## ONT – VoIP Implementation Pt.2

 The total size of a L2 frame encapsulating a VoIP packet depends on what factors?
 – Packet rate and packetization size   – IP overhead We Will Write a Custom Essay SpecificallyFor You For Only \$13.90/page!order now   – Data link overhead   – Tunneling overhead
 What is the data link overhead for each of the following data link layer protocols?   – Ethernet   – Frame Relay   – Multilink PPP (MLP)   – Dot1Q
 – Ethernet = 18 bytes   – Multilink PPP (MLP) = 6 bytes   – Frame Relay = 6 bytes    – Dot1Q = 22 bytes
 How much overhead do each of the following tunnel protocols add to the voice payload?   – IPsec Transport mode   – IPsec Tunnel mode    – L2TP   – GRE   – MPLS   – PPPoE
 – IPsec Transport mode = 30 to 53 bytes   – IPsec Tunnel mode = 50 to 73 bytes   – L2TP = 24 bytes    – GRE = 24 bytes   – MPLS = 4 bytes   – PPPoE = 8 bytes
 What are the six major steps in computing the bandwidth consumed by a VoIP call?
 1. Determine the codec and the packetization period.   2. Determine the link-specific information, including encapsulation type and if cRTP is used. Also see if any security or tunneling protocols are used.   3. Calculate the packetization size.   4. Calculate the total frame size.   5. Calculate the packet rate.   6. Calculate the total bandwidth.
 How do you determine the packetization period to be used by a VoIP call?
 The number of digital voice samples (each of which is equivalent to 10 ms of analog voice) encapsulated in each IP packet determines the packetization period. A packetization period of 20 ms, which is the default in Cisco voice-enabled devices, means that each VoIP packet will encapsulate two 10-ms digital voice samples.
 How do you calculate the packetization size (the size of voice payload) for a specific VoIP configuration?
 1. Convert the packetization period from milliseconds to seconds.   2. Multiply the codec bandwidth (converted to bits) by the packetization period (converted to seconds)   3.  Divide the result in step 2 by 8 (bits per byte)   So with G.729, codec = 8 Kbps, packetization period is default (20 ms). So (8000 * 0.020) / 8 = 20 bytes
 How do you calculate the total frame size used in a VoIP call?
 How do you calculate the packet rate used in a VoIP call?
 By taking the packetization period, and converting it from millisenconds to seconds. Then you take 1 divided by the result. For example, if the packetization period is 20 ms, you convert that to 0.020 seconds. Finally, take 1 and divide it by 0.020, and you get 50 packets per second (pps)
 How do you calculate the total bandwidth consumed by one VoIP call?
 1. Take the total frame size, and convert it from bytes to bits.   2. Multiply step one by the packet rate.   3. Convert the final result from bits to kilobits by dividing the result by 1000.   For example, 66 bytes is equal to 528 bits. Multiplied by 50 pps is 26400 bits, which converts to 26.4 Kbps.
 Describe the purpose of gateways in a packet telephony network.
 Gateways convert analog signals to digital and digital signals to analog. They might also be able to handle several types of codec, which also provide transcoding. They can also provide survivable remote site telephony (SRST) when links to the main office go down, or can by installed with CallManager Express, and function as its own CA.
 Describe the basic function of Cisco’s CallManager in a VoIP network.
 Cisco CallManager (CCM) is call processing software, and is the main component of the Cisco Unified Communication System. CCM provides the following services and functions: ; – Call processing; – call routing, signaling and accounting ; – Dial plan administration – acts as the CA for MGCP gateways and IP phones ; – Signaling and device control – performs signaling for devices and fully controls their configuration and behavior ; -; Phone feature administration -centralized storage and control of IP phone configuration files ; – Directory and XML services – centralized administration of directory services ; – Programming interface to external applications – provides an API so that applications can be written to work and communicate with CCM;
 What are the 4 different IP telelphony deployment models available?
 – Single site ; – Multisite with centralized call processing ; – Multisite with distributed call processing ; – Clustering over WAN;
 Describe the use of Call Admission Control (CAC) is a packet telephony network.
 Call admission control (CAC) is a feature that is configured to limit the number of concurrent calls, preventing VoIP traffic from ever exceeding their QoS reservations.
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