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Exam Code: C_THR95_2405

Exam Name: SAP Certified Associate - Implementation Consultant - SAP SuccessFactors Career Development Planning and Mentoring

Version: V13.25

Q & A: 72 Questions and Answers

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NEW QUESTION: 1
An administrator needs to update the workspaces schema using the authoring server schema update tool. Which function is supported by the authoring server update tool?
A. Change an existing Managed Content table to an Unmanaged table
B. Drop a table in the base schema
C. Change an existing Unmanaged table to a Managed Content table
D. Recreate workspaces schema table name
Answer: C

NEW QUESTION: 2


Answer:
Explanation:

Explanation


NEW QUESTION: 3
A customer is in the planning stages of deploying a Cisco Unified Communications solution for their company. Previously, they were leasing a traditional PBX system from the telco and they have very little experience with voice. The customer wants to know what two signaling methods between the IP phone and the Cisco Unified Communications Manager Express are available for their use. (Choose two)
A. H.323
B. MGCP
C. RTP
D. SCCP
E. SIP
Answer: D,E
Explanation:
Skinny Client Control Protocol (SCCP)
SCCP uses Cisco-proprietary messages to communicate between IP devices and Cisco CallManager. SCCP easily coexists in a multiple protocol environment. The Cisco IP Phone is an example of a device that registers and communicates with Cisco CallManager as an SCCP client. During registration, a Cisco IP phone receives its line and all other configurations from Cisco CallManager. Once it registers, it is notified of new incoming calls and can make outgoing calls. The SCCP protocol is used for VoIP call signaling and enhanced features such as Message Waiting Indication (MWI).
Session Initiation Protocol (SIP)
The Internet Engineering Task Force (IETF) developed the SIP standard for multimedia calls over IP. ASCII-based SIP works in client/server relationships as well as in peer-to-peer relationships. SIP uses requests and responses to establish, maintain, and terminate calls (or sessions) between two or more end points. Refer to the Understanding Session Initiation Protocol (SIP) chapter for more information on SIP and the interaction between SIP and Cisco Call Manager.
H.323 Protocol
The International Telecommunications Union (ITU) developed the H.323 standard for multimedia communications over packet networks. As such, the H.323 protocol is a proven ITU standard and provides multivendor interoperability. The H.323 protocol specifies all aspects of multimedia application services, signaling, and session control over an underlying packet network. Audio is standard on H.323 networks, but networks can be scaled to include both video and data. The H.323 protocol can be implemented in large enterprise networks or can be deployed over an existing infrastructure, which makes H.323 an affordable solution.
The basic components of the H.323 protocol are terminals, gateways, and gatekeepers (which provide call control to H.323 endpoints). Similar to other protocols, H.323 applies to point-to-point or multipoint sessions. However, compared to MGCP, H.323 requires more configuration on the gateway.
Media Gateway Control Protocol (MGCP)
MGCP provides Cisco CallManager a powerful, flexible and scalable resource for call control. Cisco CallManager uses MGCP to control media on the telephony interfaces of a remote gateway and also uses MGCP to deliver messages from a remote gateway to appropriate devices.
MGCP enables a call agent (media gateway controller) to remotely control and manage voice and data communication devices at the edge of multiservice IP packet networks. Because of its centralized architecture, MGCP simplifies the configuration and administration of voice gateways and supports multiple (redundant) call agents in a network. MGCP does not provide security mechanisms such as message encryption or authentication.
Using MGCP, Cisco CallManager controls call processing and routing and provides supplementary services to the gateway. The MGCP gateway provides call preservation (the gateway maintains calls during failover and fallback), redundancy, dial-plan simplification (the gateway requires no dial-peer configuration), hookflash transfer, and tone on hold. MGCP-controlled gateways do not require a media termination point (MTP) to enable supplementary services such as hold, transfer, call pickup, and call park. If the MGCP gateway loses contact with its Cisco Call Manager, it falls back to using H.323 control to support basic call handling of FXS, FXO, T1 CAS, and T1/E1 PRI interfaces.
Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_0_1/ccmsys/a08procl.html

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