Optimizing Converged Cisco Networks (ONT)

1. Optimizing Converged Cisco Networks (ONT) Module 2: Cisco VoIP Implementations

2. Lesson 2.1: Introducing VoIP Networks

3. Objectives • Describe the benefits of a VoIP network. • Describe the components of a VoIP network. • Describe the legacy analog interfaces used in VoIP networks. • Describe the digital interfaces used in VoIP networks. • Explain the 3 phases of call control. • Compare and contrast distributed and centralized call control.

4. Benefits of a VoIP Network • More efficient use of bandwidth and equipment • Lower transmission costs • Consolidated network expenses • Improved employee productivity through features provided by IP telephony: • IP phones are complete business communication devices. • Directory lookups and database applications (XML) • Integration of telephony into any business application • Software-based and wireless phones offer mobility. • Access to new communications devices (such as PDAs and cable set-top boxes)

5. Components of a VoIP Network

6. Legacy Analog and VoIP Applications Can Coexist

7. Legacy Analog Interfaces in VoIP Networks

8. Legacy Analog Interfaces in VoIP Networks 5 1 3 2 4 1

9. Digital Interfaces

10. Call Setup • Checks call-routing configuration • Determines bandwidth availability • If bandwidth is available, setup message is passed • If bandwidth is not available, busy signal is generated

11. Call Maintenance • Tracks quality parameters: • Packet loss • Jitter • Delay • Maintains or drops call based on connection quality

12. Call Teardown • Notifies devices to free resources • Resources are made available to subsequent calls

13. Distributed Call Control

14. Centralized Call Control

15. Self Check • Which type of call control uses a call agent to route the call? • What is a DSP? • Name 3 types of analog interfaces used at gateways. • What are the 3 components of basic call control? • What phase of call control involves determining if bandwidth is available to place the call?

16. Summary • The benefits of a VoIP network include more efficient use of network bandwidth and equipment, lower cost and consolidated expenses. • Legacy analog and VoIP applications and devices can coexist. • The 3 stages of a VoIP call include call setup, call maintenance, and call teardown. • VoIP can be deployed in a centralized or distributed environment.

17. Lesson 2.2: Digitizing and Packetizing Voice

18. Objectives • Describe the process of analog to digital conversion. • Describe the process of digital to analog conversion. • Explain how sampling rates are determined using the Nyquist Theorem. • Explain how quantization can lead to noise. • Explain how MOS is used to judge voice quality. • Describe the purpose of DSPs.

19. Basic Voice Encoding: Converting Analog Signals to Digital Signals • Step 1: Sample the analog signal. • Step 2: Quantize sample into a binary expression. • Step 3: Compress the samples to reduce bandwidth.

20. Basic Voice Encoding:Converting Digital Signals to Analog Signals • Step 1: Decompress the samples. • Step 2: Decode the samples into voltage amplitudes, rebuilding the PAM signal. • Step 3: Reconstruct the analog signal from the PAM signals.

21. Determining Sampling Rate with the Nyquist Theorem • The sampling rate affects the quality of the digitized signal. • Applying the Nyquist theorem determines the minimum sampling rate of analog signals. • Nyquist theorem requires that the sampling rate has to be at least twice the maximum frequency.

22. Example: Setting the Correct Voice Sampling Rate • Human speech uses 200–9000 Hz. • Human ear can sense 20–20,000 Hz. • Traditional telephony systems were designed for 300–3400 Hz. • Sampling rate for digitizing voice was set to 8000 samples per second, allowing frequencies up to 4000 Hz.

23. Quantization • Quantization is the representation of amplitudes by a certain value (step). • A scale with 256 steps is used for quantization. • Samples are rounded up or down to the closer step. • Rounding introduces inexactness (quantization noise).

24. Quantization Techniques • Linear quantization: • Lower SNR on small signals (worse voice quality) • Higher SNR on large signals (better voice quality) • Logarithmic quantization provides uniform SNR for all signals: • Provides higher granularity for lower signals • Corresponds to the logarithmic behavior of the human ear

25. Digital VoiceEncoding • Each sample is encoded using eight bits: • One polarity bit • Three segment bits • Four step bits • Required bandwidth for one call is 64 kbps (8000 samples per second, 8 bits each). • Circuit-based telephony networks use TDM to combine multiple 64-kbps channels (DS-0) to a single physical line.

26. Companding • Companding — compressing and expanding • There are two methods of companding: • Mu-law, used in Canada, U.S., and Japan • A-law, used in other countries • Both methods use a quasi-logarithmic scale: • Logarithmic segment sizes • Linear step sizes (within a segment) • Both methods have eight positive and eight negative segments, with 16 steps per segment. • An international connection needs to use A-law; mu-to-A conversion is the responsibility of the mu-law country.

27. Coding • Pulse Code Modulation (PCM) • Digital representation of analog signal • Signal is sampled regularly at uniform levels • Basic PCM samples voice 8000 times per second • Basis for the entire telephone system digital hierarchy • Adaptive Differential Pulse Code Modulation • Replaces PCM • Transmits only the difference between one sample and the next

28. Common Voice Codec Characteristics

29. Mean Opinion Score

30. A Closer Look at a DSP DSP Module A DSP is a specialized processor used for telephony applications: • Voice termination: • Works as a compander converting analog voice to digital format and back again • Provides echo cancellation, VAD, CNG, jitter removal, and other benefits • Conferencing: Mixes incoming streams from multiple parties • Transcoding: Translates between voice streams that use different, incompatible codecs Voice Network Module

31. DSP Used for Conferencing • DSPs can be used in single- or mixed-mode conferences: • Mixed mode supports different codecs. • Single mode demands that the same codec to be used by all participants. • Mixed mode has fewer conferences per DSP.

32. Example: DSP Used for Transcoding

33. Self Check • What sampling frequency is recommended by the Nyquist Theorem for reconstruction of a signal? • What is the Hz range for traditional telephone systems? • What is the implication of using 8 bits for quantization? • What is the purpose of logarithmic quantization? • What is MOS?

34. Summary • Voice-enabled routers convert analog voice signals to digital format for encapsulation in IP packets and transport over IP networks. These packets are converted back to analog at the other end. • Quantization is the process of selecting binary values to represent voltage levels of voice samples. Quantization errors arise when too few samples are taken. • There are two methods of companding: Mu-law, used in Canada, U.S., and Japan, and A-law, used in other countries. • The Mean Opinion Score (MOS) provides a numerical indication of the perceived quality of received media after compression and/or transmission.

35. Q and A

36. Resources • Voice Codec Bandwidth Calculator (requires CCO login) • http://tools.cisco.com/Support/VBC/do/CodecCalc1.do • DSP Calculator • http://www.cisco.com/cgi-bin/Support/DSP/dsp-calc.pl • Free VoIP Quality Tester • http://www.testyourvoip.com/

37. Lesson 2.3: Encapsulating Voice Packets for Transport

38. Objectives • Compare and contrast voice transport in circuit-switched and VoIP networks. • Describe causes of and solutions for jitter. • Explain the issues with IP, TCP, and UDP when transporting voice packets. • Describe encapsulation overhead issues for VoIP. • Describe header compression and when and where it should be used.

39. Voice Transport in Circuit-Switched Networks • Analog phones connect to CO switches. • CO switches convert between analog and digital. • After call is set up, PSTN provides: • End-to-end dedicated circuit for this call (DS-0) • Synchronous transmission with fixed bandwidth and very low, constant delay

40. Voice Transport in VoIP Networks • Analog phones connect to voice gateways. • Voice gateways convert between analog and digital. • After call is set up, IP network provides: • Packet-by-packet delivery through the network • Shared bandwidth, higher and variable delays

41. Jitter • Voice packets enter the network at a constant rate. • Voice packets may arrive at the destination at a different rate or in the wrong order. • Jitter occurs when packets arrive at varying rates. • Since voice is dependent on timing and order, a process must exist so that delays and queuing issues can be fixed at the receiving end. • The receiving router must: • Ensure steady delivery (delay) • Ensure that the packets are in the right order

42. VoIP Protocol Issues • IP does not guarantee reliability, flow control, error detection or error correction. • IP can use the help of transport layer protocols TCP or UDP. • TCP offers reliability, but voice doesn’t need it…do not retransmit lost voice packets. • TCP overhead for reliability consumes bandwidth. • UDP does not offer reliability. But it also doesn’t offer sequencing…voice packets need to be in the right order. • RTP, which is built on UDP, offers all of the functionality required by voice packets.

43. Protocols Used for VoIP

44. Voice Encapsulation • Digitized voice is encapsulated into RTP, UDP, and IP. • By default, 20 ms of voice is packetized into a single IP packet.

45. Voice Encapsulation Overhead • Voice is sent in small packets at high packet rates. • IP, UDP, and RTP header overheads are enormous: • For G.729, the headers are twice the size of the payload. • For G.711, the headers are one-quarter the size of the payload. • Bandwidth is 24 kbps for G.729 and 80 kbps for G.711, ignoring Layer 2 overhead.

46. RTP Header Compression • Compresses the IP, UDP, and RTP headers • Is configured on a link-by-link basis • Reduces the size of the headers substantially (from 40 bytes to 2 or 4 bytes): • 4 bytes if the UDP checksum is preserved • 2 bytes if the UDP checksum is not sent • Saves a considerable amount of bandwidth

47. cRTP Operation

48. When to Use RTP Header Compression • Use cRTP: • Only on slow links (less than 2 Mbps) • If bandwidth needs to be conserved • Consider the disadvantages of cRTP: • Adds to processing overhead • Introduces additional delays • Tune cRTP—set the number of sessions to be compressed (default is 16).

49. Self Check • What causes jitter? • Explain why IP is not well suited to voice transmission. • What issues does TCP have when considering it as the protocol for voice? • What guidelines can be used to determine when and where to use RTP header compression? • What are some disadvantages to using RTP header compression?

50. Summary • Circuit-switched calls use dedicated links. VoIP networks send voice in packets. • Jitter is caused by packets arriving at the destination at varying rates and not in the original order. • IP, TCP or UDP alone cannot be used for voice packets. IP does not guarantee reliability, flow control, error detection or error correction. TCP has unnecessary overhead. UDP needs additional functionality offered by RTP. • Encapsulation overhead for VoIP can be very large. Since voice packets are small, compression should be used to compress headers.