Lec 17 Multimedia Transport: RTP, TCP/HTTP and QUIC

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1 Multimedia Communication Lec 17 Multimedia Transport: RTP, TCP/HTTP and QUIC Zhu Li Course Web: Z. Li, Multimedia Communciation, Spring 2017 p.1

2 Outline ReCap Lecture 16 Multimedia Transport Managed IP Networks Media Transport: UDP/RTP/RTSP/RSVP Over The Top (OTT) Transport: HTTP, Webcoket Emerging Solutions: WebRTC/QUIC Summary Z. Li, Multimedia Communciation, Spring 2017 p.2

3 DASH FDH Introducing Push Based Operation Benefit: Reduce the HTTP GET overhead Avoid TCP transmission window collapse Implementation: HTTP 2.0: additional messages WebSocket: DASH sub-protocol for WebSocket Z. Li, Multimedia Communciation, Spring 2017 p.3

4 Establishing WebSocket Connection Upgrade to WebSocket over HTTP1.1 WebSocket DASH sub-protocol Z. Li, Multimedia Communciation, Spring 2017 p.4

5 WebSocket Frames Support binary data channel Opcode: o 0x0 Continuation frame, 0x1 Connection close 0x2 Ping, 0x3 Pong, 0x4 Text frame, 0x5 Binary frame 0x7-0xF Reserved Z. Li, Multimedia Communciation, Spring 2017 p.5

6 DASH SAND DASH Boxes Client, Server, DANE. SAND Messages Parameters Enhancing Delivery (PED) messages that are exchanged between DANEs, Parameters Enhancing Reception (PER) messages that are sent from DANEs to DASH clients, Metrics and Status messages that are sent from DASH clients to DANEs. Z. Li, Multimedia Communciation, Spring 2017 p.6

7 Outline ReCap Lecture 16 Multimedia Transport Managed IP Networks Media Transport: UDP/RTP/RTSP/RSVP Over The Top (OTT) Transport: HTTP, Webcoket Emerging Solutions: WebRTC/QUIC Summary Z. Li, Multimedia Communciation, Spring 2017 p.7

8 Real Time Media Streaming Real-time multimedia applications Video teleconferencing Internet Telephony (VoIP) Internet audio, video streaming (App Payload Data Unit (A-PDU)s) Z. Li, Multimedia Communciation, Spring 2017 p.8

9 Streaming Timing Ideal Streaming Packet Timing application Send time Play time Z. Li, Multimedia Communciation, Spring 2017 p.9

10 Jittering Jitter from random delays delay Send time Play time Z. Li, Multimedia Communciation, Spring 2017 p.10

11 Buffering remove jitters De-Jittering from Buffering Send time Packet Arrival Playback Time Z. Li, Multimedia Communciation, Spring 2017 p.11

12 Basic Streaming Functions Packet Sequencing o to report Payload Data Unit (PDU) loss o to report PDU reordering o to perform out-of-order decoding Time stamping and Buffering o for correct play out (CTS/DTS) o Jitter computation and de-jitter buffering Payload type identification o for media interpretation/decoding Error concealment/resync o covers up errors from lost PDU by using redundancy in most-adjacentframe Quality of Service (QoS) feedback o from receiver to sender for operation adjustment Rate control o sender reduces sending rate adaptively to network congestion Z. Li, Multimedia Communciation, Spring 2017 p.12

13 RTP/RTCP/RTSP Protocol Stack IP Network Media Transport Solution RTP- Real Time Protocol on top of UDP data channel RTCP Real Time Control Protocol, on top of UDP RTSP Real Time Session Protocol, user control messaging, on top of HTTP RTSP RTP RTCP TCP (till now) Z. Li, Multimedia Communciation, Spring 2017 p.13

14 RTP/RTCP Session RTP/RTCP audio/visual session RTP session is sending and receiving of RTP data by a group of participants For each participant, a session is a pair of transport addresses used to communicate with the group If multiple media types are communicated by the group, the transmission of each medium constitutes a session. Z. Li, Multimedia Communciation, Spring 2017 p.14

15 RTP Sync Scheme Sync via SSRC ids synchronization source - each source of RTP PDUs Identified by a unique,randomly chosen 32-bit ID (the SSRC) A host generating multiple streams within a single RTP must use a different SSRC per stream Z. Li, Multimedia Communciation, Spring 2017 p.15

16 RTP Header Payload type RTP Header Incremented by one for each RTP PDU: PDU loss detection Restore PDU sequence Identifies synchronization source Identifies contributing sources (used by mixers) Z. Li, Multimedia Communciation, Spring 2017 p.16

17 RTP Mixer RTP mixer - an intermediate system that receives & combines RTP PDUs of one or more RTP sessions into a new RTP PDU Stream may be transcoded, special effects may be performed. A mixer will typically have to define synchronization relationships between streams.thus Sources that are mixed together become contributing sources (CSRC) Mixer itself appears as a new source having a new SSRC Z. Li, Multimedia Communciation, Spring 2017 p.17

18 RTP Control Protocol (RTCP) RTCP specifies report PDUs exchanged between sources and destinations of multimedia information receiver reception report sender report source description report Reports contain statistics such as the number of RTP-PDUs sent, number of RTP- PDUs lost, inter-arrival jitter Used by application to modify sender transmission rates and for diagnostics purposes Z. Li, Multimedia Communciation, Spring 2017 p.18

19 RTCP Messages Message Types: Typically, several RTCP PDUs of different types are transmitted in a single UDP PDU Z. Li, Multimedia Communciation, Spring 2017 p.19

20 RTCP bandwidth scaling in multicast Problem What happens when there is one sender and many receivers? RTCP reports scale linearly with the number of participants and would match or exceed the amount of RTP data! More overhead than useful data! Solution RTCP attempts to limit its traffic to 5% of the session bandwidth to ensure it can scale! RTCP gives 75% of this rate to the receivers; and the remaining 25% to the sender. Example Suppose one sender, sending video at a rate of 2 Mbps. Then RTCP attempts to limit its traffic to 100 Kbps. The 75 kbps is equally shared among receivers: With R receivers, each receiver gets to send RTCP traffic at 75/R kbps. Sender gets to send RTCP traffic at 25 kbps. Z. Li, Multimedia Communciation, Spring 2017 p.20

21 RTSP Real Time Streaming Protocol (RTSP) aka Remote Controller In session user trick mode support: fwd, bwd, pause, play, etc Web Server web browser HTTP presentation descriptor Presentation descriptor media player RTSP pres. desc,streaming commands RTP/RTCP audio/video content Web Server/Media server Z. Li, Multimedia Communciation, Spring 2017 p.21

22 RTSP Methods Supported methods OPTIONS C S C S determine capabilities of server/client DESCRIBE C S get description of media stream ANNOUNCE C S announce new session description SETUP C S create media session RECORD C S start media recording PLAY C S start media delivery PAUSE C S pause media delivery REDIRECT C S redirection to another server TEARDOWN C S immediate teardown SET_PARAMETER C S change server/client parameter GET_PARAMETER C S read server/client parameter Z. Li, Multimedia Communciation, Spring 2017 p.22

23 Typical RTSP Setup RTSP session get UDP port RTSP server data source RTSP SETUP RTSP OK RTSP PLAY RTSP OK RTSP TEARDOWN RTSP OK RTP VIDEO RTP AUDIO TCP UDP RTSP client AV subsystem choose UDP port media server RTCP media player Z. Li, Multimedia Communciation, Spring 2017 p.23

24 RTP/RTCP/RTSP References [1] B. A. Forouzan, TCP/IP Protocol Suite, Third edition, [2] H. Schulzrinne, S. Casner, R. Frederick and V. Jacobson, "RTP: a transport protocol for real-time applications", RFC 3550, July [3] H. Schulzrinne, A. Rao and R. Lanphier, "Real Time Streaming Protocol (RTSP)", RFC 2326, April Z. Li, Multimedia Communciation, Spring 2017 p.24

25 Outline ReCap Lecture 16 Multimedia Transport Managed IP Networks Media Transport: UDP/RTP/RTSP/RSVP Over The Top (OTT) Transport: HTTP, Webcoket Emerging Solutions: WebRTC/QUIC Summary Z. Li, Multimedia Communciation, Spring 2017 p.25

26 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 Z. Li, Multimedia Communciation, Spring 2017 p.26

27 HTTP History Evolution of Web Content Transport: Persistent connections Virtual host support Conditional caching Digest authentication Chunked transfer encoding Enhanced compression Header compression Security requirements Interleaving requests and responses Push operations Binary instead of textual 1996 HTTP HTTP SPDY HTTP 2.0 Rise of the Internet as a Platform Web 2.0 Cloud Mobility Z. Li, Multimedia Communciation, Spring 2017 p.27

28 Current HTTP 1.1 HTTP 1.1 Features Persistent TCP connection Client side state info via Cookie Cache support RFC 2616 Web client (browser) HTTP request HTTP response (content) Web server Z. Li, Multimedia Communciation, Spring 2017 p.28

29 URL Universal Resource Locator Each file managed by a server has a unique name called a URL (Universal Resource Locator) URLs for static content: o Identifies a file called index.html, managed by a Web server at that is listening on port 80. URLs for dynamic content: o Identifies an executable file called adder, managed by a Web server at that is listening on port 8000, that should be called with two argument strings: and 213. Z. Li, Multimedia Communciation, Spring 2017 p.29

30 HTTP 1.1 Request Example HTTP GET method Client: Server: Z. Li, Multimedia Communciation, Spring 2017 p.30

31 HTTP 1.1 Request via Proxy Client: Proxy: Z. Li, Multimedia Communciation, Spring 2017 p.31

32 Content distribution networks (CDN) 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 Z. Li, Multimedia Communciation, Spring 2017 p.32

33 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 o close to users, minimizing RTT o used by Akamai, 1700 locations bring home: smaller number (10 s) of larger clusters in POPs near (but not within) access networks o used by Limelight Z. Li, Multimedia Communciation, Spring 2017 p.33

34 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 request video from 5 KINGCDN server, streamed via HTTP 3. netcinema s DNS returns URL 3 netcinema.com 2. 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 Z. Li, Multimedia Communciation, Spring 2017 p.34

35 CDN Edge Server 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 Z. Li, Multimedia Communciation, Spring 2017 p.35

36 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: o Netflix uploads studio master to Amazon cloud o create multiple version of movie (different encodings) in cloud o upload versions from cloud to CDNs o Cloud hosts Netflix web pages for user browsing three 3 rd party CDNs host/stream Netflix content: Akamai, Limelight, Level-3 Z. Li, Multimedia Communciation, Spring 2017 p.36

37 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 Z. Li, Multimedia Communciation, Spring 2017 p.37

38 Outline ReCap Lecture 16 Multimedia Transport Managed IP Networks Media Transport: UDP/RTP/RTSP/RSVP Over The Top (OTT) Transport: HTTP, Webcoket Emerging Solutions: WebRTC/QUIC Summary Z. Li, Multimedia Communciation, Spring 2017 p.38

39 WebRTC WebRTC is a browser embedded native audio/visual real time streaming solution Built on top of RTP Have firewall traversal support Widely deployed in Chrome and Firefox Main Utilities: MediaStreams access to user's camera and mic PeerConnection audio/video calls DataChannels p2p application data transfer More to come in RMCAT coverage! Z. Li, Multimedia Communciation, Spring 2017 p.39

40 QUIC Motivation HTTP Problems Single HTTP Request per TCP connection: highly inefficient due to lack of parallelism. Most browsers bypass by having multiple TCP connections. Client driven, pull based, server has no initiative. HTTP messaging overhead: uncompressed text message, overhead becomes very big. (similar to the DASH /WebSocket) Underlying TCP slow start: 3 RTT for TLS HTTPS connection to start Under-utilization: TCP traffic congestion window control slow recovery from congestion Z. Li, Multimedia Communciation, Spring 2017 p.40

41 SPDY Work SPDY is built on-top of TCP Main SPDY Features Allow many concurrent HTTP requests to run across a single TCP session. Initial TCP cwnd should be 16 Compression: o All data is compressed, Includes headers, Redundand data is removed Reduce the bandwidth currently used by HTTP by compressing headers and eliminating unnecessary headers. Enable the server to initiate communications with the client and push data to the client whenever possible. Make SSL the underlying transport protocol, for better security and compatibility with existing network infrastructure. Although SSL does introduce a latency penalty, we believe that the long-term future of the web depends on a secure network connection. In addition, the use of SSL is necessary to ensure that communication across existing proxies is not broken. Z. Li, Multimedia Communciation, Spring 2017 p.41

42 SPDY Problems Still works on top of a TCP connection Slow start (mitigated by changing init cwnd size to 16) Head of Line (HOL) blocking: o Another disadvantage of SPDY is that an out-of-order packet delivery for TCP induces head of line blocking for all the SPDY streams multiplexed on that TCP connection. Connection Latency: 3 RTT to establish a secure link Under utilization of link capacity by TCP Rate Control no loss, in order delivery, not that a big deal for media data (we have CTS/DTS) Z. Li, Multimedia Communciation, Spring 2017 p.42

43 QUIC Quick UDP Internet Connection Main QUIC Features/Design Goals: Connection establishment latency Improved congestion control more suited for media QoE Multiplexing without head-of-line blocking Forward Error Correction (FEC) reduce delay. Connection migration: native support for multipath via CID (Connection ID) Z. Li, Multimedia Communciation, Spring 2017 p.43

44 QUIC Design Goals Backwards compatible with today s Internet Low Latency in connection and response Reliable Stream support Yet remove HOL blocking Better Congestion Avoidance Better congestion measure and modeling Better pacing of transmission with APP layer QoE/QoS mapping info Works with new mobile network features E.g, CID based multiple path support Privacy & Security: comparable to TLS Z. Li, Multimedia Communciation, Spring 2017 p.44

45 Summary Managed IP Networks Media Transport RTP: media transport RTCP: congestion control, QoS/QoE feedback vehicle RTSP: remote controller for user interaction with the system Mature, but need IP layer support Over The Top (OTT) Media Transport HTTP: de-facto software infrastructure, widely deployed and supported CDN and HTTP Cache: localizing the traffic. Reducing the RTT, by replicating the content to CDN edge servers SPDY: aggregating HTTP requests over a single TCP, header compression. Server Push/Hint. A Google thing. WebSocket: two way transport, aggregating requests, push. Widely supported. New Media Transport: WebRTC: browser integrated naïve audio/visual streaming solution, on top of RTP QUIC: web over UDP, A google project. Early stage. Z. Li, Multimedia Communciation, Spring 2017 p.45

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