Part 3: Lecture 3 Content and multimedia Internet traffic Multimedia How can multimedia be transmitted? Interactive/real-time Streaming 1
Voice over IP Interactive multimedia Voice and multimedia sessions over the Internet/ IP What s the main advantage? Lower costs than on PSTN. Devices Real-time interactive applications Softphones a software program for making telephone calls over the Internet VoIP phones PC to phone PC to PC In the future: Netbooks and smartphones, that support 5G handoff (IEEE 802.21) Allow roaming between 802.11 networks and (3G) cellular networks. Videoconference With webcam 2
Video conferencing Session control How do we start the communication? Signaling. Reduced travel costs Easier access to remote experts More personal than a voice conference Signaling protocols provide all information needed to setup, control and teardown of VoIP calls. UDP and TCP are used to transport the signaling messages. VoIP signaling protocols? SIP, H.323, Skype. In the PSTN SS7 TDM Call management Transport of voice/video Interactive multimedia protocols Call management SDP In the Internet Transport of voice/video SIP H.323 RTP RTCP TCP/SCTP RSVP UDP IPv4/IPv6 3
Session Initiation Protocol SIP is described in RFC 3261 (2002) An application layer protocol used instead of SS7 to initiate and terminate voice and video calls over IP networks (VoIP) It is independent of the underlying transport protocol It can run on UDP, TCP, SCTP, RTP Call Control Application Media Application Session management: Transferring session (call Forwarding); modifying Parameters. SIP supports: User location: Correct device to which to communicate User availability: User willing and able to participate SIP RTCP RTP TCP SCTP UDP Session setup: Establish sessions parameters, e.g. port numbers User capabilities: Choice of media and coding schemes H.323 H.323 is another signaling protocol for real-time, interactive applications. 1996: ITU-T1 published Version 1 of Recommendation H.323 in Visual Telephone Systems and Equipment for LANs which provide a non-guaranteed Quality of Service not designed for the Internet only local calls, small number of users 2009: current version Operates well on WANs Widely adopted also in large installations Multimedia data transfer 4
Cumulative data 23/03/18 Streaming applications Sampling at constant rate Which problems will you have? (How will transport the data over the network?) Client-server model Streaming stored video: Server holds the content Client requests content, plays it as it downloads, can rewind/ pause Require: Timestamping: when was the data produced in relation to when it was received? Indication of packet loss: are we losing packets and can we do anything to less congestion? Indication of frame boundaries Identification of clients Good use of bandwidth 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 5
How does delay occur? Four sources of packet delays transmission A propagation packet needs to be processed packet needs to be transmitted B nodal processing queueing A B packet needs to travel on link packets needs to wait its turn queueing d trans : transmission delay: L: packet length (bits) R: link bandwidth (bps) d trans = L/R d trans and d prop very different d = d proc + d queue + d trans + d prop d prop : propagation delay: d: length of physical link s: propagation speed in medium (~2x10 8 m/sec in optical fiber) d prop = d/s Four sources of packet delay transmission A propagation End-to-end delay B nodal processing queueing d 1,R 1 d 2,R 2 d 3,R 3 d 4,R 4 d proc : nodal processing check bit errors determine output link typically < msec d = d proc + d queue + d trans + d prop d queue : queueing delay time waiting at output link for transmission depends on congestion level of router d = ( L i R i + d i s +Q i (t)) 6
Cumulative data 23/03/18 Q(t) Q(t) and playout buffers Max tolerated delay Fraction of packets (PDF) Fraction of packets (PDF) Related to the size of the playout buffer at the receiving end Delay (msec) Delay (msec) Streaming stored video: revisited Playout buffer constant bit rate video transmission variable network delay client video reception buffered video constant bit rate video playout at client fill rate x(t) client buffer drain rate d client playout delay time client-side buffering and playout delay: compensate for network-added delay, delay jitter pre-fetched video 7
Loss tolerance 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: 150 ms network loss: IP datagram lost due to network congestion (router buffer overflow) RTP loss tolerance: depending on voice encoding, losses concealed, packet loss rates between 1% and 10% can be tolerated. Real-time Transport Protocol RTP is a top-up transport protocol used for real-time applications. Think of delivery of voice and video data: Lightweight. One single message Runs over another transport protocol It support multicast. Accompanied by RTCP - RTP Control Protocol: A management protocol Allows endpoints to exchange information about data flows Used by RTP to determine how to tune its behavior RTP features Runs on top of UDP: no guarantee of reliability no guarantee of packet ordering Payload contains the real-time media Header contains information related to the payload: the source, size, encoding type It uses timestamps, sequence numbering, and delivery confirmation for each packet sent. It supports error-correction schemes for increased robustness and basic security options for encrypting packets. 8
RTP packet generation RTP header 16 bits 16 bits V P X CC M Payload type Sequence number Timestamp Synchronization Source Identifier First Contributing Source Identifier Second Contributing Source Identifier Last Contributing Source Identifier Sequence number Timestamp RTP receiver A simple counter that starts at a random value when a session begins; It increments by one with each RTP message sent in that session; It provides a mechanism for the receiver to resequence packets that arrive out of order and to detect missing elements. It carries the time index for the first sample of the RTP message. The receiver can use this field to reassemble the information stream with the appropriate timing. Streams list maintained via RTCP Source: RTP Audio and Video for the Internet (C. Perkins) 9
RTCP functions 1. Integrated media synchronization i.e. when video and audio are transmitted on different streams RTCP 2. QoS reports i.e. number of lost packets, jitters 3. Participation reports i.e. when a participant is leaving the call 4. Participation details i.e. information about the source, email address, sender names. RTP RTCP Internet RTCP Overhead traffic 6976 6977 Internet 6970 6971 RTCP sender RTP RTCP RTCP receivers each RTP session: typically a single multicast address; all RTP /RTCP packets belonging to session use multicast address RTP, RTCP packets distinguished from each other via distinct port numbers. RTCP port number = RTP port number +1 To limit traffic, each participant reduces RTCP traffic as number of conference participants increases. Typically 5% of the session bandwidth. 10
RTCP message types SDES (Sender Descriptor) message: used by an application to join an RTP session; BYE message: used when an application leaves the session; SR (Sender Report) message RR (Receiver Report) message allow traffic monitoring; Adaptive bit rate streaming APP (application) message Streaming over HTTP Progressive download Old thinking: TCP hurts multimedia transmission. We need buffering (a-la RTP). à push methods (the content is pushed from the server to the client) New insights: 1. TCP(and HTTP over TCP) passes more easily through firewalls and NAT boxes. 2. TCP congestion control does not hurt clients if they can adapt to bandwidth variations. Treat video as a regular file and starts to play as soon as enough as been received. à Pull methods (where the clients controls the speed of receival) 11
Adaptive streaming CDNs Content delivery Clients: content providers The CDN companies provide a distributed infrastructure that can more efficiently distribute the content they host. 12
Protocols Proprietary: Microsoft Smooth Streaming Apple HTTP Live Streaming Adobe HTTP Adaptive Streaming Standard: DASH Test Named Data Networking NDN Named Data Networking It has its origin in the CCN - Content Centric Networking -efforts from van Jacobson (2006). Van Jacobson presented his ideas in a talk of 2006at a Google Summit (video) We need a new architecture (and new hardware) that supports the current actual use of the Internet. 13
IP networks vs content networks Network prefix Content name The original Internet was primarily about sharing resources. The current Internet is about accessing content Destination Next hop 136.136.136.0/24 Router C 136.136.136.0/24 Destination Next hop os3.nl/an.html Router C os3.nl/an.html A E D 136.136.136.0/24 F A E D Os3.nl/an.html F B C B C Information Centric Networking (ICN) Four main focuses in the ICN approach: 1. It fundamentally decouples information from its sources, by means of a clear location-identity split: The name identifies information (content) and NOT its location 2. It solves the problem of flash crowds (sudden request of content): it exploits in-network storage (caches) and in-network informationawareness to optimally select location. 3. It allows for mobile users: it defines a publish-subscribe model for expressing availability of content and interest in the content. 4. The open internet creates an inbalance between sender and receiver: ICNs guarantee that a receiver will only get content it requested. 14
Packets Interest Interest Interest Content Interest Content Content Content Consumer send Interest Packets into the network. Producer/content holders send back Data Packets NDN router architecture Interest pkt processing 15
Data pkt processing Named Data is Easy to Secure In the Internet you secure your path....but the server may still be hacked In NDN you sign the data with a digital signature....so the users know when they get bad data Content Poisoning /youtube/video /youtube/video Analogous to Prefix Hijack in IP.. except that routers can detect it Slide from C. Papadopoulos - U Colorado Slide from C. Papadopoulos - U Colorado 16
DDoS Attacks Literature target - Cannot send unsolicited data.. but can flood Interests - Network can throttle unanswered Interests Chapter 7 Multimedia networking (for SIP and H.323) Everything about RTP Slide from C. Papadopoulos - U Colorado Awesome IT Preparation for the exam The Course material page on the Wiki is the reference for the exam. Exam is open book. 2.5 hours. See you Thursday March 29 th. 10.30-13.00 Example exam from last years (2016 and 2017) to be found online. Something _specific_ unclear: Make an appointment Send an email with a question 17