1- Protocolos de transporte con QoS

Transcripción

1 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 6 th edition Jim Kurose, Keith Ross Addison-Wesley March 2012 TECNOLOGÍAS DE RED AVANZADAS Master IC

2 Definition of multimedia Hard to find a clear-cut definition What is multimedia? 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 2

3 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 3

4 Classification of Media Type Sound Video Animation Continuous Continuous Image Text Graphics Discrete Discrete Captured From real world Synthesized By computer 4

5 Text Plain 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 5

6 Plain Text vs. Rich Text An example of Plain text Example of Rich text 6

7 Graphics Revisable document that retains structural information Consists of objects such as lines, curves, circles, etc Usually generated by graphic editor of computer programs 10 5 Example of graphics (FIG file)

8 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 Digital still image Computer software Capture and A/D conversion Synthesized image Scanned image 8 Camera

9 Images (cont.) Examples of images Binary image pixel depth 1 Gray-scale pixel depth 8 Color image pixel depth 24 Gray-scale color image Binary image 9

10 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 1 0

11 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

12 Networked Multimedia 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. A scenario of multimedia networking Image server Video server Internet 1 2

13 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 1 3

14 Technologies of Multimedia Networking 1 4 Challenges of multimedia networking 1. Conflict between media size and bandwidth limit of the network 2. Conflict between the user requirement of multimedia application and the best-effort network 3. 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

15 Multimedia Networking Systems 1 5 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

16 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) 1 6

17 Streaming stored video: 1. video recorded (e.g., 30 frames/sec) 2. video sent network delay (fixed in this example) 3. video received, played out at client (30 frames/sec) time streaming: at this time, client playing out early part of video, while server still sending later part of video

18 buffered video 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 time client-side buffering and playout delay: compensate for network-added delay, delay jitter

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

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

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

22 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

23 Streaming multimedia: HTTP multimedia file retrieved via HTTP GET send at maximum possible rate under TCP variable rate, x(t) video file TCP send buffer server TCP receive buffer client application playout buffer 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

24 Streaming multimedia: DASH DASH: D ynamic, Adaptive 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)

25 Streaming multimedia: DASH DASH: D ynamic, Adaptive 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)

26 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

27 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

28 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 netcinema.com 3 6. request video from KINGCDN server, streamed via HTTP 3. netcinema s DNS returns URL resolve via Bob s local DNS 4 4&5. Resolve via KingCDN s authoritative DNS, which returns IP address of KIingCDN server with video netcinema s authorative DNS KingCDN.com KingCDN authoritative DNS

29 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

30 Case study: Netflix 30% downstream US traffic in 2011 owns very little infrastructure, uses 3 rd party services: own registration, payment servers Amazon (3 rd 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 3 rd party CDNs host/stream Netflix content: Akamai, Limelight, Level-3

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

32 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

33 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

34 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

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

36 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

37 packets 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 packets generated packets received loss playout schedule p' - r playout schedule p - r time r p p'

38 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): d i = (1-a)d i-1 + a (r i t i ) delay estimate after ith packet small constant, e.g. 0.1 time received - time sent (timestamp) measured delay of ith packet

39 Adaptive playout delay (2) also useful to estimate average deviation of delay, v v i = (1-b)v i-1 + b r i t i d i estimates d i, v i calculated for every received packet, but used only at start of talk spurt for first packet in talk spurt, playout time is: playout-time i = t i + d i + Kv i remaining packets in talkspurt are played out periodically

40 Adaptive playout delay (3) 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.

41 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

42 VoiP: recovery from packet loss (2) 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

43 VoiP: recovery from packet loss (3) 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

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

45 P2P voice-over-ip: skype skype client operation: 1. joins skype network by contacting SN (IP address cached) using TCP 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 Skype login server

46 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

47 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) TECNOLOGÍAS DE RED AVANZADAS Master IC

48 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) Mensaje listo para envío Envío del primer bit del mensaje Primer bit del mensaje recibido Ultimo bit del mensaje recibido Mensaje listo para recepción 4 8 Retardo de acceso Retardo Retardo de de tránsito transmisión Retardo extremo-a-extremo Retardo de acceso

49 Requisitos de red (II). Varianza del retardo (Jitter) Define la variabilidad del retardo de una red Emisor t D 1 D 2 = D 1 D 3 > D 1 Receptor t Jitter físico (redes de conmutación de circuito) 4 9 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.

50 Internet y las aplicaciones multimedia 5 0 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 RTSP (Real-Time Streaming Protocol) RFC 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

51 Internet Protocols 5 1 Slide thanks to Henning Schulzrinne

52 Network support for multimedia

53 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)

54 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

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

56 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

57 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

58 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 R1 1 Mbps logical link 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

59 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 queue (waiting area) link (server) packet departures

60 Scheduling policies: priority 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? arrivals arrivals packet in service classify high priority queue (waiting area) low priority queue (waiting area) 4 link (server) departures departures

61 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? 2 arrivals packet in service departures

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

63 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)

64 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)

65 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 WFQ per-flow rate, R arriving traffic D = b/r max

66 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

67 edge router: per-flow traffic management marks packets as in-profile and out-profile Diffserv architecture marking r b 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

68 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

69 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

70 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 7-70

71 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

72 PHBs proposed: Forwarding PHB 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

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

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

75 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 TECNOLOGÍAS DE RED AVANZADAS Master IC

76 RTP (Real-time Transport Protocol) 7 6 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

77 RTP (Real-time Transport Protocol) 7 7 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.

78 Mezcladores (Mixers). RTP: Mezcladores y traductores 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) 7 8

79 RTP: Formato de mensaje (I) 32 bits V P X CC M PT Sequence number Timestamp Synchronization Source (SSRC) ID Contributing Source (CSRC) ID 7 9 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

80 RTP: Formato de mensaje (II) 32 bits V P X CC M PT Sequence number Timestamp Synchronization Source (SSRC) ID Contributing Source (CSRC) ID 8 0 CC: Identifica el número de CSRC que contribuyen a los datos M: Marca (definida según el perfil) PT: Tipo de datos (según perfil)

81 RTP: Formato de mensaje (III) 32 bits V P X CC M PT Sequence number Timestamp Synchronization Source (SSRC) ID Contributing Source (CSRC) ID 8 1 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

82 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; 8 2

83 RTP y las aplicaciones 8 3 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 A G721 A GSM A unassigned? reserved N/A N/A N/A 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.

84 RTCP (RTP Control Protocol) 8 4 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.

85 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 8 5

86 Formato de un paquete RTCP: RTCP (RTP Control Protocol) 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). 8 6

87 RTCP: Mensajes SR 32 bits 8 7 cabecera Sender info Report block 1 V P RC PT=SR=200 Longitud SSRC del sender 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 Ultimo SR (LSR) Retraso del último SR (LSR)

88 RTCP: Cálculo del Jitter Es una estimación de la variancia del tiempo de interllegada de los paquetes RTP D( i, j) ( Rj - Ri ) -( S j - Si ) ( Rj - S j ) -( Ri - Si ) S i RTP timestamp del paquete i R i Instante de llegada del paquete i J i J D i -1, i J i 1 / 16 i

89 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 TECNOLOGÍAS DE RED AVANZADAS Master IC

90 Real-Time Streaming Protocol RFC 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

91 Terminología RTSP Conferencia Media stream 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 Imagen de las transparencias Imagen del público Voz del conferenciante Imagen del conferenciante Voz del público 9 1

92 RTSP: Ejemplo de una sesión HTTP GET descripción de la sesión Web server Cliente SETUP PLAY Media server RTP audio RTP vídeo RTCP PAUSE TEARDOWN 9 2

93 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" extension-method = token "OPTIONS "PAUSE" "PLAY" "RECORD" "REDIRECT" "SETUP" "SET_PARAMETER" "TEARDOWN" extension-method Request-URI = "*" absolute_uri RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT 9 3

94 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) 9 4

95 RTSP: Una sesión completa (I) web server W media server A cliente C 3 media server V C->W: GET /twister.sdp HTTP/1.1 Host: Accept: application/sdp W->C: HTTP/ OK Content-Type: application/sdp 9 5 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

96 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 CSeq: 1 Session: Transport: RTP/AVP/UDP;unicast;client_port= ; server_port= C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0 CSeq: 1 Transport: RTP/AVP/UDP;unicast;client_port= V->C: RTSP/ OK CSeq: 1 Session: Transport: RTP/AVP/UDP;unicast;client_port= ; server_port=

97 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 CSeq: 2 Session: 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 CSeq: 2 Session: Range: smpte=0:10: A->C: RTSP/ OK CSeq: 2 Session: Range: smpte=0:10:00-0:20:00 RTP-Info: url=rtsp://audio.example.com/twister/audio.en; seq=876655;rtptime=

98 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 CSeq: 3 C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0 CSeq: 3 Session: V->C: RTSP/ OK CSeq: 3 9 8