Olaifa, John O . 135776 01 April, 2014

1. Eastern Mediterranean University Faculty of Engineering Department of Computer Engineering Voice over Internet Protocol (VoIP) Olaifa, John O. 135776 01 April, 2014 Presented to: • Assoc. Prof. Dr. Doğu Arifler by:

2. Outline • What is VoIP? • Why UDP, not TCP? • RTP • RTCP • Challenges in VoIP • SIP • Recap

3. What is VoIP? • VoIP is an IP telephony term for a set of hardware and software that enable the transmission of voice packets using Internet Protocol.

4. Why VoIP? • VoIP provides features that are essential for • Conference Calls • Real time audio transmission, • Media Streaming.

5. How VoIP works • a. A Signaling Protocolis used to establish a communication session. • b. Codecs convert analog voice to digital signal. • c. Bits are compressed for transmission and encapsulated. • d. Data is moved between endpoints using a Media Protocol and codec converts back to analog signal. • e. Signaling Protocol tears down the communication session.

6. Transport Layer Services and Protocols TheTransport Layer provides process – process data transfer essential for VoIP communication: • UDP –User Datagram Protocol • TCP – Transmission Control Protocol

7. Application Services and Protocols The Application Layer provides protocols essential for VoIP communication: • RTP – Real-time Transport Protocol • RTCP –Real-time Control Protocol • SIP –Session Initiation Protocol

8. Outline • What is VoIP? • Why UDP, not TCP? • RTP • RTCP • Challenges in VoIP • SIP • Recap

9. Why UDP and not TCP? Although, TCP provides the following valuable services, Reliable transport -No lost packets Congestion control -Prevention of congestions Flow control -Reduction of overflows It is not the best for VoIP because of: Handshaking, flow control, and windowing delays Unsuitability for multicast packets

10. Why UDP and not TCP? (contn’d) • High delivery speed • Support for bursty data • Time stamping • Sequence Numbering* • Small header size In the light of TCP weaknesses, UDP has come to stay as it offers

11. Voice packet; a closer look 20bytes 20bytes 14bytes 12bytes 8bytes • The link header contains the link layer information. • The IP header contains the Network details. • The UDP header contains the Source port, Checksum, Destination port, Length. • The RTP header contains port details, sequence no, time stamp. • Voice payload contains audio communication

12. Real-Time Transport Protocol (RTP) End-to-end delivery services for applications transmitting real-time data, such as audio and video over multicast and unicast networks. Typically used over IP and UDP Layer 1 – RTP Layer 2 – UDP Layer 3 – IP

13. RTP RTP picks a random even port from the UDP port range 16384-32767 for voice communication. RTP Header fields RTP RTP Internet Receiver Sender 16384 16386 RTP has a sister protocol RTCP which uses RTP port +1

14. RTP header tear down • Sequence Numbering • – to report PDU loss • to report PDU reordering • to perform out-of-order decoding • Time stamping • for play out • for jitter and delay calculation • Payload type identification • for media interpretation • Applications that use RTP are • Less sensitive to packet loss. • Very sensitive to packet delays.

15. RTP • All RTP and RTCP PDUs are sent to same multicast group (by all participants) • RTP supports several file formats like PCM, GSM, MP3, MPEG 1 and H.261. • RTP also has jitter compensation, multimedia streaming (tolerates packet loss) and detection of out of sequence arrival in data attributes.

16. Outline • What is VoIP? • Why UDP, not TCP? • RTP • RTCP • Challenges in VoIP • SIP • Recap

17. RTCP • RTCP provides; • Out-of band services, reception report and control information. Uses the next odd number port after RTP. RTCP RTCP Internet Receiver Sender 16385 16387 • Time stamp and wall clock of last RTP packet • Number of packets in the stream • Number of bytes sent in the stream

18. Outline • What is VoIP? • Why UDP, not TCP? • RTP • RTCP • Challenges in VoIP • SIP • Recap

19. Challenges in VoIP • Jitters: • Time fluctuation between sending and reception of packets. • Packet loss: • Non delivery caused by buffer overfill, Signal degradation, multipath fading. • End-to-end delay: • Accumulation of transmission, processing, queuing, propagation, end –system processing delay. • Eavesdropping: • Interception and playback of voice packets during transmission

20. Jitters S R R S Time Time Fluctuations in inter packets timing introduced between sending and receiving end devices.

21. 30 seconds Ideal Timing 00.00.00 00.00.10 First RTP-PDU 00.00.11 00.00.20 Second RTP-PDU 00.00.21 00.00.30 Third RTP-PDU 00.00.31 Send time Play time

22. Reality 00.00.00 Jitters 00.00.10 First RTP-PDU 00.00.11 00.00.20 Second RTP-PDU 00.00.21 00.00.30 00.00.25 00.00.40 Third RTP-PDU 00.00.35 00.00.37 Fourth RTP-PDU 00.00.41 Send time 00.00.47 00.00.51 Play time

23. Eliminating Jitters • Normalization of Voice packets before playback could be achieved by: • Prefacing each chunk with a Time stamp • Prefacing each chunk with a Sequence number • Play out delay • Adaptive delay • Fixed delay

24. Eliminating Jitters (Contnd.) 00.00.00 00.00.10 First RTP-PDU(0) application 00.00.11 Second RTP-PDU(10) 00.00.20 00.00.21 00.00.18 00.00.30 00.00.25 00.00.28 00.00.40 Third RTP-PDU(20) 00.00.35 00.00.37 00.00.38 Fourth RTP-PDU (30) 00.00.41 00.00.48 Send time 00.00.51 00.00.58 Play time

25. VoIP and packet loss Voice communication is tolerant of packet loss unlike most other IP services, tolerating a packet loss between 1-20% Encoder Encapsulator Transport Control Bobby Decoder Decapsulator Transport Control Billy Network a corrupted frame

26. VoIP and packet loss (Contnd.) Packet loss is caused by: Buffer overfill Signal degradation (noise & distortion) Attenuation Multipath fading.

27. Loss Concealment Packet loss could be concealed in any of the following ways. • Forward Error Control (FEC) • Interleaving • Receiver based repair • Packet repetition • Packet replacement

28. Packet Delay End-to-end delays are the accumulation of delays encountered by the voice packet as it heads for its destination. DelayE-E = delaytrans+delayproc+delayqueue+delayprop…. 150---400 ms =Tolerable >400 ms =Not tolerable (discarded)

29. Outline • What is VoIP? • Why UDP, not TCP? • RTP • RTCP • Challenges in VoIP • SIP • Recap

30. Session Initiation Protocol Sets up and tears down communication sessions. 1. Caller sends an “invite” 2. Registrar receives invite and forwards it to callee’s IP address 3. Callee accepts and SIP hands over to RTP

31. Outline 31 • What is VoIP? • Why UDP, not TCP? • RTP • RTCP • Challenges in VoIP • SIP • Recap

32. Recap VoIP transmits audio signals using the Internet Protocol. The UDP is employed in VoIP for its speed and support for bursty data. VoIP uses SIP, RTP and RTCP in tandem with UDP. RTP and RTCP use contiguous even and odd UDP ports respectively SIP creates the session, RTP and RTCP take over while SIP ends it. Delays are almost unavoidable but could be managed. In conclusion, we said that;

33. Basic header Ethereal capture for RTP-PDU

34. Questions ?

35. Thank you.