1- Protocolos de transporte con QoS

1- Protocolos de transporte con QoS
Clases de aplicaciones multimedia Redes basadas en IP y QoS RTP/RTCP: Transporte de flujos multimedia RTSP: Control de sesión y localización de medios Multicasting Computer Networking: A Top Down Approach 6th edition Jim Kurose, Keith Ross Addison-Wesley March 2012

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 , 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

Multimedia networking: 3 application types
streaming, stored audio, video streaming: can begin playout before downloading entire file stored (at server): can transmit faster than audio/video will be rendered (implies storing/buffering at client) e.g., YouTube, Netflix, Hulu conversational voice/video over IP interactive nature of human-to-human conversation limits delay tolerance e.g., Skype streaming live audio, video e.g., live sporting event (futbol)

Streaming stored video:
3. video received, played out at client (30 frames/sec) streaming: at this time, client playing out early part of video, while server still sending later part of video Cumulative data 2. video sent video recorded (e.g., 30 frames/sec) network delay (fixed in this example) time

Streaming stored video: revisted
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 and playout delay: compensate for network-added delay, delay jitter

Client-side buffering, playout
buffer fill level, Q(t) variable fill rate, x(t) playout rate, e.g., CBR r client application buffer, size B video server client

buffer fill level, Q(t) variable fill rate, x(t) playout rate, e.g., CBR r client application buffer, size B video server client 1. Initial fill of buffer until playout begins at tp 2. playout begins at tp, 3. buffer fill level varies over time as fill rate x(t) varies and playout rate r is constant

buffer fill level, Q(t) variable fill rate, x(t) playout rate, e.g., CBR r client application buffer, size B video server playout buffering: average fill rate (x), playout rate (r): x < r: buffer eventually empties (causing freezing of video playout until buffer again fills) x > r: buffer will not empty, provided initial playout delay is large enough to absorb variability in x(t) initial playout delay tradeoff: buffer starvation less likely with larger delay, but larger delay until user begins watching

Streaming multimedia: UDP
server sends at rate appropriate for client often: send rate = encoding rate = constant rate transmission rate can be oblivious to congestion levels short playout delay (2-5 seconds) to remove network jitter error recovery: application-level, timeipermitting RTP [RFC 2326]: multimedia payload types UDP may not go through firewalls

Streaming multimedia: HTTP
multimedia file retrieved via HTTP GET send at maximum possible rate under TCP fill rate fluctuates due to TCP congestion control, retransmissions (in-order delivery) larger playout delay: smooth TCP delivery rate HTTP/TCP passes more easily through firewalls variable rate, x(t) video file TCP send buffer TCP receive buffer application playout buffer server client

Streaming multimedia: DASH
DASH: D ynamic, A daptive S treaming over H TTP server: divides video file into multiple chunks each chunk stored, encoded at different rates manifest file: provides URLs for different chunks client: periodically measures server-to-client bandwidth consulting manifest, requests one chunk at a time chooses maximum coding rate sustainable given current bandwidth can choose different coding rates at different points in time (depending on available bandwidth at time)

Streaming multimedia: DASH
DASH: D ynamic, A daptive S treaming over H TTP “intelligence” at client: client determines when to request chunk (so that buffer starvation, or overflow does not occur) what encoding rate to request (higher quality when more bandwidth available) where to request chunk (can request from URL server that is “close” to client or has high available bandwidth)

Content distribution networks
challenge: how to stream content (selected from millions of videos) to hundreds of thousands of simultaneous users? option 1: single, large “mega-server” single point of failure point of network congestion long path to distant clients multiple copies of video sent over outgoing link ….quite simply: this solution doesn’t scale

Content distribution networks
challenge: how to stream content (selected from millions of videos) to hundreds of thousands of simultaneous users? option 2: store/serve multiple copies of videos at multiple geographically distributed sites (CDN) enter deep: push CDN servers deep into many access networks close to users used by Akamai, 1700 locations bring home: smaller number (10’s) of larger clusters in POPs near (but not within) access networks used by Limelight

CDN: “simple” content access scenario
Bob (client) requests video video stored in CDN at 1. Bob gets URL for for video from netcinema.com web page 1 2. resolve via Bob’s local DNS 2 5 6. request video from KINGCDN server, streamed via HTTP 4&5. Resolve via KingCDN’s authoritative DNS, which returns IP address of KIingCDN server with video netcinema.com 3. netcinema’s DNS returns URL 4 3 netcinema’s authorative DNS KingCDN authoritative DNS KingCDN.com

CDN cluster selection strategy
challenge: how does CDN DNS select “good” CDN node to stream to client pick CDN node geographically closest to client pick CDN node with shortest delay (or min # hops) to client (CDN nodes periodically ping access ISPs, reporting results to CDN DNS) IP anycast alternative: let client decide - give client a list of several CDN servers client pings servers, picks “best” Netflix approach

own registration, payment servers Amazon (3rd party) cloud services:
Case study: Netflix 30% downstream US traffic in 2011 owns very little infrastructure, uses 3rd party services: own registration, payment servers Amazon (3rd party) cloud services: Netflix uploads studio master to Amazon cloud create multiple version of movie (different encodings) in cloud upload versions from cloud to CDNs Cloud hosts Netflix web pages for user browsing three 3rd party CDNs host/stream Netflix content: Akamai, Limelight, Level-3

Case study: Netflix 1 1. Bob manages Netflix account Netflix registration, accounting servers Amazon cloud Akamai CDN Limelight CDN Level-3 CDN 2 2. Bob browses Netflix video 3 3. Manifest file returned for requested video 4. DASH streaming upload copies of multiple versions of video to CDNs

value-added services: call forwarding, screening, recording
Voice-over-IP (VoIP) VoIP end-end-delay requirement: needed to maintain “conversational” aspect higher delays noticeable, impair interactivity < 150 msec: good > 400 msec bad includes application-level (packetization,playout), network delays session initialization: how does callee advertise IP address, port number, encoding algorithms? value-added services: call forwarding, screening, recording emergency services: 911

speaker’s audio: alternating talk spurts, silent periods.
VoIP characteristics speaker’s audio: alternating talk spurts, silent periods. 64 kbps during talk spurt pkts generated only during talk spurts 20 msec chunks at 8 Kbytes/sec: 160 bytes of data application-layer header added to each chunk chunk+header encapsulated into UDP or TCP segment application sends segment into socket every 20 msec during talkspurt

VoIP: packet loss, delay
network loss: IP datagram lost due to network congestion (router buffer overflow) delay loss: IP datagram arrives too late for playout at receiver delays: processing, queueing in network; end-system (sender, receiver) delays typical maximum tolerable delay: 400 ms loss tolerance: depending on voice encoding, loss concealment, packet loss rates between 1% and 10% can be tolerated

Delay jitter constant bit rate transmission variable network delay (jitter) client reception constant bit rate playout at client client playout delay Cumulative data buffered data time end-to-end delays of two consecutive packets: difference can be more or less than 20 msec (transmission time difference)

VoIP: fixed playout delay
receiver attempts to playout each chunk exactly q msecs after chunk was generated. chunk has time stamp t: play out chunk at t+q chunk arrives after t+q: data arrives too late for playout: data “lost” tradeoff in choosing q: large q: less packet loss small q: better interactive experience

VoIP: fixed playout delay
sender generates packets every 20 msec during talk spurt. first packet received at time r first playout schedule: begins at p second playout schedule: begins at p’

Adaptive playout delay (1)
goal: low playout delay, low late loss rate approach: adaptive playout delay adjustment: estimate network delay, adjust playout delay at beginning of each talk spurt silent periods compressed and elongated chunks still played out every 20 msec during talk spurt adaptively estimate packet delay: (EWMA - exponentially weighted moving average, recall TCP RTT estimate): di = (1-a)di-1 + a (ri – ti) delay estimate after ith packet small constant, e.g. 0.1 time received - time sent (timestamp) measured delay of ith packet

also useful to estimate average deviation of delay, vi : vi = (1-b)vi-1 + b |ri – ti – di| estimates di, vi calculated for every received packet, but used only at start of talk spurt for first packet in talk spurt, playout time is: remaining packets in talkspurt are played out periodically playout-timei = ti + di + Kvi

Q: How does receiver determine whether packet is first in a talkspurt? if no loss, receiver looks at successive timestamps difference of successive stamps > 20 msec -->talk spurt begins. with loss possible, receiver must look at both time stamps and sequence numbers difference of successive stamps > 20 msec and sequence numbers without gaps --> talk spurt begins.

VoiP: recovery from packet loss (1)
Challenge: recover from packet loss given small tolerable delay between original transmission and playout each ACK/NAK takes ~ one RTT alternative: Forward Error Correction (FEC) send enough bits to allow recovery without retransmission (recall two-dimensional parity in Ch. 5) simple FEC for every group of n chunks, create redundant chunk by exclusive OR-ing n original chunks send n+1 chunks, increasing bandwidth by factor 1/n can reconstruct original n chunks if at most one lost chunk from n+1 chunks, with playout delay

another FEC scheme: “piggyback lower quality stream” send lower resolution audio stream as redundant information e.g., nominal stream PCM at 64 kbps and redundant stream GSM at 13 kbps non-consecutive loss: receiver can conceal loss generalization: can also append (n-1)st and (n-2)nd low-bit rate chunk

interleaving to conceal loss: audio chunks divided into smaller units, e.g. four 5 msec units per 20 msec audio chunk packet contains small units from different chunks if packet lost, still have most of every original chunk no redundancy overhead, but increases playout delay

clients: skype peers connect directly to each other for VoIP call
Voice-over-IP: Skype Skype clients (SC) proprietary application-layer protocol (inferred via reverse engineering) encrypted msgs P2P components: Skype login server supernode (SN) clients: skype peers connect directly to each other for VoIP call supernode overlay network super nodes (SN): skype peers with special functions overlay network: among SNs to locate SCs login server

P2P voice-over-IP: skype
skype client operation: 1. joins skype network by contacting SN (IP address cached) using TCP Skype login server 2. logs-in (usename, password) to centralized skype login server 3. obtains IP address for callee from SN, SN overlay or client buddy list 4. initiate call directly to callee

relay solution: Alice, Bob maintain open connection to their SNs
Skype: peers as relays problem: both Alice, Bob are behind “NATs” NAT prevents outside peer from initiating connection to insider peer inside peer can initiate connection to outside relay solution: Alice, Bob maintain open connection to their SNs Alice signals her SN to connect to Bob Alice’s SN connects to Bob’s SN Bob’s SN connects to Bob over open connection Bob initially initiated to his SN

1- Protocolos de transporte con QoS.
Clases de aplicaciones multimedia Redes basadas en IP y QoS 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

Network support for multimedia

Dimensioning best effort networks
approach: deploy enough link capacity so that congestion doesn’t occur, multimedia traffic flows without delay or loss low complexity of network mechanisms (use current “best effort” network) high bandwidth costs challenges: network dimensioning: how much bandwidth is “enough?” estimating network traffic demand: needed to determine how much bandwidth is “enough” (for that much traffic)

Providing multiple classes of service
thus far: making the best of best effort service one-size fits all service model alternative: multiple classes of service partition traffic into classes network treats different classes of traffic differently (analogy: VIP service versus regular service) granularity: differential service among multiple classes, not among individual connections history: ToS bits 0111

Multiple classes of service: scenario
H3 H1 R1 R2 H4 H2 R1 output interface queue 1.5 Mbps link

Scenario 1: mixed HTTP and VoIP
example: 1Mbps VoIP, HTTP share 1.5 Mbps link. HTTP bursts can congest router, cause audio loss want to give priority to audio over HTTP R1 R2 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 (VoIP sends higher than declared rate) policing: force source adherence to bandwidth allocations marking, policing at network edge 1 Mbps phone R1 R2 1.5 Mbps link packet marking and policing 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 1 Mbps phone 1 Mbps logical link R1 R2 1.5 Mbps link 0.5 Mbps logical link Principle 3 while providing isolation, it is desirable to use resources as efficiently as possible

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 real-world example? 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 packet arrivals packet departures queue (waiting area) link (server)

Scheduling policies: priority
high priority queue (waiting area) low priority queue arrivals classify departures link (server) priority scheduling: send 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. real world example? 2 1 3 4 5 arrivals 1 3 2 4 5 packet in service departures 1 3 2 4 5

Scheduling policies: still more
Round Robin (RR) scheduling: multiple classes cyclically scan class queues, sending one complete packet from each class (if available) real world example? 1 2 3 4 5 arrivals departures packet in service

Scheduling policies: still more
Weighted Fair Queuing (WFQ): generalized Round Robin each class gets weighted amount of service in each cycle real-world example?

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 ppm peak rate (max.) burst size: max number of pkts sent consecutively (with no intervening idle)

Policing mechanisms: implementation
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 and QoS guarantees
token bucket, WFQ combine to provide guaranteed upper bound on delay, i.e., QoS guarantee! arriving traffic token rate, r bucket size, b per-flow rate, R WFQ D = b/R max arriving traffic

Differentiated services
want “qualitative” service classes “behaves like a wire” relative service distinction: Platinum, Gold, Silver scalability: simple functions in network core, relatively complex functions at edge routers (or hosts) signaling, maintaining per-flow router state difficult with large number of flows don’t define define service classes, provide functional components to build service classes

Diffserv architecture
b marking edge router: per-flow traffic management marks packets as in-profile and out-profile scheduling . core router: per class traffic management buffering and scheduling based on marking at edge preference given to in-profile packets over out-of-profile packets

profile: pre-negotiated rate r, bucket size b
Edge-router packet marking profile: pre-negotiated rate r, bucket size b packet marking at edge based on per-flow profile rate r b user packets possible use of marking: class-based marking: packets of different classes marked differently intra-class marking: conforming portion of flow marked differently than non-conforming one

Diffserv packet marking: details
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) determine PHB that the packet will receive 2 bits currently unused DSCP unused

Classification, 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 Per-hop Behavior (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 proposed: 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

Per-connection QOS guarantees
basic fact of life: can not support traffic demands beyond link capacity 1 Mbps phone R1 R2 1.5 Mbps link 1 Mbps phone Principle 4 call admission: flow declares its needs, network may block call (e.g., busy signal) if it cannot meet needs

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

1- Protocolos de transporte multimedia.
Clases de aplicaciones multimedia Redes basadas en IP y QoS 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) ______________________________________________ PCMU A A G A GSM A ... 34-71 unassigned ? 72-76 reserved N/A N/A N/A 77-95 unassigned ? 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 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

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, , 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

Clases de aplicaciones multimedia Redes basadas en IP y QoS 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: 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: Accept: application/sdp W->C: HTTP/ OK Content-Type: application/sdp v=0 o= IN IP 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= A->C: RTSP/ OK Session: Transport: RTP/AVP/UDP;unicast;client_port= ; server_port= C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0 Transport: RTP/AVP/UDP;unicast;client_port= V->C: RTSP/ OK Session: Transport: RTP/AVP/UDP;unicast;client_port= ; server_port=

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

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

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 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 – Address allocation: Well-known (reserved) multicast addresses, assigned by IANA: x and 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

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