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6. Advanced Network Applications

6. Advanced Network Applications. 6.1 VoiceOverIP and SIP 6.2 Video over Peer-to-Peer Networks. 6.1 VoiceOverIP and SIP. Efficient realization of telephony since packet switching over IP involves a lower overhead and more flexibility than line-switched (circuit-switched) telephony.

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6. Advanced Network Applications

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  1. 6. Advanced Network Applications 6.1 VoiceOverIP and SIP 6.2 Video over Peer-to-Peer Networks

  2. 6.1 VoiceOverIP and SIP • Efficient realization of telephony since packet switching over IP involves a lower overhead and more flexibility than line-switched (circuit-switched) telephony. • Goal: Integration of telephony with multimedia communication and intelligent ser-vices in IP-based networks. • Fundamentals and Protocols * I wouldliketothank Dr. Robert Dendaand Dr. Andreas Grebe forlettingmeusesomeoftheirslides in thissubchapter..

  3. Architecture PSTN = Public Switched Telephone Network

  4. Reasons for IP Telephony (1) • Conventional telephone network: connection-oriented, line-switched, complex switching equipment (PBXs) • IP telephony: • packet-switched, statistical multiplexing gain, simple routers rather than com-plexswitches • low bandwidth due to audio compression (e.g., G.723.1 at only 5,3 -6,3 kbit/s, compared to 64 kbit/s PCM with ISDN) • Flexible signaling over a packet-switched network • Integration of multimedia data easy • Easy to upgrade with new Intelligent Network services (call forwarding, call wai-ting, multiparty connections, black lists, etc.) • Scalability of the communication service quality • Many types of terminal equipment possible: PC, IP telephone, audio/ video-conferencing equipment, etc. • Use of the existing data network is a major advantage.

  5. Reasons for IP Telephony (2) • Motivation for the user • Cost advantages • Only one network needs to be administered. • PC integration is often useful.

  6. Requirements for IP Telephony (1) • Main problem: Quality of Service • Delay Experimentally measured delays • Coding/Decoding delay:approx. 30 ms (G.729A) - 82 ms (G.723.1) • Network delay (transmission, routing, etc.): Gateway - Gateway: 30 - 100 msPC - PC: 50 - 140 ms • Access delay (operating system, sound and video cards, DSPs, ... ):Gateway - Gateway: 40 - 80 msPC - PC: 100 - 340 msBut: human ear sensitive in this regard, desirable end-to-end delay: < 100 ms • Packet loss: FEC desirable, increases delay and data rate • Multicast: Heterogeneity of the participants, dynamic join and leave might require adaptive QoS mechanisms in the network.

  7. Requirements for IP Telephony (2) • Intelligent Network services • "standard" IN/AIN services: call-waiting (“knocking”), answering machines in the network, three-way calls, blacklisting, etc. • New intelligent services: directory services (built-in yellow pages), call center support (e.g., customer record shows up on the screen when an incoming call is picked up), WWW interfaces, etc. • Signaling • lightweight signaling (Internet vs. IN/AIN) • new media, extended services, charging requires new signaling mechanisms

  8. RTP UDP/IP/LAN Interface System Control H.245 Control Q.931 Call Setup RAS Gatekeeper Interface IP Telephony Standards: ITU Recommendation H.323 ScopeofRecommendation H.323 Audio Codec G.711 G.723 G.729 Microphone / Speaker Video Codec H.261 H.263, H.264 Camera / Display Data Interface T.120 Data Equipment System Control User Interface

  9. H.323 • Version 1 of H.323 was adopted by the ITU-T in 1996, version 2 in January 1998. H.323 is supported by most products for Voice over IP (VoIP) and includes both con-ventionaltelephone devices and PCs. H.323 covers call control, multimedia and bandwidth management and defines the interfaces between LANs and other net-works.

  10. Protocols for IP Telephony (1) • MCU = Multipoint Control Unit • N-ISDN = Narrowband ISDN • B-ISDN= Broadband ISDN

  11. Protocols for IP Telephony (2) • Signaling Protocols • Control Protocol for Multimedia Communication (H.245): End-to-End Signaling, very complex • Session Initiation Protocol (SIP): a simple text-based signaling protocol for Inter-net conferences and telephony. Very widely used today. • Digital Subscriber Signaling System No.1 (DSS1, Q.931), Q.93B, Q.932: ISDN signaling protocols • RSVP – Resource ReSerVation Protocol: a receiver-oriented protocol for QoS and bandwidth reservation. A widely referenced research proposal, but it never really made it into practical use.

  12. Protocols for IP Telephony (3) • Media stream control protocols • Real-Time Protocol (RTP) / Real-Time Control Protocol (RTCP): Internet protocols. Support multimedia communication in real time. Profiles are defined for different payload types. • Real-Time Streaming Protocol (RTSP): allows bi-directional transmission based on RTP with VCR-style control functions. • "Intelligent Network" (IN, from telephony) • PSTN: Intelligent Network (IN), Advanced Intelligent Network (AIN) • IP Telephony: Telecommunication Information Networking Architecture (TINA) • Field of current research: use of modern distributed systems architecture (Java RMI, DCOM), active networks and mobile agents for the implementation of more “network intelligence”.

  13. continuous Internet Internet- continuous voice Packets Packets voice IP-Network 1. Analog/Digital Data transfer 1. Analog/Digital 1. Depacketing 1. Depaketierung 2. Voice Coding/ UDP/IP - TCP/IP 2. Kodierung / 2. Decoding 2. Dekodierung Data Reduction LL - FR - ATM - LAN Datenreduktion 3. Digital/Analog 3. Digital/Analog 3. Packeting SDH - PDH 3. Paketierung IP Backbone / Internet Intranet ( No QOS) (QOS) Principle of Voice over IP

  14. Distortion of VoIP Codecs • Codecsshow strongly varying distortion characteristics. • Psycho-acoustic codecs clearly show a stronger distortion. • Mean Opinion Score (MOS) in both cases acceptable.

  15. Power con-sumption (100 MHz Pentium) Quality Coding Method Bit rate Standard Use Codec (MOS) Delay PCM 64 kbit/s G.711 4,0 H.323 ISDN < 1 ms < 1% PSTN ACELP 5,3 kbit/s 3,88 H.324/H.323 97,5 ms 35-49% Video telephone G.723.1 MP-MLQ 6,3 kbit/s 3,88 97,5 ms 35-49% Voice over IP G.728 LD-CELP 16 kbit/s 3,93 H.323 Voice over IP 3 ms approx. 65% Voice over IP H.323 G.729 CS-ACEL 8 kbit/s 3,90 35 ms approx. 50% Frame Relay ATM real-time 3,80 at not included coding at GSM 6.10 RELP 13 kbit/s Mobil ca. 40 ms 0% errors in H.323 486PC 66MHz not included Lucent approx. 13,5% 7,3 kbit/s 3,88 35 ms Voice over IP CS-ACEL in H.323 SX7300P Comparison of Different VoIP Codecs MOS (Mean Opinion Score): opinion of test persons.> 3.8 acceptable quality> 4.0 very good quality

  16. SIP - The Session Initiation Protocol • H.323 was designed by ITU. Many people in the Internet community saw it as a typical telco product: large, complex, and inflexible. Consequently, IETF set up a committee to design a simpler and more modular way to do voice over IP. The major result is SIP (Session Initiation Protocol). • It defines telephone numbers are URLs, two-party sessions, multiparty sessions and multicast sessions. • SIP just handles setup, management, and termination of sessions. Other protocols, such as RTP/RTCP, are used for streaming data transport. • SIP is an application-layer protocol. It can run over UDP or TCP, as re-quired.

  17. SIP “Methods” • The SIP protocol is a text-based protocol modeled after HTTP.

  18. The Status Code Response Groups The callee also may supply information about its capabilities, media types and formats. Connection is done using a three-way handshake. The caller responds with an ACK message to finish the protocol and confirm receipt of the “200” message. Either party may request termination of a session by sending a message with the BYE method. When the other side acknowledges it the session is termi-nated. The OPTIONS method is used to query a machine about its own capabilities.

  19. Use of a Proxy Server and Redirection with SIP • The REGISTER method relates to SIP‘s ability to track down and connect to a user who is away from home. This message is sent to a SIP location server keeping track of who is where. That server can later be queried to find the user‘s current location. The operation of redirection is illustrated below. Location Server 3 REPLY 2 LOOKUP 4 INVITE 1 INVITE Proxy Caller Callee 6 OK 5 OK 8 ACK 7 ACK 9 DATA

  20. Comparison of H.323 and SIP (1) • Both H.323 and SIP allow two-party and multiparty calls using both computers and telephones as end points. Both support parameter negotiation, encryption and the RTP/RTCP protocols. A summary of their similarities and differences is shown on the next page.

  21. Comparison of H.323 and SIP (2)

  22. Comparison of H.323 and SIP (3) • H.323 is a typical, heavyweight, telephone-industry standard, specifying the complete protocol stack and defining precisely what is allowed and what is forbidden. This approach leads to very well defined protocols in each layer, easing the task of inter-operability. • In contrast, SIP is a typical Internet protocol that works by exchanging short lines of ASCII text. It is a lightweight protocol that works well with other Internet protocols. • The downside is that is has suffered from ongoing interoperability problems as people try to interpret what the standard really means.

  23. source receivers 6.2 Video over Peer-to-Peer Networks Live Streaming (or Internet Television, IP-TV) consistsofdistributing a video/audiosequencefrom a sourceto a setofreceivers, whicharesupposedtoplayit back whileitisreceived. Traditionally, live streaminghasbeenimplementedwith a client-serverarchitecturewheretheserverischargedwithprovidingthestreamdatato all thereceivers (multi-pleunicast).

  24. Problem Statement • Why P2P live streaming? • Solvestheissueofprovisioningserverinfrastructures. • Theoreticalself-scalabilityasusersprovidenewresources. • The systemcapacitygrowswiththenumberofusers. • Unboundedgrowthifuserscontributeasmuchastheyconsume. • Economicinterest: reducethecostofmediadistribution • The sourcerequiresonly a very limited capacity. • Easy todeploycomparedto inter-domain native multicast. • Does not requirechanges in the Internet core.

  25. Live Streaming vs. Video On-Demand • Video On-Demand (VoD) • Involvesfinite-sizedmediafiles • Files are not sensitive todelay • Aimsto just anticipatethestartofplayback. • Live streaming • involves“infinite” datastreams(thesourceconstantlysendsnewdata) • live mediastreamsarevery sensitive todelay • aimstoalwaysattainan acceptablere-ception. Live streamingis a toughproblem due totheunknownsizeofthemediaandthe large variabilityofnetworkconditi-ons.

  26. PPLive • A peer-to-peervideostreamingsystemdeveloped in China • Mainlyusedtotransmit TV chanels (P2PTV) • Based on BitTorrent, withmodifiedprioritiesand a modifiedtit-for-tat strategy; detailsare not published • Verysuccessful in China. In Germany mainlyusedforsportsevents.

  27. CoolStreaming • Also a peer-to-peer live streamingsystemfortelevisioncontent (P2PTV) • Verypopular in North America, morethanonemillionusers online • Also based on BitTorrenttechnology, withmodifications • Containssophisticatedalgorithmsto deal withthe different bandwidthsatthepeers

  28. Conclusion • Peer-to-peer live streamingis an attractivetechnologyforthedistributionof TV con-tent, in particularwhen IP Multicast is not available. • None oftheavailablesystemsintegrate a Scalable Video Codec (MPEG-4 SVC) for optimal adaptationto different peerbandwidths, orstereovideo

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