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Understanding Voice Over IP: Protocols and Challenges in Modern Communication

This document explores Voice Over IP (VoIP) technology, which facilitates voice communication over the Internet Protocol (IP). It delves into foundational aspects, including the meaning and benefits of VoIP, such as reduced costs and the integration of voice and data services. The challenges of managing speech quality, traffic, and reliability are also examined. Key protocols like RTP, RTCP, SIP, and MEGACO are discussed, highlighting their roles in signaling and data transport. An illustrative example of Skype demonstrates practical applications and limitations within VoIP systems.

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Understanding Voice Over IP: Protocols and Challenges in Modern Communication

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  1. Voice over IP B90901033 林與絜

  2. Outline • Introduction • Some Protocols • Example - Skype • Conclusion

  3. Introduction • What is VoIP? • The transport of voice traffic using the Internet Protocol (IP)

  4. Introduction (cont.) • Why VoIP? • Lower cost • The widespread availability of IP • Reduced bandwidth • Integration of voice and data applications • New service features

  5. Introduction (cont.) • VoIP Challenges • Speech quality • Managing access and prioritizing traffic • Speech-coding techniques • Network reliability and scalability

  6. Introduction (cont.) • Speech Quality • Delay • Jitter • Packet Loss • Call Set-up Time

  7. Outline • Introduction • Some Protocols • Example - Skype • Conclusion

  8. Some Protocols • RTP and RTCP • For data transport • SIP • For signaling use • MEGACO • Between signal and data

  9. RTP and RTCP • RTP (Real-Time Transport Protocol) • A transport protocol for real-time applications • RTCP (RTP Control Protocol) • A companion protocol with RTP

  10. RTP and RTCP (cont.) • Voice over UDP, not TCP • Data traffic • Asynchronous • Extremely error sensitive • Voice traffic • Synchronous – stringent delay requirements • Tolerant of errors – at most 5%

  11. RTP and RTCP (cont.) • RTP over UDP • Sequence number • Timestamp • Payload type, marker, etc. • Does not solve the QoS problems; simply provides additional information

  12. RTP and RTCP (cont.) • RTCP • Exchange messages between session users • Quality feedback • Number of lost packets • Delay • Inter-arrival jitter • Implicitly open when an RTP session is open

  13. SIP • SIP (Session Initiation Protocol) • A signaling protocol • Setup, modification, tear-down of mutimedia sessions • A powerful alternative to H.323 • More flexible, simpler

  14. SIP (cont.) • SIP Network Entities • User agents • User agent client • User agent server • Servers • Proxy server • Location server (Registrar) • Redirect server

  15. SIP (cont.) • SIP Messaging • Text-based • SIP Message: • Start line • Request or status • Message headers • Additional information of the request or response • Message Body • Describe the type of session

  16. MEGACO • MEGACO (Media Gateway Control Protocol) • Network Gateway • Signaling conversion • Media conversion

  17. MEGACO (cont.) • MGC (Media Gateway Controller) • Handling call control • Call-control intelligence • Call-related signaling • MG (Media Gateway) • Performing the media conversion • A line or trunk on circuit-switched side • An RTP port on the IP side

  18. MEGACO (cont.)

  19. Outline • Introduction • Some Protocols • Example - Skype • Conclusion

  20. Skype • A peer-to-peer VoIP client developed by KaZaa in 2003 • Allowing its users to place voice calls and send text messages to other users

  21. Skype (cont.)

  22. Skype (cont.)

  23. Skype (cont.)

  24. Skype (cont.) • Advantages • It can work seamlessly across NATs and firewalls • Better voice quality than the MSN and Yahoo IM applications

  25. Skype (cont.) • Disadvantages • The protocol is proprietary • It provides a single service, not an architecture of new services • It still has centralized elements for login authentication

  26. Outline • Introduction • Some Protocols • Example - Skype • Conclusion

  27. Conclusion • Some Topics • QoS Management & Improvement • Mobility • Teleconferencing System

  28. Reference • Daniel Collins, Carrier Grade Voice over IP, McGraw-Hill, 2003 • Salman A. Baset and Henning Schulzrinne, An Analysis of the Skype Peer-to-Peer Internet Telephony Protocol, 2004 • http://www.skype.com/

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