Voice over IP Protocols And Compression Algorithms

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1 University of Tehran Electrical and Computer Engineering School SI Lab. Weekly Presentations Voice over IP Protocols And Compression Algorithms Presented by: Neda Kazemian Amiri

2 Agenda Introduction to VoIP Protocols Compression Algorithms and Standards G. Standards Future work Algorithms 2

3 Convergence of Voice Data existed on packet-switched digital networks and Data Like two parallel universes, data and voice were in in the same space on on different planes Voice existed on proprietary, circuit-switched telephone networks [1] Algorithms 3

4 Data/Voice Convergence Why? Cost initially...but emerging business applications will be the real driver Integrated /voice mail messaging Web-based call centers Computer Telephony Integration (CTI) Intranet and Internet telephony Fax Desktop video [1] Algorithms 4

5 Data, Voice, and Video Integration Benefits One Infrastructure Provides data, voice and video Voice rides for free PBX elimination Simplified operations Budget leverage New applications to drive competitive advantage Algorithms 5

6 Personal Telephony Services Voice Mail Page Fax Personal Rules Engine Administration/ filtering Visual messaging Calendar setting Integrated search and calls IP Network [1] Cell/PDA Audio and Palm Pilot Videoconferencing Algorithms 6

7 Traditional TDM Networking Wasted Bandwidth A SNA TDM B TDM C [1] TDM uses dedicated bandwidth to establish an end-to-end connection Traffic entering a TDM device must be less than or equal to the amount of traffic leaving the TDM device Algorithms 7

8 Integrated Multiservice Networks: Data/Voice/Video SNA Data Voice PBX Multiservice Network Video Internet Packet-Based [1] Algorithms 8

9 Packet-Based Networking Voice Data Q Video Packets [1] Statistical multiplexing makes efficient use of bandwidth Input traffic into the statistical device may be greater than the output traffic Algorithms 9

10 Voice over Packet Networks Allow Real-Time Voice on Data Networks 1. Speech is converted to digital voice packets Network 2. Packets are sent over a data network 3. Digital voice packets are converted to analog speech [1] Algorithms 10

11 Converting from Voice to Data DSP CODEC 1. Person speaks into telephone 2. CODEC (coder-decoder) converts the signal from analog to digital data packets suitable for transmission over a TCP/IP data network 3. Digital signal processing (DSP) chip compresses the packets for transmission over the data network Algorithms 11

12 Converting from Data Back to Voice DSP CODEC 4. DSP chip uncompresses the packets 5. CODEC software converts the signal from digital data packets back to analog voice 6. Recipient listens to the voice on their telephone Algorithms 12

13 Choosing VoIP Converged Networks Voice, Video & Data over an IP network Reduced the costs of managing parallel networks Allows voice to be an IP application Handles phone-to-computer voice communications unified messaging Web-enabled call centers Centralized or distributed architectures Add features where they are needed Algorithms 13

14 Basic Components of VoIP Voice Processing Codec (Coder/Decoder) Echo Cancellation unit Voice Activity/Idle noise detector Call Processing PSTN interface Signaling protocol management Call establishment/teardown Telephone number mapping Packet Processing Convert data stream from codec to packet format Adds appropriate transport headers Convert signaling protocols: Telephony to packet signaling Network Management Performance monitoring Fault detection Security management Algorithms 14

15 Silence Suppression by Voice Activity Detection Voice Activity Detection (VAD) Speech Silence Speech No Cells Cells Cells [1] Algorithms 15

16 The Protocols!! Algorithms 16

17 Definition of Protocol A protocol is a set of formal rules which describe how to transmit data. Protocols may deal with the way data is formatted, including: syntax of messages terminal to computer dialogue sequencing of messages Algorithms 17

18 VoIP Unions Providing Standard ITU-T: International Telecommunications Union International standards body for telephony ITU-T H.323 International Telecommunications Union recommendation for multimedia (including voice) networking over IP IMTC: International Multimedia Teleconferencing Consortium International standards body providing recommendations for multimedia networking over IP, including VoIP IETF: Internet Engineering Task Force (IETF) Internet standards body Algorithms 18

19 H.323 standard Voice over IP Components Specifies call setup and interoperability G. standards Specify analog-to-digital conversion and compression Realtime Transport Protocol (RTP) Manages end-to-end connections to minimize the effect of packets lost or delayed in transit Internet Protocol (IP) Specifies routing of packets on the network Algorithms 19

20 Protocols H.323 an ITU standard describes how multimedia communications occur between terminals, network equipment, and assorted services on IP networks. SIP (Session Initiation Protocol) a competing IETF standard performs similar functions to H.323 May see merger of these two into one Algorithms 20

21 H.323 A Closer Look Terminals (what people see/hear) Gateways (control and routing ) Multipoint Control Units (provides conference capabilities ) E N D P O I N T S Gatekeepers (access to other environments) Algorithms 21

22 Terminals H.323 client endpoints They could be: Multimedia PCs Any stand-alone device A simple telephone Expectation by H.323: Must support audio communication. Video, data support optional Algorithms 22

23 Gateways Optional Component of H.323 implementation. Used as interface between different networks e.g. LAN & PSTN Functions: Data format translation Audio/video codec translation (DSP s) jitter buffers, echo cancellation, and packet processing Call setup, termination from both sides of the network Algorithms 23

24 Multipoint Control Units MCUs are also optional in a H.323 implementation Needed only when multiparty conferences are desired Functions: Provides capability of video-conferencing with more than one party. Acts as a coordinator of multiparty conferences Algorithms 24

25 Gatekeepers Brains of a H.323 network controls access to the gateway Expectation by H.323: Address translation Admissions Control Bandwidth Control Zone Management Routing Capabilities network signaling Algorithms 25

26 The Protocols H.323 recommendation is a framework document that describes how the various pieces fit together H defines the call signaling between endpoints and the Gatekeeper RTP/RTCP is used to transmit media such as audio and video over IP networks H define the procedures and protocol for communication within and between Peer Elements H.245 is the protocol used to control establishment and closure of media channels within the context of a call and to perform conference control H.450.x is a series of supplementary service protocols H.460.x is a series of version-independent extensions to the base H.323 protocol T.120 specifies how to do data conferencing T.38 defines how to relay fax signals V defines how to relay modem signals H.235 defines security within H.323 systems Algorithms 26

27 Characteristics of H.323 Centralized and Distributed Control Endpoints can communicate directly or through a server Focused provides users with voice, data, and video conferencing over a packet-switched network Extensibility Fully backward compatible with previous versions. Equipment manufactures can extend functionality by inserting their own additions to the protocol. Integration with Internet Standards Algorithms 27

28 Voice Quality Guidelines Score Quality Excellent Good Fair Poor Bad Description of Impairment Imperceptible Just Perceptible, Not Annoying Perceptible and Slightly Annoying Annoying but Not Objectionable Very Annoying and Objectionable Mean Opinion Score Subjective Quality (MOS) Hybrid Coders (LD-CELP and CS-ACELP) Vocoders (Older Technology) Waveform Coders (ADPCM) kbps [6] Algorithms 28

29 Algorithms 29 Linear Predictive Coding and Speech Synthesis ) ( ) ( ) ( ) ( z A G z X z S z H = = = + = p j j j n s a n Gx n s 1 ) ( ) ( ) ( = = p j j a j z z A 1 1 ) ( [6]

30 Code Excited Linear Prediction (CELP) a) Coder and b) Decoder [3] [6] Algorithms 30

31 The Audio and Video Audio: PCM-Based: G.711, G.721, G.722, G.726 CELP-Based: G.723.1, G.728, G.729 Video Protocols of H.323 H.261 codec (for channels with bandwidths p*64 kb/s) H.263 codec (for low bit rate transmission without loss of quality) Algorithms 31

32 ITU G. Standards for Audio G.711 The first standard for speech compression. PCM, Frame size = 0.125ms 64Kbps used in PSTN MOS=4.1 G.721 ADPCM, Frame size = 0.125ms 32Kbps used in PSTN. Year of introduction: 84 G.722 It is like G.721. maximum bit rate 64Kbps. G.726 ADPCM, Frame size = 0.125ms 16, 24, 32 and 40 Kbps MOS=3.85 Year of introduction: 90 G hybrid coder, Frame size = 30ms with MP-MLQ algorithm its bit rate is 6.3 Kbps. (MOS=3.9) with ACELP algorithm its bit rate is 5.3 Kbps. (MOS=3.65) used for videophones. G.728 hybrid coder LD-CELP algorithm, Frame size =.625ms 16 Kbps. uses 5 samples frames. MOS=3.61 Year of introduction: 92 G.729 hybrid coder CS-ACELP algorithm, Frame size = 10ms 8 Kbps uses 10ms frames. MOS=3.92 Year of introduction:95 Algorithms 32

33 Comparison of Some Standards in the Terms of Quality and Bit Rate Algorithms 33

34 LD-CELP Encoder Used in G.728 [5] Algorithms 34

35 LD-CELP Decoder Used in G.728 [5] Algorithms 35

36 Future Work Studying & comparing the VoIP Protocols in more detail H.323 SIP MGCP (Media Gateway Control Protocol) Skinny ( a cisco standard) Studying CELP-based compression algorithms and codec recommendations G.729 G.728 G Algorithms 36

37 References [1] Cisco systems, [2] Swale, R., Voice over IP: systems and solutions, The Institution of Electrical Engineers, 2001 [3] ITU-T Recommendation G.729, Coding of Speech at 8Kbps Using Conjugate-Structure Algebraic Code Excited Linear Prediction (CS-ACELP), 1995 [4] ITU-T Recommendation G.723.1, Dual Rate Speech Coder for Multimedia Communications Transmitting at 5.3 and 6.3Kbps, 1996 [5 ITU-T Recommendation G.728, coding of Speech at 16Kbps Using Low-Delay Code Excited Linear Prediction, 1992 [6] Kondoz, A. M., Digital Speech, Coding for Low Bit Rate Communications Systems, John Wiley & Sons, Algorithms 37

38 References [7] Xydeas, C., An Overview of Speech Coding Techniques, Speech coding Techniques and applications, IEE colloquium on 14 Apr. 1992, pp [8] Kipper, U., Reininger, H., and Wolf, D., CELP Coding with Adaptive Excitation Codebooks, IEEE., [9] Schroeder, M. R., and Atal, B. S., Code-Excited Linear Prediction (CELP): High-Quality Speech at Very Low Bit Rates, IEEE, [10] Owen, F.E., PCM and Digital Transmission Systems, McGRAW-HILL Book Company, [11] Proakis, J. G., Salehi, M., Contemporary Communication Systems Using MATLAB, PWS Publishing Company, 1997 [12] Brunner, S., and Ali, A. A., Voice over IP, Understanding VoIP Networks, Juniper Networks, Inc., [13] M. Banerjee, B. A. Vani, S. Madhusudhan and S. Monga, Optimizations of ITU G.729 Speech Codec, IEEE 60th Vehicular Technology Conference, vol. 6, pp , Sept Algorithms 38

39 Algorithms 39

40 Questions? Algorithms 40

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