Cisco 300-815 Exam Dumps & Practice Test Questions

Question 1

Which two protocols are commonly used in Cisco Unified Communications Manager (CUCM) for call signaling? (Choose 2.)

A. SIP (Session Initiation Protocol)
B. SCCP (Skinny Client Control Protocol)
C. RTP (Real-time Transport Protocol)
D. HTTP (Hypertext Transfer Protocol)
E. SNMP (Simple Network Management Protocol)

Answer: A, B

Explanation:
In Cisco Unified Communications Manager (CUCM), call signaling is the process of setting up, managing, and terminating voice or video calls. This involves the exchange of messages that establish parameters such as codec selection, device capabilities, call setup, and teardown. CUCM uses several protocols for these tasks, but two protocols are predominantly used for call signaling: SIP (Session Initiation Protocol) and SCCP (Skinny Client Control Protocol).

Option A, SIP, is a widely adopted signaling protocol in the industry. It is based on a text-based format and is used to initiate, manage, and terminate voice and video sessions across IP networks. CUCM supports SIP for communication with IP phones, gateways, trunks, and other SIP-enabled devices. SIP provides more advanced capabilities and flexibility than older protocols and is often preferred in modern VoIP deployments due to its interoperability and scalability.

Option B, SCCP, is a Cisco proprietary signaling protocol. It was one of the original protocols used in CUCM environments for managing IP phone communications. SCCP is a lightweight protocol optimized for Cisco IP Phones, providing efficient communication with CUCM. Though newer deployments are often SIP-based, SCCP remains in use, especially in legacy environments with Cisco devices.

Option C, RTP, is a media transport protocol, not a signaling protocol. RTP handles the actual delivery of voice or video media during a call, once the signaling has established the session. It is essential for the operation of VoIP systems but is not involved in the signaling phase of the call.

Option D, HTTP, is used by CUCM for web-based management and XML services, but it does not play a role in call signaling between endpoints.

Option E, SNMP, is used for network monitoring and management. It allows administrators to monitor CUCM systems, gather statistics, and configure devices, but it is not a call signaling protocol.

To summarize, the two primary signaling protocols used by CUCM are SIP and SCCP. They facilitate session setup, modification, and teardown between endpoints in a voice or video communication system. Therefore, the correct answers are A and B.

Question 2

Which two of the following are required for a Cisco CallManager Express (CME) system to work? (Choose 2.)

A. A minimum of one Cisco Unified Communications Manager (CUCM) server
B. A Cisco router with a Voice Interface Card (VIC) for voice port connections
C. IP phones with compatible software loaded
D. A connection to the Internet for external calling
E. An external PSTN gateway for call routing

Answer: B, C

Explanation:
Cisco CallManager Express (CME) is a router-based VoIP call processing system designed for small to medium-sized businesses. It provides basic IP telephony features without requiring a full-scale Cisco Unified Communications Manager (CUCM) deployment. CME runs directly on Cisco Integrated Services Routers (ISRs) and is ideal for branch offices or standalone environments.

Option B, a Cisco router with a Voice Interface Card (VIC), is a requirement when analog or digital voice connections to the PSTN or legacy phone systems are needed. The VIC provides physical interfaces for FXO (Foreign Exchange Office) or FXS (Foreign Exchange Station) ports. While not every CME deployment uses a VIC (some use SIP trunks or Ethernet-only IP communications), having a VIC enables essential PSTN connectivity, making it a common and important hardware component in traditional CME setups.

Option C, IP phones with compatible software loaded, is absolutely necessary. CME supports various Cisco IP phones, and they must run firmware versions compatible with the IOS version on the router running CME. These phones communicate directly with the CME router using SCCP or SIP, depending on the configuration. Without IP phones (or compatible softphones), the CME system cannot function as an IP telephony solution.

Option A, a CUCM server, is not required for CME. CME is designed to operate independently of CUCM, making it ideal for standalone deployments. While CUCM and CME can coexist in larger environments, CME does not depend on CUCM to provide telephony services.

Option D, a connection to the Internet, is not mandatory for CME functionality. Internal calling between IP phones works without any external connectivity. Internet access may be required for SIP trunking or software updates, but it is not a strict requirement for a basic CME system to function.

Option E, an external PSTN gateway, is not a requirement if the CME router itself includes PSTN connectivity (via FXO ports or a SIP trunk). In many cases, the router running CME also acts as the voice gateway, eliminating the need for an external one.

In conclusion, the essential components for a functioning CME system are a Cisco ISR with voice capabilities (B) and compatible IP phones (C). These provide the hardware and endpoint infrastructure necessary for local VoIP services, making B and C the correct answers.

Question 3

Which two features are available with Cisco Expressways for remote access to UC services? (Choose 2.)

A. Secure SIP trunking
B. Mobile and remote access for Jabber clients
C. Integration with Microsoft Teams
D. High availability and disaster recovery configuration
E. Voice and video traffic encryption

Answer: B, E

Explanation:
Cisco Expressway is a core component of Cisco's collaboration architecture, primarily used to enable secure, seamless remote access to internal Unified Communications (UC) services. It is particularly effective in environments with remote users, such as employees working from home or mobile locations, without requiring VPN access.

Option B, mobile and remote access (MRA) for Jabber clients, is one of the primary features of the Cisco Expressway solution. This feature allows Cisco Jabber clients (and Cisco IP phones with supported firmware) to connect securely to internal services like presence, messaging, voice, and video from outside the corporate firewall. The MRA functionality is made possible by the collaboration of Expressway-C (inside the firewall) and Expressway-E (outside the firewall), creating a secure traversal for signaling and media traffic.

Option E, voice and video traffic encryption, is another important capability. Cisco Expressway supports encrypted signaling (via TLS) and media encryption (via SRTP) for both internal and remote users. This ensures that voice and video communications remain secure, even when traversing untrusted networks like the public internet. This encryption is especially critical in maintaining confidentiality, integrity, and compliance with privacy standards.

Option A, secure SIP trunking, is not directly provided by Expressway. While SIP trunks can be secured (e.g., with TLS), this is typically handled by Session Border Controllers (SBCs) or CUCM, not the Expressway system. Expressway focuses more on endpoint-to-infrastructure communication rather than infrastructure-to-infrastructure (e.g., trunking).

Option C, integration with Microsoft Teams, is not a native Expressway function. While Cisco does offer solutions for interoperability between Webex and Microsoft Teams, this is generally achieved using third-party services or other Cisco tools, not Expressway.

Option D, high availability and disaster recovery configuration, while important for any enterprise system, is not a core feature unique to Expressway. These are architectural choices or system design considerations rather than a direct feature of the Expressway platform.

Therefore, the most accurate features available through Cisco Expressway for remote access are B, mobile and remote access for Jabber clients, and E, voice and video traffic encryption.

Question 4

Which two troubleshooting tools are commonly used when diagnosing SIP-related issues in Cisco Unified Communications environments? (Choose 2.)

A. Cisco Unified Real-Time Monitoring Tool (RTMT)
B. Wireshark for packet capture and analysis
C. Cisco Jabber for video diagnostics
D. Cisco Prime Collaboration
E. Cisco Firepower Threat Defense

Answer: A, B

Explanation:
When diagnosing SIP-related issues in Cisco Unified Communications (UC) environments, administrators rely on tools that can capture, analyze, and correlate SIP signaling and media events. SIP (Session Initiation Protocol) is responsible for session establishment, call setup, tear down, and signaling in VoIP communications. Proper visibility into SIP messaging is crucial for identifying root causes of failed or degraded calls.

Option A, the Cisco Unified Real-Time Monitoring Tool (RTMT), is a commonly used application provided by Cisco for monitoring the health and performance of CUCM and related collaboration components. It allows administrators to view real-time statistics, collect logs, and access trace files, including those related to SIP signaling. RTMT is especially useful when pulling CallManager traces, which are needed to analyze SIP INVITE, 100 Trying, 180 Ringing, and 200 OK messages that define the call flow.

Option B, Wireshark, is a powerful packet capture and protocol analysis tool. It is widely used to inspect SIP message flows and RTP media streams on the network. Wireshark allows administrators to track SIP dialogs, check codec negotiation, identify malformed packets, and troubleshoot issues such as one-way audio, call setup failure, or unexpected call teardown. It is extremely valuable when SIP messages need to be analyzed at a very granular level.

Option C, Cisco Jabber, is a client application for voice, video, presence, and messaging. While it may indicate call quality or connection status, it is not a diagnostic or troubleshooting tool. It cannot provide logs or SIP trace information beyond the user interface.

Option D, Cisco Prime Collaboration, focuses on provisioning, assurance, and capacity planning. While it does offer some insight into call metrics and service assurance, it lacks the in-depth SIP-level analysis tools required for detailed troubleshooting.

Option E, Cisco Firepower Threat Defense (FTD), is a network security solution for intrusion prevention and firewall capabilities. Although it may detect and block malicious traffic, it is not used to troubleshoot SIP signaling or UC-specific issues.

In conclusion, the most effective tools for SIP-related troubleshooting in Cisco UC environments are RTMT, which provides trace log access, and Wireshark, which captures and analyzes SIP and RTP packets. Therefore, the correct answers are A and B.

Question 5

Which two features does Cisco Unified Communications Manager (CUCM) offer for high availability? (Choose 2.)

A. Cisco Unified CM HA for automatic failover of servers
B. Clustered CUCM servers for load balancing calls
C. Redundant links between CUCM and other network devices
D. Call forwarding to external mobile numbers in case of failure
E. Unified Communications Manager Express (CME) as a backup system

Answer: A, B

Explanation:
Cisco Unified Communications Manager (CUCM) is designed to offer high availability (HA) and redundancy for enterprise voice and video environments. HA ensures that communication services remain functional even in the event of hardware or software failures, thus minimizing downtime and preserving business continuity.

Option A, Cisco Unified CM HA for automatic failover of servers, refers to one of CUCM’s core HA features. In a CUCM cluster, multiple servers are deployed with designated roles (publisher and subscribers). If one subscriber fails, registered phones can automatically fail over to another subscriber, ensuring continuity. This automatic failover mechanism is built into the redundant server architecture, where phones are configured with primary, secondary, and tertiary CUCM nodes.

Option B, clustered CUCM servers for load balancing calls, is another important feature. In a CUCM cluster, the servers not only provide redundancy but also support load balancing, where call processing is distributed across multiple subscriber nodes. This design helps to ensure performance efficiency and provides scalability and fault tolerance. In larger environments, clustering across multiple geographic locations (Cluster over WAN) is also possible for enhanced resilience.

Option C, redundant links between CUCM and other network devices, is generally part of network-level redundancy, not specific to CUCM. While it’s a good practice to have network link redundancy, this is not a built-in CUCM feature, and its configuration is typically handled at the infrastructure level (switches, routers).

Option D, call forwarding to external mobile numbers, may be part of business continuity planning but is not considered a CUCM high availability mechanism. It requires preconfigured call forwarding settings and depends on external service availability, making it more of a fallback strategy than a native HA feature.

Option E, Unified Communications Manager Express (CME) as a backup, is a possibility in SRST (Survivable Remote Site Telephony) scenarios, but CME is a separate platform, not a high availability feature within CUCM itself. While it can provide limited fallback calling at branch offices during WAN outages, it's not part of the CUCM cluster-based high availability framework.

In summary, CUCM offers robust high availability through automatic server failover and clustering for load balancing, making A and B the correct answers.

Question 6

Which two Cisco Unified Communications Manager (CUCM) configurations are required to support a secure SIP trunk between CUCM and an external SIP provider? (Choose 2.)

A. SIP TLS configuration for encrypted signaling
B. Configuration of Media Termination Points (MTP)
C. Configuring SRTP for secure media encryption
D. Enabling SIP trunk registration with basic authentication
E. Configuring SIP route patterns for outgoing calls

Answer: A, C

Explanation:
To establish a secure SIP trunk between Cisco Unified Communications Manager (CUCM) and an external SIP provider, both signaling and media encryption must be configured. These configurations ensure that communications are secure, tamper-proof, and confidential across the untrusted network, typically the internet.

Option A, SIP TLS configuration for encrypted signaling, is a critical requirement. TLS (Transport Layer Security) encrypts SIP signaling messages, such as INVITE, ACK, and BYE, between CUCM and the SIP provider. Enabling TLS for the SIP trunk ensures that signaling data (such as call setup details and credentials) cannot be intercepted or modified in transit. In CUCM, this involves setting the SIP trunk security profile to use TLS and assigning appropriate certificates and trust lists to validate the connection.

Option C, SRTP for secure media encryption, is equally important. SRTP (Secure Real-time Transport Protocol) provides encryption, message authentication, and integrity for the RTP streams that carry the actual voice or video data. CUCM must be configured to use SRTP on the SIP trunk, ensuring that media packets are protected from eavesdropping or tampering.

Option B, configuration of Media Termination Points (MTP), may be necessary in certain interoperability scenarios (e.g., DTMF relay or codec mismatches), but MTP is not required specifically for secure SIP trunking. It's more of a media compatibility mechanism, not a security control.

Option D, enabling SIP trunk registration with basic authentication, is not typically required for secure SIP trunks. Many SIP trunks operate in peer-to-peer mode without registration. Moreover, basic authentication (username/password) is not inherently secure unless combined with TLS, and digest authentication is often used instead.

Option E, configuring SIP route patterns, is part of call routing, not directly related to securing the SIP trunk. While necessary for outbound calling, it does not contribute to the encryption or authentication aspects of SIP security.

In conclusion, the two most essential configurations for securing a SIP trunk are enabling TLS for signaling (A) and SRTP for media encryption (C). These ensure that both call setup and media streams are transmitted securely, meeting compliance and privacy requirements.

Question 7

Which two options are necessary to deploy Cisco Jabber on a mobile device for remote communication? (Choose 2.)

A. Cisco Unified Communications Manager Express (CME)
B. Cisco Expressway for mobile and remote access
C. Cisco Unity Connection for voicemail services
D. An active SIP trunk to the PSTN for mobile connectivity
E. Cisco Webex for Teams deployment

Answer: B, C

Explanation:
Cisco Jabber is a unified communications application that provides instant messaging, voice, video, voicemail, and presence services. When deploying Cisco Jabber for mobile devices and remote communication, the architecture must support secure, reliable access to internal UC services from outside the corporate network, without requiring VPN connections.

Option B, Cisco Expressway for mobile and remote access (MRA), is one of the most critical components. It enables Cisco Jabber clients to securely access CUCM, IM and Presence, and voicemail services over the internet. The Expressway deployment consists of Expressway-C (inside the DMZ) and Expressway-E (external-facing), working together to facilitate secure traversal of signaling and media traffic. Without Expressway, mobile devices outside the corporate firewall would not be able to register with CUCM.

Option C, Cisco Unity Connection for voicemail services, is essential for voicemail integration. When users deploy Cisco Jabber, they expect not just voice and video calling but also access to voicemail features like playback, message waiting indicators, and voicemail to email. Cisco Unity Connection provides these features and is integrated into the Jabber client experience, making it a necessary component in a full Jabber deployment.

Option A, Cisco Unified Communications Manager Express (CME), is not typically used for Cisco Jabber mobile deployments. CME is a standalone, router-based call processing system, and does not support the advanced Jabber feature set, particularly remote access via Expressway.

Option D, an active SIP trunk to the PSTN, is not a requirement for deploying Jabber on a mobile device. While a SIP trunk may be needed for external calling, it is not a prerequisite for remote Jabber deployment, which focuses on access to internal services over the internet.

Option E, Cisco Webex for Teams deployment, refers to a different collaboration platform. While Webex and Jabber are both Cisco products, they are not dependent on each other. Jabber uses CUCM, Expressway, and Unity Connection, while Webex Teams (now part of Webex App) functions as a cloud-based solution.

Therefore, the correct components required for mobile Jabber deployment are B and C.

Question 8

Which two of the following are common reasons for voice quality issues in a VoIP deployment? (Choose 2.)

A. High latency and jitter on the network
B. Misconfigured dial plans
C. Inadequate bandwidth for real-time communication
D. Outdated firmware on IP phones
E. Excessive packet loss during voice call transmission

Answer: A, E

Explanation:
Voice over IP (VoIP) relies heavily on network performance to maintain high-quality voice communication. Since voice is transmitted in real-time, it is particularly sensitive to latency, jitter, and packet loss. These impairments can significantly degrade audio quality, leading to choppy, delayed, or unintelligible speech.

Option A, high latency and jitter, is a major cause of voice quality problems. Latency refers to the time it takes for voice packets to travel from the sender to the receiver. If latency exceeds 150 milliseconds one-way, it can lead to noticeable delays in conversation, impacting the natural flow of dialogue. Jitter, on the other hand, is the variation in packet arrival times. High jitter causes packets to arrive out of order, which can result in audio glitches or gaps, especially if the jitter buffer is not sufficient to compensate.

Option E, excessive packet loss, is another common and severe issue. Voice traffic typically uses UDP, which does not retransmit lost packets. Even a packet loss rate of 1–2% can degrade call quality, making audio sound robotic or broken. Packet loss is often caused by network congestion, faulty hardware, or unstable links, and must be minimized for VoIP to function reliably.

Option B, misconfigured dial plans, while it can affect call routing and call setup, does not directly impact voice quality. It may prevent calls from being completed but doesn’t influence audio fidelity once the call is established.

Option C, inadequate bandwidth, is a potential contributor to poor quality but often manifests through packet loss or increased jitter/latency, which are more direct symptoms. Without enough bandwidth, the network may become congested, leading to the impairments noted above. However, bandwidth inadequacy is an indirect cause, whereas latency and packet loss are direct indicators of quality issues.

Option D, outdated firmware on IP phones, can cause functional bugs or compatibility issues, but it is not among the most common or immediate causes of audio degradation. Firmware issues tend to affect features and stability rather than real-time audio streams unless there's a known defect.

Therefore, the two most direct and common reasons for poor voice quality in VoIP deployments are A, high latency and jitter, and E, excessive packet loss.

Question 9

Which two troubleshooting steps should be taken if a Cisco IP phone cannot register with Cisco Unified Communications Manager (CUCM)? (Choose 2.)

A. Verify that the phone is connected to the correct VLAN and has network access
B. Check for a DHCP server providing the correct IP address and options for CUCM
C. Verify that the phone’s MAC address is registered in the CUCM database
D. Ensure that the phone has a valid TLS certificate for secure registration
E. Confirm the CUCM server is fully loaded with all system configurations

Answer: A, B

Explanation:
When a Cisco IP phone fails to register with Cisco Unified Communications Manager (CUCM), the issue is typically caused by network configuration errors, DHCP issues, or connectivity problems. Troubleshooting such issues involves ensuring that the phone can communicate with CUCM and is receiving all necessary network information to attempt registration.

Option A, verifying VLAN and network access, is one of the first steps to troubleshoot registration issues. Cisco IP phones often use Voice VLANs to separate voice traffic from data traffic. If a phone is not on the correct VLAN or cannot obtain an IP address, it will fail to reach CUCM. Administrators should ensure that the switch port is configured correctly, that CDP or LLDP is functioning properly (to instruct the phone which VLAN to use), and that the phone can ping the CUCM server.

Option B, checking for a DHCP server providing correct IP address and options, is another critical step. IP phones rely on DHCP not only to get an IP address but also to obtain Option 150 (or Option 66), which tells the phone where to find the TFTP server (typically the CUCM). Without this information, the phone cannot download its configuration file or proceed with the registration process. Ensuring the DHCP scope includes the proper options and that leases are being issued correctly is essential.

Option C, verifying the phone’s MAC address in the CUCM database, is not a required step for basic registration. While CUCM can be configured to use secure registration that verifies the MAC, most deployments allow non-preconfigured phones to register, assuming their device types are permitted. This step becomes relevant if the CUCM is restricting registration to known devices only, which is not always the case.

Option D, ensuring the phone has a valid TLS certificate, applies to secure SIP registration or encrypted environments, which are not always deployed. In non-secure setups, phones can register without certificates. Thus, unless the environment mandates secure device authentication, this step is unnecessary.

Option E, confirming the CUCM server is fully loaded with all configurations, is vague and non-specific. While proper configuration is vital, there’s no single “fully loaded” state that directly affects registration. The more relevant factor is whether the TFTP and CallManager services are running and reachable.

In summary, the two most effective initial troubleshooting steps are ensuring proper network access (A) and verifying DHCP configuration (B).

Question 10

Which two options are necessary for configuring high availability for a Cisco WebEx Calling deployment? (Choose 2.)

A. Configuring geographically distributed WebEx Calling data centers
B. Using DNS-based failover to automatically switch to backup servers
C. Implementing QoS policies for VoIP traffic
D. Deploying a dual-ISP connection for Internet redundancy
E. Enabling SIP trunks for direct communication with the PSTN

Answer: A, B

Explanation:
Cisco WebEx Calling is a cloud-based telephony solution, and high availability (HA) is a critical requirement for maintaining uninterrupted service in case of failures or disruptions. High availability in WebEx Calling relies primarily on geographic redundancy and intelligent routing to ensure continuous operation, even if a data center or connection becomes unavailable.

Option A, configuring geographically distributed WebEx Calling data centers, is essential for cloud-based high availability. Cisco WebEx Calling leverages a global architecture, with multiple data centers across regions. In a properly designed HA deployment, calls and services can fail over to another geographically distinct data center if the primary one becomes unreachable. This georedundancy allows for automatic or manual redirection of services, minimizing downtime for users.

Option B, using DNS-based failover, is also crucial. DNS SRV and A records can be used to prioritize and distribute service requests across multiple IP addresses or data centers. In the event that one server or data center becomes unavailable, the DNS system redirects the call setup or registration traffic to the backup server automatically, enabling seamless failover. This mechanism is widely used in cloud telephony and SIP deployments for fault tolerance.

Option C, implementing QoS (Quality of Service) policies, is vital for call quality but does not directly contribute to high availability. QoS helps ensure low latency and reduced jitter, but it doesn't offer resilience in the event of a system or network failure.

Option D, deploying a dual-ISP connection, while beneficial for internet redundancy, is not a WebEx Calling-specific HA mechanism. It is a local network-level design choice that improves access reliability but is not necessary for configuring HA within the WebEx Calling platform itself.

Option E, enabling SIP trunks to the PSTN, is typically not part of WebEx Calling's core architecture. WebEx Calling manages PSTN connectivity through Cisco cloud-connected PSTN (CCP) providers or local gateways, but SIP trunking to external carriers is not directly related to WebEx’s built-in HA mechanisms.

In conclusion, the key components of high availability in WebEx Calling are the geographic distribution of data centers (A) and DNS-based failover mechanisms (B). These features ensure service continuity even during major outages or disruptions.