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| Section | Weight | Objectives |
|---|---|---|
| Topic 1: Cisco IOS XE Gateway and Media Resources | 15% | - Configure voice gateway elements (DTMF, Voice translation rules, Codec preference lists, VoIP and POTS dial peers) - Describe IOS XE media resources (Conferencing, MOH, Transcoding) - Troubleshoot IOS XE dial plans and dial peer matching |
| Topic 2: Call Control | 25% | - Describe Mobile and Remote Access (MRA) - Implement toll fraud prevention on CUCM - Describe the Cisco Unified Communications Manager (CUCM) digit analysis process - Configure globalized call routing in CUCM (Route patterns, Translation patterns, Standard Local Route Group) |
| Topic 3: Collaboration Applications | 5% | - Configure Cisco Unity Connection (MWI, SIP integration) - Configure Cisco Jabber (Switching modes, Deployment modes) |
| Topic 4: Quality of Service (QoS) | 15% | - Describe problems that can lead to poor voice and video quality (Jitter, Latency, Packet loss) - Describe the QoS requirements for these applications (Voice, Video, Data) - Describe the class models for providing QoS (DiffServ, IntServ) - Configure LLQ (class map, policy map, service policy) |
| Topic 5: Infrastructure and Design | 20% | - Describe the key design elements of the following: Sizing, Bandwidth, High availability, Disaster recovery, Dial plan, Security (certificates, SRTP, TLS) - Troubleshoot network components in a Cisco collaboration solution (DNS, NTP, CDP/LLDP, IP connectivity, Firewall traversal) - Describe the purpose of Edge devices in the Cisco Collaboration architecture (Expressway, CUBE) |
| Topic 6: Protocols, Codecs, and Endpoints | 20% | - Identify the appropriate codec for a given scenario (G.711, G.729, Opus) - Troubleshoot these elements of a SIP conversation (SDP, Offer/Answer model, DTMF, Early media) - Configure and troubleshoot endpoint registration (On-premises, CUCM, Expressway) - Describe SIP interworking between CUCM and Cisco Expressway |
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NEW QUESTION # 124
Refer to the exhibit.
https://i.postimg.cc/C57TkczG/image.png
A call is falling to establish between two SIP Devices The called device answers with these SOP Which SOP parameter causes issue?
Answer: D
Explanation:
The RTP port is used to send and receive media packets during a call. If the RTP port is set to 0, the called device will not be able to send or receive media packets, and the call will fail.
The other options are not correct because:
* A. The calling device did not offer a ptime value: The ptime value is used to specify the amount of time between each media packet. If the calling device does not offer a ptime value, the called device will use the default value of 20 milliseconds.
* B. The media stream is set to sendonly: The media stream is set to sendonly when the called device is only able to send media packets, and not receive them. This is not a problem, and the call will still succeed.
* C. The payload for G.711ulaw must be 18: The payload for G.711ulaw is the type of media packet that is used. The payload must be set to 18 for G.711ulaw, but this is not a problem, and the call will still succeed.
NEW QUESTION # 125
An administrator is asked to implement toll fraud prevention in Cisco UCM, specifically to restrict off-net to off-net call transfers. How is this implemented?
Answer: A
Explanation:
a) Enforce ad-hoc conference restrictions: This will prevent users from creating ad-hoc conferences, but it will not prevent them from transferring calls.
c) Implement time-of-day routing: This will allow calls to be routed to different destinations based on the time of day, but it will not prevent users from transferring calls.
d) Use the correct route filters: This will allow calls to be filtered based on the destination, but it will not prevent users from transferring calls.
Explanation:
To restrict off-net to off-net call transfers, an administrator can set the "Block Offnet to Offnet Transfer" service parameter to "On". This will prevent users from transferring calls from one external number to another external number.
The other options are not correct because:
NEW QUESTION # 126
An engineer builds the configuration on a Cisco IOS gateway for the dial-peers:
Which command is required to complete the configuration?
Answer: A
NEW QUESTION # 127
Where is the hybrid local gateway for cloud calling deployed?
Answer: D
NEW QUESTION # 128
Refer to the exhibit.
What is the result of applying these two rules to a voice translation-profile for use with an ISDT1 PRI on a Cisco Voice gateway?
Answer: B
NEW QUESTION # 129
......
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