Which statement is true about a VoIP network? The circuit switching technology creates a single, integrated network.

Only 64 kb/s channels are needed for each call. Multiple, single channels are combined to create the voice and data circuits. VoIP estadsticamente multiplexa el trfico de voz junto con el trfico de datos. 2

Refer to the exhibit. On the basis of the two scenarios, which two statements are true about the impact of the VoIP packet size and rate on network performance? (Choose two.) The first scenario introduces smaller IP overhead and lower bandwidth consumption but will incre packets. The first scenario introduces additional IP overhead, which results in higher bandwidth con transport the voice packets. El primer escenario introduce adicional IP sobre la cabezera, que se traduce en el consumo de mayor ancho de banda necesario para transportar los paquetes de voz. The first scenario introduces the VoIP packets transportation at lower packet rate, which results in consumption to transport the voice packets. The second scenario introduces smaller IP overhead and lower bandwidth consumption but will increase the delay for the voice packets. El segundo escenario presenta menores IP sobre la cabecera y un menor consumo de ancho de banda, pero aumentar el retraso de los paquetes de voz.

The second scenario introduces additional IP overhead, which results in higher bandwidth consum the voice packets. The second scenario introduces the VoIP packets transportation at higher packet rate, which result consumption to transport the voice packets. 3

Refer to the exhibit. Which command is used only with a dial peer that is on the POTS network? destination-patternxxxx session target ipv4:ip-address portport-number * dial-peer voicexvoip 4

Refer to the exhibit. Which two statements are true about the distributed call processing deployment? two.) Calls are primarily routed over the PSTN in digital form. The PSTN can be used as a backup path for all intersite calls if the WAN link is down.

All IP phones are served by the Cisco Unified CallManager cluster at the headquarters site. The IP WAN will carry call signaling, voice packets, and data packets for all intersite communication. La WAN IP llevar a la sealizacin de llamadas, los paquetes de voz y paquetes de datos pa las comunicaciones entre sitios. Full CallManager functionality is available only at the headquarters site. 5 Which statement is true about gateways? They provide native support for half-duplex transmission of conversations only. Before the telephone signal reaches the gateway, it must already be in digital format. They can facilitate full-duplex transmission of conversations. Se puede facilitar la transmisin full-duplex de conversaciones. They can provide control to prevent a network from being oversubscribed. Which statement is true about the overhead in a VoIP network? Data link overhead will remain constant throughout the network. Codecs have the same bandwidth requirements regardless of the codec type. The packetization overhead relies solely on the packet rate. IPsec and tunneling protocols add headers of various sizes. What are two benefits that are derived from implementing a VoIP network? (Choose two.) a reduction in overall bandwidth requirements the tax deductions for transmission costs la consolidacin de los gastos de la red una mejora en la productividad de los empleados a travs de las caractersticas que se proporcionan por la telefona IP the facilitation in education delivery

la consolidacin de los gastos de la red una mejora en la productividad de los empleados a travs de las caractersticas que se proporciona telefona IP Which two factors must be considered when determining the total bandwidth that is required for a VoIP call? (Choose two.) serialization rate tasa de paquetizacin tamao total del paquete QoS delay data-link checksum

What are three functions of a voice gateway? (Choose three.) provides CAC for VoIP packet streams provides an ingress point for an analog or digital voice stream and routes it to the proper d on the network provides a half-duplex communication stream to the phone performs data compression performs analog-to-digital and digital-to-analog conversion for analog phones performs the proper TCP header compression that is based upon the available bandwidth of the n 1 Which three factors will have an impact on the percentage of bandwidth that is saved by voice activity detection (VAD)? (Choose three.) the codec in use full-duplex versus half-duplex conversation sampling rate analog or digital phone signal msica en espera (MoH) ruido de fondo What are two functions of a Cisco CallManager? (Choose two.) to provide dial tone to analog and digital phones para determinar las rutas de enrutamiento de llamadas to provide the hardware interface to the PSTN to provide dial plans for H.323 and SIP compatible devices para almacenar y descargar configuraciones de telfono

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1 A Cisco router can act as a voice gateway. In this role, which two functions are added as a result? (C 2 two.) centralized call management conversin de seales analgicas a formato digital xml applications to users advanced integrated module (AIM) resources for conferencing and transcoding encapsulacin de voz en paquetes IP 13

Refer to the exhibit. Which sequence correctly matches the column titles and associates voice encapsulations to their features? X - UDP, Y - TCP, Z - UDP X - TCP, Y - RTP, Z - UDP X - TCP, Y - UDP, Z - RTP X - UDP, Y - RTP, Z - TCP X - RTP, Y - TCP, Z - UDP X - RTP, Y - UDP, Z - TCP 4 1 Which two statements are true about the utilization of codecs in a VoIP network? (Choose two.) G.711 cuts bandwidth requirements by 50 percent. G.711 produce 64 kb / s de flujo de datos. G.711 cuts the sample size to 4 bits per sample. G.729 would be more useful on a slow WAN link than G.711 would be. G.729 sera ms til en un enlace WAN lenta que sera G.711. G.729 and G.711 should be used together on the same network to improve the voice quality. G.729 cuts the sample size to 4, 3, or 2 bits per sample.

15 In a VoIP network, what do call agents and gatekeepers have in common? Both systems provide routing and central management of all endpoints. Both systems can provide Call Admission Control (CAC). Both systems are options within a VoIP network because their services can be performed by other Both systems provide tracking services for voicemail boxes, conference bridges, and transcoding r 16

Refer to the exhibit. The exhibit depicts the signaling and call processing that is associated with a call from the branch office to headquarters. Which statement describes what occurs in step 2 in the exhibit? The IP phones start sending and receiving RTP packets. The Cisco Unified CallManager sends a signaling message to the destination IP phone. El Cisco Unified CallManager busca el nmero llamado en En la tabla de enrutamiento de llamadas. The IP phone sends signaling messages to a member of the Cisco Unified CallManager cluster. Which three steps are taken to convert a signal from an analog phone 17 into a digital data stream for transmission over an IP network? (Choose three.) Sampling (muestreo) Decoding quantization compression Companding Filtering 18

Which statement is true about gatekeepers? They perform full-duplex transmission of conversations. They provide address translation between private and public IP addresses. They interconnect the VoIP network with traditional telephony devices. They provide routing and central management of all endpoints (terminals, gateways, and M a given zone. Ofrecen enrutamiento y gestin central de todos los puntos finales (terminales, gateways y M

una zona determinada. 1 9

Refer to the exhibit. Router R1 is a voice-enabled router. What are three functions that are performed by the DSP in R1 with the voice flow from T1 to T2? The DSP samples the analog signal at periodic intervals to provide an output of the sampling as a pulse amplitude modulation (PAM) signal. Las muestras de DSP de la seal analgica a intervalos peridicos para proporcionar una salida de la toma de muestras como la modulacin de amplitud de pulso (PAM) de la seal. The DSP decodes the digital voice samples to the amplitude value of the samples and rebuilds th amplitude modulation (PAM) signal. The DSP matches the pulse amplitude modulation (PAM) signal to a segmented scale to provide quantization. El DSP coincide con la modulacin de amplitud de pulso (PAM) de la seal a una escala de segmentos para proporcionar cuantizacin. The DSP passes the pulse amplitude modulation (PAM) signal through a properly designed filter reconstructs the original analog signal. The DSP compresses voice samples to reduce bandwidth requirements on the network. El DSP comprime muestras de voz para reducir los requisitos de ancho de banda en la red. The DSP compresses the voice samples to convert the analog signals into digital. 20 Which three functions are performed by the Cisco Unified CallManager? (Choose three.) the interconnection of traditional telephony systems dial plan administration the conversion of analog signals into digital format signaling and device control call processing the support of dual tone multifrequency (DTMF) relay