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

2. Converting Analog Voice to Digital:
The average human can hear frequencies of 20-20,000 Hz
Human speech uses frequencies from Hz
Telephone channels typically transmit frequencies of Hz
The Nyquist theorem is able to reproduce frequencies of 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 1

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 Kbps 4.1
Bitrate Codec (ilBC)
G Kbps
G Kbps
G.729a 8 Kbps 3.7
G Kbps
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: bytes
Frame-Relay: bytes
Point-to-point Protocol (PPP): 6 bytes
Layer 3 and 4, network and transport
IP: bytes
UDP: 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: bytes
MPLS: bytes
Ipsec: 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
(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
(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
(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
(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 (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
Data Link IP
RTP
UDP
Audio Payload
Payload Type
Sequence Number
Time Stamp

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. 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

27. 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. 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

31. 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

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