1. CS644 Advanced Topics in Networking VoIP Wed, 2004/9/8

2. 2 VoIP Basics Codecs Performance issues –Delay, loss, quality –Other subjective/objective measures Available tools

3. 3 History of VoIP 1980s –Packet voice over satellite links in Lincoln Lab Early 1990s –MBone popularized audio and video conferencing tools –Nevot, vat, vic, rat Late 1990s –Commercialization of VoIP

4. 4 Digitizing Voice 쏼 ~~ 라 쏴 ~ ㄹ ~~ ㄹ ~~~~~~~ 라 ~~ Time

5. 5 Talkspurts Talk and Silence in Voice 쏼 ~~ 라 쏴 ~ 라 ~~ Time Silence period

6. 6 Voice to Packets 쏼 ~~ 라 쏴 ~ 라 ~~ Time Look-ahead

7. 7 Popular Waveform Codecs PCM (Pulse Code Modulation) –G.711 –Audible frequency (300-3KHz) –By Nyquist theorem, sample 8bits at 8KHz = 64 Kbps ADPCM (Adaptive Differential PCM) –G.726 –Sample differences: 4bit at 8KHz

8. 8 Alternative for compression companding –non-linear quantization: μ-law (G.711) waveform –exploit statistical correlation model –model voice, extract parameters subband –split signal into bands and cod e individually (MPEG audio) exploit masking properties of human ear

9. 9 Other Codecs G.729 for mobile telephony –Conjugate structure, algebraic-code-excited linear prediction (CS- ACELP) –80bits at 100Hz = 8 kbps –requires 40% of 100MHz Pentium –G.729A: reduced-complexity version G.723.1 for videotelephony –MP-MLQ (Multipulse, multilevel quantization) –30ms blocks of 240 16-bit samples into 24B = 6.3kbps –requires 25% of 100MHz Pentium GSM full rate –GSM 06.10, 13Kbps –Regular Pulse Excited (RPE) codec –GSM 06.20 = half-rate

10. 10 Basic Attributes of Codecs Bit rate –economy of bandwidth –in multimedia applications, more b/w to video Silence compression –voice activity detector (VAD) –discontinuous transmission (DTX) –comfort noise generator (CNG) Complexity –limiting factor in terms of memory and CPU (e.g. in portable devicies) Delay –frame-size, look-ahead: algorithmic delay –processing delay: depends on complexity and H/W Resilience to loss Quality Others –layered coding –decoding telephone digits?

11. 11 RTP/RTCP Use of Real Time Protocol –adds sequence number, timestamp, payload type Use of Real Time Control Protocol –transmits control packets to participants –offers feedback on performance –correlate and synchronize different media streams

12. 12 RTP Header VersionPaddingExtension bit A/V encoding methodSequence Number Timestamp Synchronization Source Identifier (SSRC) Payload

13. 13 Session Management H.323 SIP (Session Initiation Protocol) –e2e, client-server session signalling protocol –not a transport/QoS reservation/gateway control protocol

14. 14 H.323 Suite of ITU-T recomm. protocols for multimedia collaboration among 2+ entities H.225.0-Q.931 –pt2pt call signalling –initiation/proceeding/alerting/connection/terminat ion H.245 –call-control –master-slave determination/terminal capability set/logical channel management/

15. 15 H.323 Call Flow Setup Alerting/Connect Capabilities Exchange / MSD Open Logical Channel Open Logical Channel Acknowledge RTP Stream RTCP Stream H.225 (TCP port 1720) H.245 (TCP dynamic port) UDP End System A End System B

16. 16 Key Components of SIP SIP end devices –UA client (originates calls) –UA server (listens for incoming calls) SIP workhorses –proxy server relays call signaling (i.e. acts as both client/server) –redirect server redirects callers to other servers –registar accept registration requests from users maintains user’s whereabouts at a Location Server

17. 17 Basic SIP Capabilities user location –correct device with which to comm to reach a particular user user availability –willing/able to take part in a session user capabilities –determines choice of media, codecs session setup –establishes session parameters, s.a. port nubmers session management –call forwarding, modifying session parameters

18. 18 SIP RFC 2543 Methods INVITE ACK BYTE CANCEL OPTIONS REGISTER

19. 19 Session/Call Control - SIP cisco.com proxy princeton.edu proxy mypc.cisco.comyourpc.princeton.edu Sue@cisco.comMeeyoung@princeton.edu

20. 20 Performance Issues Delay –transmission, router-processing, propagation, and queueing delays –packetization Loss –network drop –late arrival FEC (Forward Error Correction) –Adds redundant data –Issues?

21. 21 Playout Delay Adjustment 쏼 ~~ 라 쏴 ~ 라 ~~ 쏴 ~~ 라 쏴 ~ 라 ~~ Playout Delay

22. 22 Voice Quality Rating Measures MOS (Mean Opinion Score) –range of 1.0 to 4.5 (very poor to best) –mean of response scores of human beings PSQM –Perceptual Speech Quality Measurement –Automated human listener –Designed to measure the perceived quality of codec-quality voice based on –Evaluates original signal against “perceptual model” and compares coded signal againt it based on distortion, effects of noise, and overall perceptual fidelity E-model –ITU-T Recommendation G.107 –range of 0 to 100 –R = R 0 -I s - I d - I e +A –effects of noise, simultaneous impairments at quantization, mouth- to-delay impairment, signal distortion, advantage factor (willingness to tolerate deterioration)

23. 23 Readily Available Tools Audio on Linux –/dev/audio for μ-law –/dev/dsp for general samples Java sound I/F –java.sun.com/proudcts/java-media/sound/ –higher layer Java Media Framework (JMF) includes RTP

24. 24 Other Public VoIP Tools rat (Robust Audio Tool) –http://www- mice.cs.ucl.ac.uk/multimedia/software/rat/http://www- mice.cs.ucl.ac.uk/multimedia/software/rat/ Public VoIP S/W repository –http://www.vovida.org/ Many VoIP freeware for Windows

25. 25 References Voice over IP Fundamentals, J. Davidson, J. Peters, 2000, Cisco Press ITU-T Recomm. P.861 [PSQM] IP Telephony with H.323, V. Kumar, M. Korpi, S. Sengodan, 2000, Wiley [H.323] C. Boutremans, J-Y Le Boudec, “Adaptive Joint Playout Buffer and FEC Adjustment for Internet Telephony”, INFOCOM 2003