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| Section | Weight | Objectives |
|---|---|---|
| Cisco IOS XE Gateway and Media Resources | 15% | - Troubleshoot IOS XE dial plans and dial peer matching - Describe IOS XE media resources (Conferencing, MOH, Transcoding) - Configure voice gateway elements (DTMF, Voice translation rules, Codec preference lists, VoIP and POTS dial peers) |
| Call Control | 25% | - Describe the Cisco Unified Communications Manager (CUCM) digit analysis process - Implement toll fraud prevention on CUCM - Configure globalized call routing in CUCM (Route patterns, Translation patterns, Standard Local Route Group) - Describe Mobile and Remote Access (MRA) |
| Collaboration Applications | 5% | - Configure Cisco Jabber (Switching modes, Deployment modes) - Configure Cisco Unity Connection (MWI, SIP integration) |
| Infrastructure and Design | 20% | - Describe the key design elements of the following: Sizing, Bandwidth, High availability, Disaster recovery, Dial plan, Security (certificates, SRTP, TLS) - Describe the purpose of Edge devices in the Cisco Collaboration architecture (Expressway, CUBE) - Troubleshoot network components in a Cisco collaboration solution (DNS, NTP, CDP/LLDP, IP connectivity, Firewall traversal) |
| Quality of Service (QoS) | 15% | - Describe the QoS requirements for these applications (Voice, Video, Data) - Configure LLQ (class map, policy map, service policy) - Describe problems that can lead to poor voice and video quality (Jitter, Latency, Packet loss) - Describe the class models for providing QoS (DiffServ, IntServ) |
| Protocols, Codecs, and Endpoints | 20% | - Troubleshoot these elements of a SIP conversation (SDP, Offer/Answer model, DTMF, Early media) - Identify the appropriate codec for a given scenario (G.711, G.729, Opus) - Configure and troubleshoot endpoint registration (On-premises, CUCM, Expressway) - Describe SIP interworking between CUCM and Cisco Expressway |
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NEW QUESTION # 315
An engineer builds the configuration on a Cisco IOS gateway for the dial-peers:
Which command is required to complete the configuration?
Answer: B
NEW QUESTION # 316
Refer to the exhibit. When making a call to a MRA client, what are the combinations of protocol on each of the different sections A-B-C?
Answer: B
Explanation:
https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X12-5/Cisco-Expressway-IP-Port-Usage-for-Firewall-Traversal-Deployment-Guide-X12-5.pdf
NEW QUESTION # 317
How are E.164 called-party numbers normalized on a globalized call-routing environment is Cisco Unified Communications Manger?
Answer: C
Explanation:
Normalized called-party numbers: E.164 format with the +prefix is used for external destination.
Therefore, called-number normalization is the result of globalization. Internal directory numbers are used for internal destinations. Normalization is achieved by stripping or translating the called numbers to internally used directory numbers.
NEW QUESTION # 318
Which action prevent toll fraud in Cisco Unified Communications Manager?
Answer: A
Explanation:
https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified- communications-manager-express/107626-cme-toll-fraud.html
NEW QUESTION # 319
Refer to the exhibit.
Which codec should an engineer select for a call made between "Dallas-REG' & "Austin-REG'?
Answer: C
Explanation:
The codec preference list for the "Dallas-REG" region is "Factory Default low loss". This list includes the following codecs in order of preference:
* G.729
* G.711
* OPUS
* MP4A-LATM
The codec preference list for the "Austin-REG" region is "Factory Default low loss". This list includes the following codecs in order of preference:
* G.729
* G.711
* OPUS
* MP4A-LATM
Since both regions have the same codec preference list, the codec that will be used for a call made between
"Dallas-REG" and "Austin-REG" is G.729.
G).729 is a narrowband speech codec that was developed by the ITU-T in 1988. It is a low-bitrate codec that provides good quality speech at a bitrate of 8 kbps. G.729 is widely used in VoIP applications and is the default codec for many VoIP systems.
G).711 is a wideband speech codec that was developed by the ITU-T in 1972. It is a high-bitrate codec that provides excellent quality speech at a bitrate of 64 kbps. G.711 is not as widely used as G.729 due to its high bitrate requirements.
OPUS is a lossy audio codec that was developed by the IETF in 2012. It is a low-bitrate codec that provides good quality speech at a bitrate of 6 kbps. OPUS is widely used in VoIP applications and is the default codec for many VoIP systems.
MP4A-LATM is a lossy audio codec that was developed by the IETF in 1999. It is a high-bitrate codec that provides excellent quality speech at a bitrate of 24 kbps. MP4A-LATM is not as widely used as G.729 or OPUS due to its high bitrate requirements.
NEW QUESTION # 320
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