It uses UDP for multiplexing and includes fields for payload type, sequence number, timestamp and SSRC for stream identification. Mixers and translators are used to combine or modify RTP streams while maintaining synchronization between sources.
RTP provides end-to-end delivery services for real-time audio and video traffic over UDP. It includes payload identification, sequence numbering, timestamping and delivery monitoring. RTP runs on top of UDP and is designed to support multimedia conferences using IP multicast. RTCP monitors quality of service and conveys information about participants. Profiles and payload formats can customize RTP to support specific applications.
The document discusses several standard and proprietary streaming media protocols. It introduces Real-Time Transport Protocol (RTP) and Real-Time Control Protocol (RTCP) which transport streaming media and provide quality of service reports. It also describes Real Time Streaming Protocol (RTSP) which provides playback controls. Synchronized Multimedia Integration Language (SMIL) is mentioned as an XML language for multimedia content. Major companies like Real, Microsoft, and Apple are noted to use similar but proprietary protocols instead of the standards.
RTP/RTCP are protocols for delivering audio and video over IP networks. RTP is used for media transport and provides sequencing and timing information. RTCP is used for quality of service monitoring through feedback on packet loss and delay. RTP and RTCP typically operate over UDP for real-time delivery without retransmissions. RTCP packets including sender and receiver reports provide synchronization and allow participants to monitor call quality.
The document discusses media VoIP protocols and technologies. It provides an overview of:
- RTP and RTCP protocols for transporting audio and video over IP networks. RTP provides sequencing and time stamping while RTCP provides quality feedback.
- Common audio and video codecs like G.711, H.261, H.263 that are used to compress media.
- Mechanisms for mixing and translating streams to connect networks of different capabilities.
- DTMF signaling which can be carried in-band or out-of-band using SIP INFO.
The transport layer provides process-to-process communication and utilizes three main protocols: UDP, TCP, and SCTP. UDP is a connectionless protocol that does not guarantee delivery, while TCP provides reliable, ordered delivery through a connection-oriented approach. SCTP also provides reliable delivery with the added capability of multiple streams. Key aspects of these protocols include port numbers, packet/segment formatting, and connection establishment handshaking.
This document summarizes information about the Stream Control Transmission Protocol (SCTP), Real-time Transport Protocol (RTP), and RTP Control Protocol (RTCP). SCTP is a transport layer protocol that provides reliable message delivery like TCP with multi-streaming capabilities. RTP is used for delivering audio and video over IP and defines packet formats with timestamps and sequence numbers. RTCP monitors RTP transmission quality and provides feedback on metrics like packet loss.
This document summarizes an assignment submitted by Abhishek Kesharwani on the Stream Control Transmission Protocol (SCTP), Real-time Transport Protocol (RTP), and RTP Control Protocol (RTCP). It describes SCTP as a transport layer protocol that provides reliable message transport like TCP with multi-streaming capabilities. RTP is defined as a standard packet format for delivering audio and video over IP networks, often used with RTCP for quality monitoring and stream synchronization. The document outlines key features of SCTP including multihoming and message ordering, and components of RTP including its data transfer and control protocols.
SCTP is a transport layer protocol that provides reliable transmission of data streams over connectionless networks. It supports multi-homing by allowing endpoints to connect over multiple IP addresses, allowing fault tolerance by switching connections if one path fails. SCTP establishes connections using a four-way handshake and supports ordered and unordered delivery of data chunks. It uses sequence numbers, acknowledgments, and retransmissions to ensure reliable and error-free data transfer. Flow control is implemented using a receiver window and congestion control uses mechanisms similar to TCP.
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.
Overview of transport protocols as alternatives to TCP and UDP.
TCP and UDP are the two transport protocols (OSI layer 4) that are predominantly used by applications in IP based networks.
The properties of TCP and UDP are complementary in that TCP provides many quality of service features that UDP lacks.
Therefore, TCP is mainly used in applications that require a certain level of reliable transport connection while UDP is used when reliability is of secondary importance but speed and simplicity are important.
There are, however, alternatives to TCP and UDP. SCTP (Stream Control Transmission Protocol) was defined some time ago and was meant to eventually replace TCP. It provides the same features as TCP but fixes some of the shortcomings of TCP. Alternatives for UDP exist as well such as Reliable UDP and UDP redundancy.
This document provides an overview of transport layer protocols TCP, UDP, and SCTP. It discusses the history and evolution of TCP, including key developments like congestion control algorithms. UDP is described as a connectionless and unreliable protocol. SCTP is introduced as a protocol developed to transport telephony signaling over IP networks. It addresses limitations of TCP like head-of-line blocking and provides features like multi-homing and message orientation. The document defines SCTP terminology and describes its chunks, states, congestion control approach, and similarities to TCP. In summary, it serves as a high-level introduction to transport protocols with a focus on motivations and capabilities of SCTP.
TCP is a connection-oriented protocol that ensures reliable delivery of data through sequence numbers, acknowledgments, and retransmissions. It has larger headers than UDP but provides reliability. UDP is connectionless and does not guarantee delivery, making it faster but less reliable than TCP. Key applications using TCP include HTTP, FTP, and SMTP, while UDP is used for DNS, VoIP, and streaming applications requiring low latency.
International Journal of Engineering Research and Applications (IJERA) is a team of researchers not publication services or private publications running the journals for monetary benefits, we are association of scientists and academia who focus only on supporting authors who want to publish their work. The articles published in our journal can be accessed online, all the articles will be archived for real time access.
Our journal system primarily aims to bring out the research talent and the works done by sciaentists, academia, engineers, practitioners, scholars, post graduate students of engineering and science. This journal aims to cover the scientific research in a broader sense and not publishing a niche area of research facilitating researchers from various verticals to publish their papers. It is also aimed to provide a platform for the researchers to publish in a shorter of time, enabling them to continue further All articles published are freely available to scientific researchers in the Government agencies,educators and the general public. We are taking serious efforts to promote our journal across the globe in various ways, we are sure that our journal will act as a scientific platform for all researchers to publish their works online.
This document summarizes key concepts from Chapter 3 of the textbook on transport layer protocols:
1. The transport layer provides logical communication between processes running on different hosts, abstracting the underlying network infrastructure. It multiplexes data from multiple sockets and demultiplexes received data to the appropriate socket.
2. UDP and TCP are the main transport protocols in the Internet. UDP is connectionless while TCP provides reliable, connection-oriented data transfer using sequence numbers, acknowledgments, and congestion control.
3. TCP uses congestion control including a congestion window, additive increase/multiplicative decrease, and slow start to dynamically control the sender's transmission rate based on detected packet loss as a signal of
The document provides an overview of several transport layer protocols and concepts:
- TCP is a connection-oriented protocol that provides reliable, ordered delivery through the use of acknowledgments, flow control, and error control. UDP is a simpler connectionless protocol.
- Sockets provide an interface between applications and the network, allowing applications to build distributed client-server systems.
- Congestion control is important to prevent network congestion and ensure fair resource sharing between flows. TCP uses end-to-end congestion control while some protocols use network feedback.
- RTP and RTCP are used for real-time media delivery over UDP, providing sequencing, timing information, and quality monitoring without guarantees.
- SCT
This document discusses TCP performance over mobile ad hoc networks (MANETs). It begins with an overview of TCP and how it was designed for wired networks. In MANETs, TCP faces challenges from node mobility, which can cause network partitions and route changes. It also discusses how lower network layers like the MAC layer and routing protocols can impact TCP. Several solutions are presented to improve TCP for MANETs, including modifying TCP to better handle mobility-related issues and providing it feedback to distinguish route failures from congestion.
Analytical Research of TCP Variants in Terms of Maximum ThroughputIJLT EMAS
This paper is comparative, throughput analysis, for
the TCP variants as for New Reno, Westwood & High Speed,
and it analyzes the outcomes in simulated environment for NS -3
(version 3.25) simulator with reference to multiple varying
network parameters that includes network simulation time,
router bandwidth, varying traffic source counts to observe which
is one of the best TCP variant in different scenarios. Analysis
was done using dumbbell topology to figure out the comparative
maximum throughput of TCP variants. The analysis gives result
as TCP Variant “NewReno” is good when low bandwidth is used,
while TCP Variant “HighS peed” is good in terms of using large
bandwidths in comparison to Westwood. Network traffic flow
was observed in NetAnim tool.
Cosa hanno in comune un mattoncino Lego e la backdoor XZ?Speck&Tech
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What will you get from this session?
1. Insights into integrating generative AI.
2. Understanding how this integration enhances test automation within the UiPath platform
3. Practical demonstrations
4. Exploration of real-world use cases illustrating the benefits of AI-driven test automation for UiPath
Topics covered:
What is generative AI
Test Automation with generative AI and Open AI.
UiPath integration with generative AI
Speaker:
Deepak Rai, Automation Practice Lead, Boundaryless Group and UiPath MVP
2. OBJECTIVE
To get to understand
about the RTCP
protocol flow diagram
It’s working properties Benefit of RTCP
Type of information
stream it works on
Where used?
5. INTRODUCTION
AS WE KNOW THAT IN RTP (REAL-TIME
TRANSFER PROTOCOL) WE CAN ONLY
SEND THE DATA FROM SOURCE TO THE
TERMINALS/USERS BUT CAN’T GET THE
DATA FROM THE USERS.
SO THIS DISADVANTAGE CAN BE
OVERCOME BY USING RTCP (REAL-TIME
TRANSFER CONTROL PROTOCOL) .
6. WHAT IS RTCP?
The messages that can control the transmission
and quality of data (QoS) as well as also allow the
recipients so that they can send feedback to the
source or sources. A protocol designed for this
purpose, which is known as Real-time Transport
Control Protocol (RTCP).
7. HOW RTCP WORKS?
RTCP operates
alongside with the
RTP and share
information with it
but at same time
every RTCP has
different port
– User Datagram
Protocol) to work
independently
from the RTP
.
UDP – it’s just like
transport control
protocol but it
doesn’t provide
acknowledgement of
the sent packets.
RTCP doesn’t use well
known UDP port but
instead of that it uses
temporary port that
must be an odd-
numbered port that
follows port number
selected for RTP
.
9. RTCP is basically based on the periodic transmission of control packets to all participants
in the network being on the fourth layer i.e. transport layer), using the same distribution
process as the data packets.
It is recommended that the fraction of network bandwidth allocated to the RTCP is not
more than 5% and at least 25% of bandwidth is for source reports.
So, the primary function of RTCP is to provide feedback on the quality of the data
distribution.
It specifies report PDUs exchanged between source and destination.
10. CONTINUE…
PDU – Protocol data unit nothing but it gives an abstract
idea of the data packets, but it has lesser overhead than
Although PDU for the transport layer protocol is called
segment
The reports contain statistics such as the number of RTP-
PDUs sent, number of RTP-PDUs lost, inter arrival jitter.
It is used by application to modify sender transmission rate
and diagnostics purposes.
11. SO RTCP IS VERY
USEFUL IN THE
REAL TIME
APPLICATIONS
LIKE VIDEO
CONFERENCING
WHERE WE
ALWAYS WANT
TO KNOW THE
STATUS OF THE
TERMINALS THAT
AS WE HAVE SEEN
IN THE DIAGRAM
THAT EVERY SENDER
WILL SEND RTP AS
WELL AS RTCP, BUT
EVERY TERMINAL OR
DESTINATION
SENDS THE
FEEDBACK
THROUGH RTCP.
MULTICAST
DISTRIBUTION IS
USED JUST LIKE
DATA PACKETS.
12. SYNCHRONIZATION OF STREAMS
Timestamp doesn’t get tied up with the individual video or audio sampling clocks like RTP.
Each sender report in the RTCP contains:-
The timestamp of RTP PDU .
The wall-clock time for when PDU was created.
so receiver use this association to synchronize video and audio playout.
Internetwork
RTP audio
RTCP audio
RTP video
RTP video
13. WHY RTCP?
What happens when there is one sender and many receivers?
RTP can’t do anything in this scenario as it focus on individual
video but RTCP can do this by creating reports linearly with the
number of participants and would match or exceed the amount of
RTP data, but more overhead than useful data.
14. SOLUTION
RTCP attempts to limit its traffic to 5% of the session bandwidth
to ensure it can scale!
RTCP gives 75% of this rate to the receivers; and the remaining
25% to the sender.
15. Suppose one sender, sending video at a rate of 2 Mbps. Then RTCP attempts to limit its traffic to 100
Kbps.
The 75 kbps is equally shared among receivers:
• With R receivers, each receiver gets to send RTCP traffic at 75/R kbps.
Sender gets to send RTCP traffic at 25 kbps.
16. Version: (2 bits) identifies the version of RTP
.
P (Padding): (1 bits) this is used to indicate if there is any extra padding bytes at the end of the RTP packet. It might be used to fill up a block of certain
size like for block required by an encryption algorithm .
RC ( Reception Report Count ): (5 bits) It is the number of reception report blocks contained in this packet.
PT (Packet Type): (8 bits) It contains a constant to identify RTCP packet type.
Length: ( 16 bits ) It indicates the length of this RTCP packet.
SSRC: (32 bits ) Synchronization source identifier uniquely identifies the source of stream.
17. RTCP packets have a length field in header; aligned to 32
bits --- stackable
Sent in a compound packet of at least 2 RTCP packets
18. Basically RTCP has five types of messages namely :-
• Sender Report
• Receiver’s Report
• Source Description Message
• Bye Message
• Application Specific Message
19. SENDER MESSAGE
Sender report is
sent after a fixed
interval by the
active sender in a
conference to
report transmission
as well as stats of
reception for all
RTP packets
transmitted during
this time period.
After receiving these
RTP messages, receiver
get some help in
synchronization process
and thus it is very
important in audio
video transmission or
for any smoother real
time experience.
It contains current time,
no. of packets sent, no.
of bytes sent, timestamp
20. SSRC (synchronization source) : identify sender
Sender info. block:
• NTP (Network transport protocol) timestamp sent time (wall clock time or roundtrip delay, etc.)
• RTP timestamp: corresponding to NTP but in formats of RTP data packet timestamp; used for intra & inter media
synchronization.
• Sender’s packet count & octet count
Multiple report blocks, each block has
• SSRC_n, fraction lost, number of lost
• Highest sequence number received
• Inter-arrival jitter, LSR (Last SR timestamp ), DLSR (Delay Since Last SR)
21. RECEIVER’S MESSAGE
Passive participants are those that don’t send any RTP packet and for them receiver
report is used to inform the sender and other receivers about the quality of service(QoS).
So basically, we get to know about fraction of packets lost, last sequence no., average
interarrival jitter
22. SOURCE DESCRIPTION MESSAGE
The Source Sends a source description message at every fixed interval to give some extra information
about itself containing details like name of source, its mail id, contact number or source controller.
It contains CNAME (canonical name) unique for every different terminal/user.
23. BYE MESSAGE
When ever a stream is to be in completion or to shut down a
stream there should be a bye message and this last time shut
down message is bye report/message.
It is sent by source to announcing for leaving the conference.
This is direct announcement for other sources about absence of
other source.
24. APPLICATION
SPECIFIC
MESSAGE
Now if we want to make our application more
extensible then RTCP allows application-specific
RTCP packets which is introduced by RFC 3611.
This can be used to extend the type of application.
25. ROUND TRIP TIME CALCULATION
SR packet contains: NTP (= t1)
RR packet contains: Last SR timestamp (LSR = t1), Delay Since Last SR (DLSR=t3-t2)
Roundtrip time = t4 - t3 + t2 - t1 = t4 – (t3-t2) –t1
= t4 – DLSR - LSR
sender
receiver
SR RR
DLSR
t1
t2 t3
t4
26. So basically, RTCP is used to get more features of
multimedia and efficiently share the bandwidth for the
better accessibility.
27. SUMMARY
We got to know about the flow of RTCP
how it act as a companion to RTP
What RTCP means
Features of RTCP
Types of report it sends
Application