Tested of 300-075 test question materials and faq for Cisco certification for customers, Real Success Guaranteed with Updated 300-075 pdf dumps vce Materials. 100% PASS Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) exam Today!2016 Apr 300-075 Study Guide Questions:Q166. Which statement is true when considering a Cisco VoIP environment for regional configuration? A. G.711 requires 128K of bandwidth per call. B. G.729 requires 24K of bandwidth per call. C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified Communications Manager environment. D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only use G.711 between regions. Answer: C Q167. Which two options are requirements for deploying an H.323 gateway with Cisco Unified Communications Manager? (Choose two.) A. Cisco Unified Communications Manager and the H.323 gateway must be configured use the same TCP port for H.323 calls. B. The H.245TCSTimeout timer must be set to at least 25. C. Cisco voicemail ports must be active. D. The Media Exchange Interface Capability Timer must be set to less than 20. E. The Media Exchange Timer must be set to less than 20. Answer: A,B Q168. Video calls using 384 kbps need to be supported across a gatekeeper-controlled trunk. What value should be entered into the gatekeeper to support this bandwidth? A. 768 kbps B. 384 kbps C. 512 kbps D. 192 kbps Answer: B Explanation: Incorrect Answer: A, C, D A 384-kb/s video call may comprise G.711 at 64 kb/s (for audio) plus 320 kb/s (for video). This sum does not include overhead. If the audio codec for a video call is G.729 (at 24 kb/s), the video rate increases to maintain a total bandwidth of 384 kb/s. Link: http://www.ciscosystems.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08video.html#wp1059726 Q169. Which statement about SIP precondition is most correct? A. When configuring SIP precondition, the SIP trunk must have access to an RSVP agent. B. When configuring SIP precondition, the IP phones must have access to an RSVP agent. C. When configuring SIP precondition, the IP phones and SIP trunk must have access to an RSVP agent. D. RSVP agents are only required for the IP phones. SIP trunks require RSVP agents only when fall back to local RSVP is configured. E. SIP trunk will always require RSVP agents regardless of what RSVP type is configured. Answer: D Q170. Which option indicates the best QoS parameters for interactive video? A. 1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning B. 5% Max Loss, 5 s One-way Latency, 30 ms Jitter, 20% Overprovisioning C. 0% Max Loss, 100 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning D. 1% Max Loss, 160 ms One-way Latency, 60 ms Jitter, 10% Overprovisioning Answer: A Up to date 300-075 exam prep:Q171. Which option best describes a service that assembles a network model from configured locations and link data in one or more clusters? A. LBM B. Weight C. LBM Hub D. Shadow Answer: A Q172. In a Cisco Unified Communications Manager centralized call processing model, what is the best CAC method recommended for this type of deployment? A. QoS-based B. location-based C. RSVP-based D. region-based E. gateway-based F. gatekeeper-based Answer: B Q173. This is the configuration on the voice gateway: telephony-service max-ephones 30 max-dn 60 preference 0 srst mode auto-provision all srst dn line-mode dual srst dn template 3 srst ephone description srst fallback auto-provision phone srst ephone template 5 Which ephone-dn would be expected upon activation of SRST? A. ephone-dn 1 dual-linenumber 7001description 7001name 7001ephone-dn-template 5This DN is learned from srst fallback ephones B. ephone-dn 1 dual-linenumber 7001description 7001name 7001ephone-dn-template 3This DN is learned from srst fallback ephones C. ephone-dn 1number 7001description 7001name 7001ephone-dn-template 5This DN is learned from srst fallback ephones D. ephone-dn 1number 7001description 7001name 7001ephone-dn-template 3This DN is learned from srst fallback ephones Answer: A Q174. Which statement is correct about AAR? A. The end users see, "Network Congestion Rerouting?" but AAR is otherwise transparent to the end user and works without user intervention. B. AAR will display "not enough bandwidth" on the IP phone while it reroutes the call. C. AAR allows calls to be rerouted because of insufficient Cisco Unified Border Element controlled bandwidth to an ITSP. D. AAR allows calls to be rerouted due to insufficient gatekeeper controlled IP WAN bandwidth. Answer: A Explanation: Incorrect Answer: B, C, D Automated alternate routing (AAR) provides a mechanism to reroute calls through the PSTN or other network by using an alternate number when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth. With automated alternate routing, the caller does not need to hang up and redial the called party. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b03aar.ht ml Q175. Which commands are needed to configure Cisco Unified Communications Manager Express in SRST mode? A. telephony-service and srst mode B. telephony-service and moh C. call-manager-fallback and srst mode D. call-manager-fallback and voice-translation Answer: A Exact 300-075 prep:Q176. Which command displays the detailed configuration of all the Cisco Unified IP phones, voice ports, and dial peers of the Cisco Unified SRST router? A. show call-manager-fallback all B. show dial-peer voice summary C. show ephone summary D. show voice port summary Answer: A Q177. Which trunk should you use in an H.323 gatekeeper-controlled network? A. H.323 B. H.225 C. SIP D. Intercluster E. MGCP FXO trunk F. MGCP T1/E1 trunk Answer: B Q178. Which action configures CAC utilizing only Cisco Unified Communications Manager software? A. Configure Cisco Unified Communications Manager regions. B. Configure Cisco Unified Communications Manager locations. C. Configure Cisco Unified Communications Manager RSVP-enabled locations. D. Configure Cisco Unified Communications Manager MTPs. Answer: B Q179. You are the Cisco Unified Communications Manager in Certpaper.com. You use a remote site MGCP gateway to provide redundancy when connectivity to the central Cisco Unified Communications Manager cluster is lost. How to enable IP phones to establish calls to the PSTN when they have registered with the gateway? (Choose three.) A. POTS dial peers must be added to the gateway to route calls from the IP phones to the PSTN. B. The default service must be enabled globally. C. The command ccm-manager mgcp-fallback must be configured. D. COR needs to be configured to disallow outbound calls. Answer: A,B,C Explanation: Incorrect Answer: D Class of restriction: Cisco Unified Communications Manager Business Edition 3000 supports class of service (CoS) with respect to geographic reach as follows: – Campus – Local – National – International – Emergency services . Call waiting . Default ringtones . MoH . Speed dials: Single-button, not BLF Link: http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps11370/data_sheet_ c78-651909.html Q180. Which technologies provide remote-site redundancy for Cisco IP Phones during a WAN failure? A. SRST and MGCP fallback B. SRST and TEHO C. TEHO and MGCP fallback D. SRST and AAR Answer: A