1- Protocolos de transporte con QoS Clases de aplicaciones multimedia Redes basadas en IP y QoS Gestión de los recursos: IntServ vs DiffServ RSVP RTP/RTCP: Transporte de flujos multimedia RTSP: Control de sesión y localización de medios Multicasting Computer Networking: A Top Down Approach Featuring the Internet, 3rd edition. Jim Kurose, Keith Ross Addison-Wesley, July 2004. Thanks to : RADCOM technologies H. Shulzrinne Paul. E. Jones (from packetizer.com)

Definition of multimedia What is multimedia? Definition of multimedia Hard to find a clear-cut definition In general, multimedia is an integration of text, graphics, still and moving images, animation, sounds, and any other medium where every type of information can be represented, stored, transmitted and processed digitally Characteristics of multimedia Digital – key concept Integration of multiple media type, usually including video or/and audio May be interactive or non-interactive

Text, Graphics, image, video, animation, sound, etc. Various Media Types Text, Graphics, image, video, animation, sound, etc. Classifications of various media types Captured vs. synthesized media Captured media (natural) : information captured from the real world Example: still image, video, audio Synthesized media (artificial) : information synthesize by the computer Example: text, graphics, animation Discrete vs. continuous media Discrete media: space-based, media involve the space dimension only Text, Image, Graphics Continuous media: time-based, media involves both the space and the time dimension Video, Sound, Animation

Classification of Media Type Sound Video Image Animation Text Graphics Captured From real world Synthesized By computer Discrete Continuous

Text Plain text Rich text Unformatted Characters coded in binary form ASCII code All characters have the same style and font Rich text Formatted Contains format information besides codes for characters No predominant standards Characters of various size, shape and style, e.g. bold, colorful

Plain Text vs. Rich Text An example of Plain text Example of Rich text

Revisable document that retains structural information Graphics Revisable document that retains structural information Consists of objects such as lines, curves, circles, etc Usually generated by graphic editor of computer programs Example of graphics (FIG file)

2D matrix consisting of pixels Images 2D matrix consisting of pixels Pixel—smallest element of resolution of the image One pixel is represented by a number of bits Pixel depth– the number of bits available to code the pixel Have no structural information Two categories: scanned vs. synthesized still image Computer software Capture and A/D conversion Digital still image Synthesized image Scanned Camera

Images (cont.) Examples of images Binary image – pixel depth 1 Gray-scale – pixel depth 8 Color image – pixel depth 24 Gray-scale image color image Binary image

Video – moving images or moving pictures Video vs. Animation Both images and graphics can be displayed as a succession of view which create an impression of movement Video – moving images or moving pictures Captured or Synthesized Consists of a series of bitmap images Each image is called a frame Frame rate: the speed to playback the video (frame per second) Animation – moving graphics Generated by computer program (animation authoring tools) Consists of a set of objects The movements of the objects are calculated and the view is updated at playback

Sound 1-D time-based signal Speech vs. non-speech sound Speech – supports spoken language and has a semantic content Non-speech – does not convey semantics in general Natural vs. structured sound Natural sound – Recorded/generated sound wave represented as digital signal Example: Audio in CD, WAV files Structured sound – Synthesize sound in a symbolic way Example: MIDI file

Local vs. networked multimedia Local: storage and presentation of multimedia information in standalone computers Sample applications: DVD Networked: involve transmission and distribution of multimedia information on the network Sample applications: videoconferencing, web video broadcasting, multimedia Email, etc. Image server A scenario of multimedia networking Internet Video server

Consideration of Networked Multimedia Requirements of multimedia applications on the network Typically delay sensitive end-to-end delay delay jitter: Jitter is the variability of packet delays within the same packet stream Quality requirement Satisfactory quality of media presentation Synchronization requirement Continuous requirement (no jerky video/audio) Can tolerant some degree of information loss

Technologies of Multimedia Networking Challenges of multimedia networking Conflict between media size and bandwidth limit of the network Conflict between the user requirement of multimedia application and the best-effort network How to meet different requirements of different users? Media compression – reduce the data volume Address the 1st challenge Image compression Video compression Audio compression Multimedia transmission technology Address the 2nd and 3rd challenges Protocols for real-time transmission Rate / congestion control Error control

Multimedia Networking Systems Live media transmission system Capture, compress, and transmit the media on the fly (example?) Send stored media across the network Media is pre-compressed and stored at the server. This system delivers the stored media to one or multiple receivers. (example?) Differences between the two systems For live media delivery: Real-time media capture, need hardware support Real-time compression– speed is important Compression procedure can be adjusted based on network conditions For stored media delivery Offline compression – better compression result is important Compression can not be adjusted during transmission

Classes of multimedia applications Streaming stored audio and video Streaming live audio and video Real-time interactive audio and video

Streaming Stored Multimedia: What is it? 100% streaming: at this time, client playing out early part of video, while server still sending later part of video 3. video received, played out at client Cumulative data 2. video sent 1. video recorded network delay time

Streaming vs. Download of Stored Multimedia Content Download: Receive entire content before playback begins High “start-up” delay as media file can be large ~ 4GB for a 2 hour MPEG II movie Streaming: Play the media file while it is being received Reasonable “start-up” delays Reception Rate >= playback rate. Why?

Streaming Stored Multimedia: Interactivity VCR-like functionality: client can pause, rewind, FF, push slider bar 10 sec initial delay OK 1-2 sec until command effect OK RTSP often used (more later) timing constraint for still-to-be transmitted data: in time for playout

Streaming Multimedia: Client Buffering constant bit rate video transmission variable network delay client video reception constant bit rate video playout at client client playout delay Cumulative data buffered video time Client-side buffering, playout delay compensate for network-added delay, delay jitter

Streaming Multimedia: Client Buffering Client-side buffering, playout delay compensate for network-added delay, delay jitter constant drain rate, d variable fill rate, x(t) buffered video

Interactive, Real-Time Multimedia applications: IP telephony, video conference, distributed interactive worlds end-end delay requirements: audio: < 150 msec good, < 400 msec OK includes application-level (packetization) and network delays higher delays noticeable, impair interactivity session initialization how does callee advertise its IP address, port number, encoding algorithms?

Internet multimedia: simplest approach audio or video stored in file files transferred as HTTP object received in entirety at client then passed to player audio, video not streamed: no, “pipelining,” long delays until playout!

Progressive Download browser GETs metafile browser launches player, passing metafile player contacts server server downloads audio/video to player

Streaming from a streaming server This architecture allows for non-HTTP protocol between server and media player Can also use UDP instead of TCP.

Multimedia Over Today’s Internet TCP/UDP/IP: “best-effort service” no guarantees on delay, loss But multimedia apps requires QoS and level of performance to be effective! Today’s Internet multimedia applications use application-level techniques to mitigate (as best possible) effects of delay, loss

Streaming Multimedia: UDP or TCP? server sends at rate appropriate for client (oblivious to network congestion!) often send rate = encoding rate = constant rate then, fill rate = constant rate - packet loss short playout delay (2-5 seconds) to compensate for network delay jitter error recover: time permitting TCP send at maximum possible rate under TCP fill rate fluctuates due to TCP congestion control larger playout delay: smooth TCP delivery rate HTTP/TCP passes more easily through firewalls

1- Protocolos de transporte con QoS. Clases de aplicaciones multimedia Redes basadas en IP y QoS Gestión de los recursos: IntServ vs DiffServ RSVP RTP/RTCP: Transporte de flujos multimedia RTSP: Control de sesión y localización de medios Multicasting Thanks to : RADCOM technologies H. Shulzrinne Paul. E. Jones (from packetizer.com)

Productividad (Throughput) Requisitos de red. Se definen 3 parámetros críticos que la red debería suministrar a las aplicaciones multimedia: Productividad (Throughput) Número de bits que la red es capaz de entregar por unidad de tiempo (tráfico soportado). CBR (streams de audio y vídeo sin comprimir) VBR (ídem comprimido) Ráfagas (Peak Bit Rate y Mean Bit Rate) Retardo de tránsito (Transit delay) Retardo extremo-a-extremo Retardo de acceso de tránsito Retardo de transmisión Mensaje listo para envío Envío del primer bit del mensaje Primer bit del mensaje recibido Ultimo bit del mensaje recibido Mensaje listo para recepción

Varianza del retardo (Jitter) Requisitos de red (II). Varianza del retardo (Jitter) Define la variabilidad del retardo de una red. Jitter físico (redes de conmutación de circuito) Suele ser muy pequeño (ns) LAN jitter (Ethernet, FDDI). Jitter físico + tiempo de acceso al medio. Redes WAN de conmutación de paquete (IP, X.25, FR) Jitter físico + tiempo de acceso + retardo de conmutación de paquete en conmutadores de la red. 1 2 3 D1 D2 = D1 D3 > D1 t Emisor Receptor

Internet y las aplicaciones multimedia ¿Qué podemos añadir a IP para soportar los requerimientos de las aplicaciones multimedia? Técnicas de ecualización de retardos (buffering) Protocolos de transporte que se ajusten mejor a las necesidades de las aplicaciones multimedia: RTP (Real-Time Transport Protocol) RFC 1889. RTSP (Real-Time Streaming Protocol) RFC 2326. Técnicas de control de admisión y reserva de recursos (QoS) RSVP (Resource reSerVation Protocol) RFC 2205 Arquitecturas y protocolos específicos: Protocolos SIP (RFC 2543), SDP (RFC 2327), SAP (RFC 2974), etc.. ITU H.323

Slide thanks to Henning Schulzrinne Internet Protocols Slide thanks to Henning Schulzrinne

Multimedia, Quality of Service: What is it? Multimedia applications: network audio and video (“continuous media”) network provides application with level of performance needed for application to function. QoS

Improving QOS in IP Networks Thus far: “making the best of best effort” Future: next generation Internet with QoS guarantees RSVP: signaling for resource reservations Differentiated Services: differential guarantees Integrated Services: firm guarantees simple model for sharing and congestion studies:

Principles for QOS Guarantees Example: 1Mbps IPphone, FTP share 1.5 Mbps link. bursts of FTP can congest router, cause audio loss want to give priority to audio over FTP Principle 1 packet marking needed for router to distinguish between different classes; and new router policy to treat packets accordingly

Principles for QOS Guarantees (more) what if applications misbehave (audio sends higher than declared rate) policing: force source adherence to bandwidth allocations marking and policing at network edge: similar to ATM UNI (User Network Interface) Principle 2 provide protection (isolation) for one class from others

Principles for QOS Guarantees (more) Allocating fixed (non-sharable) bandwidth to flow: inefficient use of bandwidth if flows doesn’t use its allocation Principle 3 While providing isolation, it is desirable to use resources as efficiently as possible

Principles for QOS Guarantees (more) Basic fact of life: can not support traffic demands beyond link capacity Principle 4 Call Admission: flow declares its needs, network may block call (e.g., busy signal) if it cannot meet needs

1- Protocolos de transporte con QoS. Clases de aplicaciones multimedia Redes basadas en IP y QoS Gestión de los recursos: IntServ vs DiffServ RSVP RTP/RTCP: Transporte de flujos multimedia RTSP: Control de sesión y localización de medios Multicasting Thanks to : RADCOM technologies H. Shulzrinne Paul. E. Jones (from packetizer.com)

Scheduling And Policing Mechanisms scheduling: choose next packet to send on link FIFO (first in first out) scheduling: send in order of arrival to queue discard policy: if packet arrives to full queue: who to discard? Tail drop: drop arriving packet priority: drop/remove on priority basis random: drop/remove randomly

Scheduling Policies: more Priority scheduling: transmit highest priority queued packet multiple classes, with different priorities class may depend on marking or other header info, e.g. IP source/dest, port numbers, etc..

Scheduling Policies: still more round robin scheduling: multiple classes cyclically scan class queues, serving one from each class (if available)

Scheduling Policies: still more Weighted Fair Queuing: generalized Round Robin each class gets weighted amount of service in each cycle

Goal: limit traffic to not exceed declared parameters Policing Mechanisms Goal: limit traffic to not exceed declared parameters Three common-used criteria: (Long term) Average Rate: how many pkts can be sent per unit time (in the long run) crucial question: what is the interval length: 100 packets per sec or 6000 packets per min have same average! Peak Rate: e.g., 6000 pkts per min. (ppm) avg.; 1500 pps peak rate (Max.) Burst Size: max. number of pkts sent consecutively (with no intervening idle)

Policing Mechanisms Token Bucket: limit input to specified Burst Size and Average Rate. bucket can hold b tokens tokens generated at rate r token/sec unless bucket full over interval of length t: number of packets admitted less than or equal to (r t + b).

Policing Mechanisms (more) token bucket, WFQ combine to provide guaranteed upper bound on delay, i.e., QoS guarantee! WFQ token rate, r bucket size, b per-flow rate, R D = b/R max arriving traffic

IETF Integrated Services architecture for providing QOS guarantees in IP networks for individual application sessions resource reservation: routers maintain state info of allocated resources, QoS req’s admit/deny new call setup requests: Question: can newly arriving flow be admitted with performance guarantees while not violated QoS guarantees made to already admitted flows?

Intserv: QoS guarantee scenario Resource reservation call setup, signaling (RSVP) traffic, QoS declaration per-element admission control request/ reply QoS-sensitive scheduling (e.g., WFQ)

Arriving session must : declare its QOS requirement Call Admission Arriving session must : declare its QOS requirement R-spec: defines the QOS being requested characterize traffic it will send into network T-spec: defines traffic characteristics signaling protocol: needed to carry R-spec and T-spec to routers (where reservation is required) RSVP

Intserv QoS: Service models [RFC2211, RFC2212] Guaranteed service: worst case traffic arrival: leaky-bucket-policed source simple (mathematically provable) bound on delay [Parekh 1992, Cruz 1988] Controlled load service: "a quality of service closely approximating the QoS that same flow would receive from an unloaded network element." WFQ token rate, r bucket size, b per-flow rate, R D = b/R max arriving traffic

IETF Differentiated Services Concerns with Intserv: Scalability: signaling, maintaining per-flow router state difficult with large number of flows Flexible Service Models: Intserv has only two classes. Also want “qualitative” service classes “behaves like a wire” relative service distinction: Platinum, Gold, Silver Diffserv approach: simple functions in network core, relatively complex functions at edge routers (or hosts) Don’t define service classes, provide functional components to build service classes

Diffserv Architecture Edge router: per-flow traffic management marks packets as in-profile and out-profile r b marking scheduling . Core router: per class traffic management buffering and scheduling based on marking at edge preference given to in-profile packets Assured Forwarding

Edge-router Packet Marking profile: pre-negotiated rate A, bucket size B packet marking at edge based on per-flow profile Rate A B User packets Possible usage of marking: class-based marking: packets of different classes marked differently intra-class marking: conforming portion of flow marked differently than non-conforming one

Classification and Conditioning Packet is marked in the Type of Service (TOS) in IPv4, and Traffic Class in IPv6 6 bits used for Differentiated Service Code Point (DSCP) and determine PHB that the packet will receive 2 bits are currently unused

Classification and Conditioning may be desirable to limit traffic injection rate of some class: user declares traffic profile (e.g., rate, burst size) traffic metered, shaped if non-conforming

Forwarding (PHB) PHB result in a different observable (measurable) forwarding performance behavior PHB does not specify what mechanisms to use to ensure required PHB performance behavior Examples: Class A gets x% of outgoing link bandwidth over time intervals of a specified length Class A packets leave first before packets from class B

Assured Forwarding: 4 classes of traffic Forwarding (PHB) PHBs being developed: Expedited Forwarding: pkt departure rate of a class equals or exceeds specified rate logical link with a minimum guaranteed rate Assured Forwarding: 4 classes of traffic each guaranteed minimum amount of bandwidth each with three drop preference partitions

1- Protocolos de transporte multimedia. Clases de aplicaciones multimedia Redes basadas en IP y QoS Gestión de los recursos: IntServ vs DiffServ RSVP RTP/RTCP: Transporte de flujos multimedia RTSP: Control de sesión y localización de medios Multicasting

Signaling in the Internet no network signaling protocols in initial IP design connectionless (stateless) forwarding by IP routers best effort service + = New requirement: reserve resources along end-to-end path (end system, routers) for QoS for multimedia applications RSVP: Resource Reservation Protocol [RFC 2205] “ … allow users to communicate requirements to network in robust and efficient way.” i.e., signaling ! earlier Internet Signaling protocol: ST-II [RFC 1819]

RSVP Design Goals accommodate heterogeneous receivers (different bandwidth along paths) accommodate different applications with different resource requirements make multicast a first class service, with adaptation to multicast group membership leverage existing multicast/unicast routing, with adaptation to changes in underlying unicast, multicast routes control protocol overhead to grow (at worst) linear in # receivers modular design for heterogeneous underlying technologies

specify how resources are to be reserved RSVP: does not… specify how resources are to be reserved rather: a mechanism for communicating needs determine routes packets will take that’s the job of routing protocols signaling decoupled from routing interact with forwarding of packets separation of control (signaling) and data (forwarding) planes

RSVP: overview of operation senders, receiver join a multicast group done outside of RSVP senders need not join group sender-to-network signaling path message: make sender presence known to routers path teardown: delete sender’s path state from routers receiver-to-network signaling reservation message: reserve resources from sender(s) to receiver reservation teardown: remove receiver reservations network-to-end-system signaling path error reservation error

Call Admission Session must first declare its QOS requirement and characterize the traffic it will send through the network R-spec: defines the QOS being requested T-spec: defines the traffic characteristics A signaling protocol is needed to carry the R-spec and T-spec to the routers where reservation is required; RSVP is a leading candidate for such signaling protocol

A token bucket specification RSVP request (T-Spec) A token bucket specification bucket size, b token rate, r the packet is transmitted onward only if the number of tokens in the bucket is at least as large as the packet peak rate, p p > r maximum packet size, M minimum policed unit, m All packets less than m bytes are considered to be m bytes Reduces the overhead to process each packet Bound the bandwidth overhead of link-level headers

Call Admission Call Admission: routers will admit calls based on their R-spec and T-spec and base on the current resource allocated at the routers to other calls.

Integrated Services: Classes Guaranteed QOS: this class is provided with firm bounds on queuing delay at a router; envisioned for hard real-time applications that are highly sensitive to end-to-end delay expectation and variance Controlled Load: this class is provided a QOS closely approximating that provided by an unloaded router; envisioned for today’s IP network real-time applications which perform well in an unloaded network

An indication of the QoS control service requested R-spec An indication of the QoS control service requested Controlled-load service and Guaranteed service For Controlled-load service Simply a Tspec For Guaranteed service A Rate (R) term, the bandwidth required R  r, extra bandwidth will reduce queuing delays A Slack (S) term The difference between the desired delay and the delay that would be achieved if rate R were used With a zero slack term, each router along the path must reserve R bandwidth A nonzero slack term offers the individual routers greater flexibility in making their local reservation Number decreased by routers on the path.

QoS Routing: Multiple constraints A request specifies the desired QoS requirements e.g., BW, Delay, Jitter, packet loss, path reliability etc Two type of constraints: Additive: e.g., delay Maximum (or Minimum): e.g., Bandwidth Task Find a (min cost) path which satisfies the constraints if no feasible path found, reject the connection

Path msgs: RSVP sender-to-network signaling path message contents: address: unicast destination, or multicast group flowspec: bandwidth requirements spec. filter flag: if yes, record identities of upstream senders (to allow packets filtering by source) previous hop: upstream router/host ID refresh time: time until this info times out path message: communicates sender info, and reverse-path-to-sender routing info later upstream forwarding of receiver reservations

RSVP: simple audio conference H1, H2, H3, H4, H5 both senders and receivers multicast group m1 no filtering: packets from any sender forwarded audio rate: b only one multicast routing tree possible H3 H2 R1 R2 R3 H4 H1 H5

RSVP: building up path state H1, …, H5 all send path messages on m1: (address=m1, Tspec=b, filter-spec=no-filter,refresh=100) Suppose H1 sends first path message in out L1 L2 L6 in out L3 L7 L4 m1: m1: in out L5 L7 L6 m1: H2 H3 L3 L2 R1 L6 L7 R2 R3 L4 H4 L1 L5 H1 H5

RSVP: building up path state next, H5 sends path message, creating more state in routers in out L1 L2 L6 L6 in out L3 L7 L4 m1: m1: L1 in out L5 L5 L7 L6 m1: L6 H2 H3 L3 L2 R1 L6 L7 R2 R3 L4 H4 L1 L5 H1 H5

RSVP: building up path state H2, H3, H5 send path msgs, completing path state tables in out L1 L2 L6 L2 L6 in out L3 L7 L4 L3 L4 m1: m1: L1 L7 in out L5 L5 L7 L6 L7 m1: L6 H2 H3 L3 L2 R1 L6 L7 R2 R3 L4 H4 L1 L5 H1 H5

reservation msgs: receiver-to-network signaling reservation message contents: desired bandwidth: filter type: no filter: any packets address to multicast group can use reservation fixed filter: only packets from specific set of senders can use reservation dynamic filter: senders who’s packets can be forwarded across link will change (by receiver choice) over time. filter spec reservations flow upstream from receiver-to-senders, reserving resources, creating additional, receiver-related state at routers

RSVP: receiver reservation example 1 H1 wants to receive audio from all other senders H1 reservation msg flows uptree to sources H1 only reserves enough bandwidth for 1 audio stream reservation is of type “no filter” – any sender can use reserved bandwidth H2 H3 L3 L2 R1 L6 L7 R2 R3 L4 H4 L1 L5 H1 H5

RSVP: receiver reservation example 1 H1 reservation msgs flows uptree to sources routers, hosts reserve bandwidth b needed on downstream links towards H1 L6 in out L3 L4 L7 in out L1 L2 m1: m1: L1 (b) L2 L6 L3 L4 L7 (b) in out L5 L6 L7 m1: L5 L6 (b) L7 H2 H3 b b L3 L2 b b b b R1 L6 L7 R2 R3 L4 b H4 L1 L5 H1 H5

RSVP: receiver reservation example 1 (more) next, H2 makes no-filter reservation for bandwidth b H2 forwards to R1, R1 forwards to H1 and R2 (?) R2 takes no action, since b already reserved on L6 L6 in out L3 L4 L7 in out L1 L2 m1: m1: L1 (b) L2 (b) L6 L3 L4 L7 (b) in out L5 L6 L7 m1: L5 L6 (b) L7 H2 H3 b b b L3 L2 b b b b R1 L6 L7 R2 R3 L4 b b H4 L1 L5 H1 H5

RSVP: receiver reservation: issues What if multiple senders (e.g., H3, H4, H5) over link (e.g., L6)? arbitrary interleaving of packets L6 flow policed by leaky bucket: if H3+H4+H5 sending rate exceeds b, packet loss will occur L6 in out L3 L4 L7 in out L1 L2 m1: m1: L1 (b) L2 (b) L6 L3 L4 L7 (b) in out L5 L6 L7 m1: L5 L6 (b) L7 H2 H3 b b b L3 L2 b b b b R1 L6 L7 R2 R3 L4 b b H4 L1 L5 H1 H5

H2 will want to receive from H4 (only) RSVP: example 2 H1, H4 are only senders send path messages as before, indicating filtered reservation Routers store upstream senders for each upstream link H2 will want to receive from H4 (only) H2 H2 H3 H3 L3 L3 L2 L2 R1 L6 L7 R2 R3 L4 H4 L1 H1

RSVP: example 2 H1, H4 are only senders send path messages as before, indicating filtered reservation in out L1, L6 L3(H4-via-H4 ; H1-via-R3 ) L4(H1-via-R2 ) L7(H4-via-H4 ) in out L4, L7 L2(H1-via-H1 ; H4-via-R2 ) L6(H1-via-H1 ) L1(H4-via-R2 ) H2 H2 H3 H3 R2 L3 L3 L2 L2 R1 L6 L7 R3 L4 H4 L1 H1 in out L6, L7 L6(H4-via-R3 ) L7(H1-via-R1 )

receiver H2 sends reservation message for source H4 at bandwidth b RSVP: example 2 receiver H2 sends reservation message for source H4 at bandwidth b propagated upstream towards H4, reserving b in out L1, L6 L3(H4-via-H4 ; H1-via-R2 ) L4(H1-via-R2 ) L7(H4-via-H4 ) in out L4, L7 L2(H1-via-H1 ;H4-via-R2 ) L6(H1-via-H1 ) L1(H4-via-R2 ) (b) (b) H2 H2 H3 H3 b R2 L3 L3 L2 L2 b b b R1 L6 L7 R3 L4 H4 L1 L1 H1 in out L6, L7 L6(H4-via-R3 ) L7(H1-via-R1 ) (b)

RSVP: soft-state senders periodically resend path msgs to refresh (maintain) state receivers periodically resend resv msgs to refresh (maintain) state path and resv msgs have TTL field, specifying refresh interval in out L1, L6 L3(H4-via-H4 ; H1-via-R3 ) L4(H1-via-R2 ) L7(H4-via-H4 ) in out L4, L7 L2(H1-via-H1 ;H4-via-R2 ) L6(H1-via-H1 ) L1(H4-via-R2 ) (b) (b) H2 H2 H3 H3 b R2 L3 L3 L2 L2 b b b R1 L6 L7 R3 L4 H4 L1 L1 H1 in out L6, L7 L6(H4-via-R3 ) L7(H1-via-R1 ) (b)

RSVP: soft-state gone fishing! suppose H4 (sender) leaves without performing teardown eventually state in routers will timeout and disappear! in out L1, L6 L3(H4-via-H4 ; H1-via-R3 ) L4(H1-via-R2 ) L7(H4-via-H4 ) in out L4, L7 L2(H1-via-H1 ;H4-via-R2 ) L6(H1-via-H1 ) L1(H4-via-R2 ) (b) (b) H2 H2 H3 H3 b R2 L3 L3 L2 L2 b b b R1 L6 L7 gone fishing! R3 L4 H4 L1 L1 H1 in out L6, L7 L6(H4-via-R3 ) L7(H1-via-R1 ) (b)

1- Protocolos de transporte multimedia. Clases de aplicaciones multimedia Redes basadas en IP y QoS Gestión de los recursos: IntServ vs DiffServ RSVP RTP/RTCP: Transporte de flujos multimedia RTSP: Control de sesión y localización de medios Multicasting

RTP (Real-time Transport Protocol) Se basa en el concepto de sesión: la asociación entre un conjunto de aplicaciones que se comunican usando RTP Una sesión es identificada por: Una dirección IP multicast Dos puertos: Uno para los datos y otro para control (RTCP) Un participante (participant) puede ser una máquina o un usuario que participa en una sesión Cada media distinto es trasmitido usando una sesión diferente. Por ejemplo, si se quisiera transmitir audio y vídeo harían falta dos sesiones separadas  Esto permite a un participante solamente ver o solamente oír

RTP (Real-time Transport Protocol) Audio-conferencia con multicast y RTP Sesión de audio: Una dirección multicast y dos puertos Datos de audio y mensajes de control RTCP. Existirá (al menos) una fuente de audio que enviará un stream de segmentos de audio pequeños (20 ms) utilizando UDP. A cada segmento se le asigna una cabecera RTP La cabecera RTP indica el tipo de codificación (PCM, ADPCM, LPC, etc.) Número de secuencia y fechado de los datos. Control de conferencia (RTCP): Número e identificación de participantes en un instante dado. Información acerca de cómo se recibe el audio. Audio y Vídeo conferencia con multicast y RTP Si se utilizan los dos medios, se debe crear una sesión RTP independiente para cada uno de ellos. Una dirección multicast y 2 puertos por cada sesión. Existencia de participantes que reciban sólo uno de los medios. Temporización independiente de audio y vídeo.

RTP: Mezcladores y traductores Mezcladores (Mixers). Permiten que canales con un BW bajo puedan participar en una conferencia. El mixer re-sincroniza los paquetes y hace todas las conversiones necesarias para cada tipo de canal. Traductores (Translators). Permiten conectar sitios que no tienen acceso multicast (p.ej. que están en una sub-red protegida por un firewall)

RTP: Formato de mensaje (I) 32 bits V P X CC M PT Sequence number Timestamp Synchronization Source (SSRC) ID Contributing Source (CSRC) ID V: versión; actualmente es la 2 P: si está a 1 el paquete tiene bytes de relleno (padding) X: presencia de una extensión de la cabecera

RTP: Formato de mensaje (II) 32 bits V P X CC M PT Sequence number Timestamp Synchronization Source (SSRC) ID Contributing Source (CSRC) ID CC: Identifica el número de CSRC que contribuyen a los datos M: Marca (definida según el perfil) PT: Tipo de datos (según perfil)

RTP: Formato de mensaje (III) 32 bits V P X CC M PT Sequence number Timestamp Synchronization Source (SSRC) ID Contributing Source (CSRC) ID Sequence number: Establece el orden de los paquetes Timestamp: Instante de captura del primer octeto del campo de datos SSRC: Identifica la fuente de sincronización CSRC: Fuentes que contribuyen

RTP header definition /* * RTP data header */ typedef struct { unsigned int version:2; unsigned int p:1; unsigned int x:1; unsigned int cc:4; unsigned int m:1; unsigned int pt:7; u_int16 seq; u_int32 ts; u_int32 ssrc; u_int32 csrc[1]; } rtp_hdr_t;

RTP y las aplicaciones La especificación de RTP para una aplicación particular va acompañada de: Un perfil (profile) que defina un conjunto de códigos para los tipos de datos transportados (payload) El formato de transporte de cada uno de los tipos de datos previstos Ej.: RFC 1890 para audio y vídeo PT encoding audio/video clock rate channels name (A/V) (Hz) (audio) ______________________________________________ 0 PCMU A 8000 1 1 1016 A 8000 1 2 G721 A 8000 1 3 GSM A 8000 1 ... 34-71 unassigned ? 72-76 reserved N/A N/A N/A 77-95 unassigned ? 96-127 dynamic ? PCMU Corresponde a la recomendación CCITT/ITU-T G.711. El audio se codifica con 8 bits por muestra, después de una cuantificación logarítmica. PCMU es la codificación que se utiliza en Internet para un media de tipo audio/basic.

RTCP (RTP Control Protocol) RTCP se basa en envíos periódicos de paquetes de control a los participantes de una sesión RTP Permite realizar una realimentación de la calidad de recepción de los datos (estadísticas). Los paquetes de control siempre llevan la identificación de la fuente RTP: CNAME Asociar más de una sesión a un mismo fuente (sincronización). El envío de estos paquetes debe ser controlado por cada participante (sistema ampliable). Control de sesión (opcional) Información adicional de cada participante. Entrada y salida de participantes en las sesión. Negociación de parámetros y formatos.

RTCP (RTP Control Protocol) RTCP permite controlar el trafico que no se auto-limita (p.ej cuando el número de fuentes aumenta) Para ello se define el ancho de banda de la sesión. RTCP se reserva el 5% (bwRTCP) A cada fuente se le asigna 1/4 de bwRTCP El intervalo entre cada paquete RTCP es > 5 sec

RTCP (RTP Control Protocol) Formato de un paquete RTCP: Existen distintos tipos de paquetes RTCP: SR (Sender Report): Informes estadísticos de transmisión y recepción de los elementos activos en la sesión. RR (Receiver Report): Informes estadísticos de recepción en los receptores. SDES (Source Description): Información del participante (CNAME, e-mail, etc). BYE: Salida de la sesión. APP: Mensajes específicos de la aplicación. Cada paquete RTCP tiene su propio formato. Su tamaño debe ser múltiplo de 32 bits (padding). Se pueden concatenar varios paquetes RTCP en uno (compound RTCP packet).

RTCP: Mensajes SR V P RC PT=SR=200 Longitud SSRC del sender 32 bits NTP timestamp msw NTP timestamp lsw RTP timestamp Contador de los paquetes del sender Contador de los bytes del sender SSRC_1 Frac perd Total paquetes perdidos Num sec más alto recibido Jitter de inter-llegada Retraso del último SR (LSR) Ultimo SR (LSR) Report block 1 Sender info cabecera

RTCP: Cálculo del Jitter Es una estimación de la variancia del tiempo de inter-llegada de los paquetes RTP Si  RTP timestamp del paquete i Ri  Instante de llegada del paquete i

1- Protocolos de transporte multimedia. Clases de aplicaciones multimedia Redes basadas en IP y QoS Gestión de los recursos: IntServ vs DiffServ RSVP RTP/RTCP: Transporte de flujos multimedia RTSP: Control de sesión y localización de medios Multicasting

Real-Time Streaming Protocol RFC 2326 Tiene la función de un “mando a distancia por la red” para servidores multimedia Permite establecer y controlar uno o más flujos de datos sincronizados NO existe el concepto de conexión RTSP sino de sesión RTSP Además, una sesión RTSP no tiene relación con ninguna conexión especifica de nivel transporte (p.ej. TCP o UDP) Los flujos de datos no tienen por que utilizar RTP Está basado en HTTP/1.1 Diferencias importantes: No es stateless Los clientes y servidores pueden generar peticiones

Terminología RTSP Conferencia Media stream Presentación: Una instancia única de un medio continuo: Un stream audio, Un stream vídeo Una “whiteboard” Presentación: Es el conjunto de uno o más streams, que son vistos por el usuario como un conjunto integrado Voz del público Imagen del conferenciante Imagen del público Imagen de las transparencias Voz del conferenciante

RTSP: Ejemplo de una sesión HTTP GET Web server descripción de la sesión SETUP Cliente PLAY Media server RTP audio RTP vídeo RTCP PAUSE TEARDOWN

RTSP: Comandos de petición Request = Request-Line ; *( general-header | request-header | entity-header ) CRLF [ message-body ] Request-Line = Method SP Request-URI SP RTSP-Version CRLF Method = "DESCRIBE“ | "ANNOUNCE" | "GET_PARAMETER" | "OPTIONS“ | "PAUSE" | "PLAY" | "RECORD" | "REDIRECT" | "SETUP" | "SET_PARAMETER" | "TEARDOWN" | extension-method extension-method = token Request-URI = "*" | absolute_URI RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT

RTSP: Mensajes de respuesta Response = Status-Line ; *( general-header | response-header | entity-header ) CRLF [ message-body ] Status-Line = RTSP-version SP Status-Code SP Reason-Phrase CRLF Status-Code = 1xx: Información (Comando recibido, procesando,..) | 2xx: Exito (Comando recibido y ejecutado con éxito) | 3xx: Re-dirección (Comando recibido pero aún no completado) | 4xx: Error del cliente (El comando tiene errores de sintaxis) | 5xx: Error del servidor (Error interno del servidor)

RTSP: Una sesión completa (I) web server W cliente C media server A media server V 1 3 2 4 C->W: GET /twister.sdp HTTP/1.1 Host: www.example.com Accept: application/sdp W->C: HTTP/1.0 200 OK Content-Type: application/sdp v=0 o=- 2890844526 2890842807 IN IP4 192.16.24.202 s=RTSP Session m=audio 0 RTP/AVP 0 a=control:rtsp://audio.example.com/twister/audio.en m=video 0 RTP/AVP 31 a=control:rtsp://video.example.com/twister/video

RTSP: Una sesión completa (II) C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0 CSeq: 1 Transport: RTP/AVP/UDP;unicast;client_port=3056-3057 A->C: RTSP/1.0 200 OK Session: 12345678 Transport: RTP/AVP/UDP;unicast;client_port=3056-3057; server_port=5000-5001 C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0 Transport: RTP/AVP/UDP;unicast;client_port=3058-3059 V->C: RTSP/1.0 200 OK Session: 23456789 Transport: RTP/AVP/UDP;unicast;client_port=3058-3059; server_port=5002-5003

RTSP: Una sesión completa (III) C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0 CSeq: 2 Session: 23456789 Range: smpte=0:10:00- V->C: RTSP/1.0 200 OK Range: smpte=0:10:00-0:20:00 RTP-Info: url=rtsp://video.example.com/twister/video; seq=12312232;rtptime=78712811 C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0 Session: 12345678 A->C: RTSP/1.0 200 OK RTP-Info: url=rtsp://audio.example.com/twister/audio.en; seq=876655;rtptime=1032181

RTSP: Una sesión completa (IV) C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0 CSeq: 3 Session: 12345678 A->C: RTSP/1.0 200 OK C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0 Session: 23456789 V->C: RTSP/1.0 200 OK

1- Protocolos de transporte multimedia. Clases de aplicaciones multimedia Redes basadas en IP y QoS Gestión de los recursos: IntServ vs DiffServ RSVP RTP/RTCP: Transporte de flujos multimedia RTSP: Control de sesión y localización de medios Multicasting

Multicast = Efficient Data Distribution Src Src

Need for efficient one-to-many delivery of same data Applications: Why Multicast ? Need for efficient one-to-many delivery of same data Applications: News/sports/stock/weather updates Distance learning Configuration, routing updates, service location Pointcast-type “push” apps Teleconferencing (audio, video, shared whiteboard, text editor) Distributed interactive gaming or simulations Email distribution lists Content distribution; Software distribution Web-cache updates Database replication

Why Not Broadcast or Unicast? Send a copy to every machine on the net Simple, but inefficient All nodes must process packet even if they don’t care Wastes more CPU cycles of slower machines (“broadcast radiation”) Network loops lead to “broadcast storms” Replicated Unicast: Sender sends a copy to each receiver in turn Receivers need to register or sender must be pre-configured Sender is focal point of all control traffic Reliability => per-receiver state, separate sessions/processes at sender

Multicast Apps Characteristics Number of (simultaneous) senders to the group The size of the groups Number of members (receivers) Geographic extent or scope Diameter of the group measured in router hops The longevity of the group Number of aggregate packets/second The peak/average used by source Level of human interactivity Lecture mode vs interactive Data-only (eg database replication) vs multimedia

Reliable Multicast vs. Unreliable Multicast When a multicast message is sent by a process, the runtime support of the multicast mechanism is responsible for delivering the message to each process currently in the multicast group. As each participating process may be on a separate host, due to factors such as failures of network links and/or network hosts, routing delays, and differences in software and hardware, the time between when a message is sent and when it is received may vary among the recipient processes. Moreover, a message may not be received by one or more of the processes at all.

Classification of multicasting mechanisms in terms of message delivery Unreliable multicast: The arrival of the correct message at each process is not guaranteed. Reliable multicast: Guarantees that each message is eventually delivered in a non-corrupted form to each process in the group. The definition of reliable multicast requires that each participating process receives exactly one copy of each message sent. It does not put any restriction of the order the messages delivered. Reliable multicast can be further classified based on the order of the delivery of the messages: unordered, FIFO, causal order, atomic order.

Classification of reliable multicast -- unordered An unordered reliable multicast system guarantees the safe delivery of each message, but it provides no guarantee on the delivery order of the messages. Example: Processes P1, P2, and P3 have formed a multicast group. Three messages, m1, m2, m3 have been sent to the group. An unordered reliable multicast system may deliver the messages to each of the three processes in any of these: m1-m2-m3, m1-m3-m2, m2-m1-m3, m2-m3-m1, m3-m1-m2, m3-m2-m1

Classification of reliable multicast - FIFO If process P sent messages mi and mj, in that order, then each process in the multicast group will be delivered the messages mi and mj, in that order. Note that FIFO multicast places no restriction on the delivery order among messages sent by different processes. For example, P1 sends messages m11 then m12, and P2 sends messages m21 then m22. It is possible for different processes to receive any of the following orders: m11-m12-m21-m22, m11-m21-m12-m22, m11-m21-m22-m12, m21-m11-m12-m22 m21-m11-m22-m12 m21-m22-m11-m12.

Classification of reliable multicast – Causal order If message mi causes (results in) the occurrence of message mj, then mi will be delivered to each process prior to mj. Messages mi and mj are said to have a causal or happen-before relationship. For example, P1 sends a message m1, to which P2 replies with a multicast message m2. Since m2 is triggered by m1, the two messages share a causal relationship of m1-> m2. A causal-order multicast message system ensures that these two messages will be delivered to each of the processes in the order of m1- m2.

Classification of reliable multicast – Atomic order In an atomic-order multicast system, all messages are guaranteed to be delivered to each participant in the exact same order. Note that the delivery order does not have to be FIFO or causal, but must be identical for each process. Example: P1 sends m1, P2 sends m2, and P3 sends m3. An atomic system will guarantee that the messages will be delivered to each process in only one of the six orders: m1-m2- m3, m1- m3- m2, m2- m1-m3, m2-m3-m1, m3-m1- m2, m3-m2-m1.

IP Multicast Architecture Service model Hosts Host-to-router protocol (IGMP) Routers Multicast routing protocols (various)

IP Multicast model: RFC 1112 Message sent to multicast “group” (of receivers) Senders need not be group members A group identified by a single “group address” Use “group address” instead of destination address in IP packet sent to group Groups can have any size; Group members can be located anywhere on the Internet Group membership is not explicitly known Receivers can join/leave at will Packets are not duplicated or delivered to destinations outside the group Distribution tree constructed for delivery of packets No more than one copy of packet appears on any subnet Packets delivered only to “interested” receivers => multicast delivery tree changes dynamically Network has to actively discover paths between senders and receivers

IP Multicast Addresses Class D IP addresses 224.0.0.0 – 239.255.255.255 Address allocation: Well-known (reserved) multicast addresses, assigned by IANA: 224.0.0.x and 224.0.1.x Transient multicast addresses, assigned and reclaimed dynamically, e.g., by “sdr” program Each multicast address represents a group of arbitrary size, called a “host group” There is no structure within class D address space like subnetting => flat address space 1 Group ID

IP Multicast Service Sending Receiving Uses normal IP-Send operation, with an IP multicast address specified as the destination Must provide sending application a way to: Specify outgoing network interface, if >1 available Specify IP time-to-live (TTL) on outgoing packet Enable/disable loop-back if the sending host is/isn't a member of the destination group on the outgoing interface Receiving Two new operations Join-IP-Multicast-Group(group-address, interface) Leave-IP-Multicast-Group(group-address, interface) Receive multicast packets for joined groups via normal IP-Receive operation