CCNA Voice Official exam Certification

1. CHAPTER 7 • Gateway and Trunk Concepts CCNA Voice Official exam Certification

2. Converting Analog Voice to Digital: • The average human can hear frequencies of 20-20,000 Hz • Human speech uses frequencies from 200-9000 Hz • Telephone channels typically transmit frequencies of 300-3400 Hz • The Nyquist theorem is able to reproduce frequencies of 300-4000 Hz

3. Converting Analog Voice to Digital continued: • Sample at twice the highest frequency to reproduce accurately (Nyquist) • Quantization is the term used to describe the process of converting an analog signal into a numeric quantity • Since an eight (8) bit binary number can represent a value from zero (0) through two-hundred fifty-five (255) we use the Most Significant Digit (MSD) to represent positive/negative value • A zero (0) in the MSD represents a positive (+) value • A one (1) in the MSD represents a negative (-) value • The result is a range of zero through positive one-hundred twenty-seven (0 through +127) and negative one through negative one-hundred twenty-seven (-1 through -127) • Answer: -76

4. Converting Analog Voice to Digital continued: • Codec’s convert Analog voice into Digital transmissions. • Different Codec’s convert in different methods with more or less complexity • Available Codec’s: • G.711 • Internet low Bitrate Codec (iLBC) • G.729 • G.726 • G.729a • G.728 • Is the Codec supported in the system • How many Digital Signal Processors (DSP’s) are used

5. Converting Analog Voice to Digital continued: • Does the Codec meet satisfactory quality levels • How much bandwidth does the Codec consume • How does the Codec handle packet loss • Does the Codec support multiple sample size

6. Codec’s: • Codec Bandwidth MOS • Consumed • G.711 64 Kbps 4.1 • Internet Low 15.2 Kbps 4.1 • Bitrate Codec (ilBC) • G.729 8 Kbps 3.92 • G.726 32 Kbps 3.85 • G.729a 8 Kbps 3.7 • G.728 16 Kbps 3.61 • MOS (Mean Opinion Score) is determined by listeners listening to the phrase “Nowadays, a chicken leg is a rare dish.” and scoring the quality of the connection on a one to five scale.

7. Calculating Total Bandwidth Needed per Call: • Determine sample size: A larger sample is more efficient (Example: 30 bytes of voice to 50 bytes of overhead 30/80x100%=37.5% is Voice)(Example: 20 bytes of voice to 50 bytes of overhead 20/70x100%=28.5% is voice) • A larger sample takes longer to prepare, so in circuits with delay the voice call will not be as good. • Bandwidth can be saved using Voice Activity Detection (VAD) where no packets are sent during a time when there is no voice • VAD can account for 35-40% of total call time • RTP header compression does not repeat the header after the first packet since the information will stay the same for the length of the call saving 40%

8. Calculating Total Bandwidth Needed per Call continued: • Determine CODEC used • Determine sample size • Determine layer overhead • Layer 2 datalink • Ethernet: 20 bytes • Frame-Relay: 4-6 bytes • Point-to-point Protocol (PPP): 6 bytes • Layer 3 and 4, network and transport • IP: 20 bytes • UDP: 8 bytes • Real-time Transport Protocol (RTP): 12 bytes • Typically layers 3 and 4 are always 40 bytes

9. Calculating Total Bandwidth Needed per Call continued: • Bytes-per-packet = (Sample_size * Codec_bandwidth) / 8 • Total_bandwidth = Packet_size * Packets_per_second • Add any additional overhead: • GRE/L2TP: 24 bytes • MPLS: 4 bytes • Ipsec: 50-57 bytes • Call A: Call B: • 30 mSec Sample size 20 mSec Sample size • G.711 Codec G.729 Codec • Ethernet network Frame-relay network (4 byte)

10. Calculating Total Bandwidth Needed per Call continued: • Call A: • (.03 * 64Kbps) = 1.92Kbps / 8 = 240 bytes • 240 + 20 (ethernet) + 40 (layer 3 and 4) = 300 bytes • 300 * (1 / .03) = 10K bytes per second • 10K * 8 = 80Kbps • Call B: • (.02 * 8Kbps) = 160bps / 8 = 20 bytes • 20 + 4 (frame-relay) + 40 (layer 3 and 4) = 64 bytes • 64 * (1 / .02) = 3.2K bytes per second • 3.2K * 8 = 25.6Kbps

11. Calculating Total Bandwidth Needed per Call Compared continued: • Call B: G.729 • (.02 * 8Kbps) = 160bps / 8 = 20 bytes • 20 + 4 (frame-relay) + 40 (layer 3 and 4) = 64 bytes • 64 * (1 / .02) = 3.2K bytes per second • 3.2K * 8 = 25.6Kbps • Call B: G.711 • (.02 * 64Kbps) = 128Kbps / 8 = 160 bytes • 160 + 4 (frame-relay) + 40 (layer 3 and 4) = 204 bytes • 204 * (1 / .02) = 10.2K bytes per second • 10.2K * 8 = 81.6Kbps • Savings of 68.6% using the G.729 Codec!

12. Digital Signal processors: • DSP’s perform the function of sampling, encoding, and compression of all audio signals coming into the router. • DSP’s might be located on the routers motherboard • DSP’s might also be add on modules similar to SIMM memory modules on the motherboard called Packet Voice DSP Modules (PVDM) • DSP modules can contain multiple DSP circuits • PVDM2-8: Provides .5 DSP chip • PVDM2-16: Provides 1 DSP chip • PVDM2-32: Provides 2 DSP chips • PVDM2-48: Provides 3 DSP chips • PVDM2-64: Provides 4 DSP chips • Codec’s G.711 (a-law and u-law) (u-law is United States, Japan) (a-law All others), G.726, G.729a, and G.729ab are all of medium complexity • Codec’s G.728, G.723, G.729, G.729b and iLBC are all high complexity

13. Digital Signal processors: • To calculate the number of DSP’s needed use the Cisco DSP calculator http://www.cisco.com/cgi-bin/Support/DSP/dsp-calc.pl (Must have Cisco CCO account)

18. RTP and RTCP: • Real-time Transport Protocol (RTP) operates at the transport layer (layer 4) of the OSI model • Real-time Transport Control Protocol (RTCP) also operates at the transport layer (layer 4) of the OSI model • They both work on top of User datagram Protocol (UDP) • Two transport layer protocols simultaneously working is highly unusual but is what happens with voice and video! • UDP works as normal to provide port numbers and header checksums • RTP adds time stamps, sequence numbers, and header information

19. RTP and RTCP continued: • The payload will specify if the packet is handling voice or video • Once established RTP will use even numbered port from between 16,384 and 32,767 • RTP streams are one-way! If a two-way communication takes place then a second session is established • RTCP also engages at the same time and establishes a session using an odd numbered port from the same range that follows the RTC even numbered port chosen • RTCP will account for: • Packet Count • Packet Delay • Packet Loss • Jitter (delay variations) • RTP carries the voice while RTCP does the accounting • RTCP is used to evaluate if there is enough bandwidth or services to complete a call of good quality

20. Internet Low Bitrate Codec (iLBC): • Industry nonproprietary compression codec that is universally supported • Developed in 2000 to provide high-quality, bandwidth-savvy, available to all industry vendors • Provides a bit rate of 15.2 Kbps when coded using a 20 mSec sample size, and 13.3 Kbps when using a 30 mSec sample size • Is a high complexity codec (more DSP required) • High quality approaching G.711 (64 Kbps). The best of any compression codec • Limited support at this time. Cisco phone models that support iLBC: 7906G, 7911G, 7921G, 7942G, 7945G, 7962G, 7965G, and 7975G

21. Trunking the PSTN to CME: • Foreign Exchange Station (FXS) ports typically connect analog phones, fax machines, and modems to the CME router • Foreign Exchange Office (FXO) ports normally connect the PSTN to the CME router, or PBX system • Earth and Magneto (E&M) or Ear and Mouth connects from the PSTN directly to a PBX system

22. Digital Connections: • Channel Associated Signaling (CAS) uses robbed bits from the voice data flow for signaling and control functions. Does affect the voice quality slightly (in-band-signaling) • Common Channel Signaling (CCS) uses a separate channel for all signaling and control functions (out-of-band signaling)

23. Trunking Connections Between Systems: • Common language must be used or conversion between languages • Available languages are H.323, Session Initiation protocol (SIP), Media Gateway Control protocol (MGCP), and Skinny Client Control Protocol (SCCP) • SCCP is Cisco proprietary

24. H.323: • International Telecommunications Union (ITU) accepted in 1996. • Designed to carry multimedia over Integrated Services Digital Network (ISDN) • Based or modeled on the Q.931 protocol • Cryptic messages based in binary • Designed as a peer-to-peer protocol so each station functions independently • More configuration tasks • Each gateway needs a full knowledge of the system • Can configure a single H.323 Gatekeeper that has all system information • Each end system can contact the gatekeeper before making a connection • Gatekeeper can perform Call Admission Control (CAC) to determine if resources are available before a call is accepted • Gatekeeper and Gateway can be the same device

27. SIP: • SIP was designed by the IETF as an alternative to H.323 • SIP is a single protocol whereas H.323 is a suite of protocols as FTP is a single protocol within the TCP/IP protocol suite • SIP is designed to set up connections between multimedia endpoints • Uses other protocols (UDP, RTP, TCP….) to transfer voice or video data • Messaging is in clear ASCII text • Vendors can create their own “add-ons” to the SIP protocol • SIP is still evolving • SIP is destined to become the only voice and video protocol

29. MGCP: • IETF standard with developmental aid from Cisco • All devices under a central control • Voice gateway becomes a dumb terminal • Allows minimal local configuration • Single point of failure • Uses UDP port 2427

31. SCCP: • Often called “skinny” protocol • Cisco proprietary • Similar to MGCP in that it is a stimulus/response protocol • Allows Cisco to deploy new features in their phones • Cisco phones must work with Cisco systems (CME, CUCM,CUCME…) • Cisco phones can also use other protocols such as SIP or MGCP with downloaded firmware

32. Internet Telephone Service Providers: • New service providers that provide phone services over the internet (Vonage) • They interface with the traditional phone service providers (PSTN) • Bundle voice and data together

33. End of Chapter 7