CCNA Voice Official exam Certification CHAPTER 7 Gateway and Trunk Concepts

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

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 1

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

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

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.

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%

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

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)

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

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!

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

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)

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 Data Link IP RTP UDP Audio Payload Payload Type Sequence Number Time Stamp

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

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

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

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)

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

International Telecommunications Union (ITU) accepted in 1996. 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

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

IETF standard with developmental aid from Cisco 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

Often called “skinny” protocol 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

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

End of Chapter 7