AudioCodes Certified Associate (ACA) Exam — Questions and Answers

Question 1: Which DSCP value is recommended by Cisco and widely used for marking VoIP RTP media traffic to ensure priority queuing?

BE (DSCP 0)
EF (Expedited Forwarding, DSCP 46) (Correct answer)
AF41 (DSCP 34)
CS3 (DSCP 24)

Correct answer: EF (Expedited Forwarding, DSCP 46)

EF (Expedited Forwarding, DSCP 46) is the standard DSCP marking for real-time voice RTP traffic, ensuring it receives lowest latency and jitter in network queues.

Question 2: Which RTP header field is used by AudioCodes and VoIP devices to detect out-of-order or duplicate packets?

SSRC
Sequence Number (Correct answer)
Timestamp
Payload Type

Correct answer: Sequence Number

The RTP Sequence Number increments by one for each packet sent, allowing receivers to detect missing, out-of-order, or duplicate packets.

Question 3: What does a SIP '488 Not Acceptable Here' response indicate during call troubleshooting?

The SIP URI was not found in the dial plan
The destination is busy
The called party does not support any of the codecs offered in the SDP (Correct answer)
The SIP proxy requires authentication

Correct answer: The called party does not support any of the codecs offered in the SDP

A 488 response means the remote end rejected the SDP offer because none of the proposed codecs, media types, or parameters are acceptable, indicating a codec mismatch.

Question 4: What does the AudioCodes SNMP trap 'acAlarmsActiveAlarmTableEntry' indicate when received by a monitoring system?

A firmware upgrade has completed successfully
A new active alarm has been raised on the device requiring operator review (Correct answer)
A call quality threshold has been crossed
A SIP registration has expired

Correct answer: A new active alarm has been raised on the device requiring operator review

The acAlarmsActiveAlarmTableEntry SNMP trap notifies the management system that a new alarm entry has been added to the active alarm table, requiring attention.

Question 5: What is 'SIP Over WebSocket (WSS)' and why is it relevant to AudioCodes deployments?

A transport for SIP signaling over encrypted WebSocket connections, enabling browser-based WebRTC clients to connect to AudioCodes SBCs (Correct answer)
A SIP compression scheme for low-bandwidth links
A firewall traversal method for SRTP
A media transport replacing RTP for low-latency audio

Correct answer: A transport for SIP signaling over encrypted WebSocket connections, enabling browser-based WebRTC clients to connect to AudioCodes SBCs

SIP over WSS tunnels SIP signaling in encrypted WebSocket connections, allowing WebRTC browsers and web-based clients to register and call through an AudioCodes SBC.

Question 6: Which AudioCodes CLI command is used to display the current running configuration on a Mediant gateway?

display config
show running-config (Correct answer)
get running
list config

Correct answer: show running-config

The 'show running-config' command displays the active configuration currently loaded in memory on AudioCodes Mediant devices.

Question 7: Which AudioCodes HA state indicates the unit is fully synchronized and prepared to assume the active role?

Warm Standby
Passive Monitor
Hot Standby (Correct answer)
Initializing

Correct answer: Hot Standby

A unit in 'Hot Standby' state is fully synchronized with the active unit and can assume the active role immediately without any delay or data loss.

Question 8: What does 'SRTP with SDES key exchange' mean in an AudioCodes SBC context?

Keys are exchanged using DTLS before media starts
SRTP encryption keys are exchanged in-band within the SDP offer/answer, relying on SIP TLS for key security (Correct answer)
Keys are derived from a pre-shared password
Keys are negotiated via MIKEY protocol

Correct answer: SRTP encryption keys are exchanged in-band within the SDP offer/answer, relying on SIP TLS for key security

SDES (Session Description Protocol Security Descriptions) embeds SRTP key material directly in SDP, making TLS mandatory for the SIP signaling to protect those keys.

Question 9: What does 'ICMP Port Unreachable' response to SIP UDP packets indicate in an AudioCodes deployment?

The codec negotiation was rejected
The remote SIP server is not listening on the destination UDP port, confirming the SIP service is down or misconfigured (Correct answer)
A firewall is rate-limiting SIP packets
The remote server rejected the SIP credentials

Correct answer: The remote SIP server is not listening on the destination UDP port, confirming the SIP service is down or misconfigured

An ICMP Port Unreachable message in response to a UDP SIP packet means no application is listening on that port at the destination, indicating the SIP service is stopped or on a different port.

Question 10: What does 'Registration Account' configuration on an AudioCodes gateway enable?

SNMP trap destination registration
DNS-based SRV record advertisement
Local user authentication for the web GUI
The gateway to register a SIP URI with a remote proxy on behalf of connected endpoints (Correct answer)

Correct answer: The gateway to register a SIP URI with a remote proxy on behalf of connected endpoints

Registration Accounts allow the gateway to send SIP REGISTER messages to a remote proxy, enabling the device to appear as a registered SIP UA for inbound routing.

Question 11: What is AudioCodes known for?

Developing voice networking and media processing solutions (Correct answer)
Offering traditional telephone services.
Manufacturing mobile phones.
Providing cloud-based solutions only.

Correct answer: Developing voice networking and media processing solutions

AudioCodes is a leading vendor known for its expertise in developing a wide range of voice networking and media processing solutions. Their product portfolio includes VoIP gateways, Session Border Controllers (SBCs), IP phones, and management software, all designed to facilitate and enhance voice communication over IP networks for enterprises and service providers.

Question 12: What is the recommended approach for upgrading firmware on an AudioCodes HA pair to minimize downtime?

Disable HA, upgrade both units independently, then re-enable HA
Upgrade both units simultaneously to avoid version mismatches
Upgrade the standby unit first, perform a manual switchover, then upgrade the originally-active unit (Correct answer)
Upload new firmware to the EMS and let it push to both units automatically at 2 AM

Correct answer: Upgrade the standby unit first, perform a manual switchover, then upgrade the originally-active unit

Upgrading the standby first, then switching over, then upgrading the other unit allows rolling upgrades with minimal (or zero) service disruption.

Question 13: What is the purpose of the AudioCodes 'Call Detail Record (CDR)' log?

To log SIP registration events only
To record per-call statistics including duration, codec, disconnect cause, and endpoint details for billing and troubleshooting (Correct answer)
To store SNMP trap history
To capture real-time RTP quality metrics per packet

Correct answer: To record per-call statistics including duration, codec, disconnect cause, and endpoint details for billing and troubleshooting

CDR logs capture a complete record for every call including caller/called ID, start/end times, duration, codec used, disconnect cause code, and IP addresses for auditing and billing.

Question 14: What is AudioCodes' primary focus in providing telecommunication solutions?

To manage mobile networks.
To sell enterprise hardware.
To provide consumer mobile phones.
To offer solutions for VoIP, unified communications, and media processing (Correct answer)

Correct answer: To offer solutions for VoIP, unified communications, and media processing

AudioCodes' primary focus in telecommunication solutions is to provide a broad portfolio for Voice over IP (VoIP), unified communications (UC), and media processing. They specialize in developing technologies and products that enable high-quality, secure, and reliable voice and multimedia communication over IP networks, catering to both enterprises and service providers.

Question 15: What is the role of network redundancy in VoIP systems?

To maintain network availability during failures (Correct answer)
To reduce the number of devices in use.
To increase network costs.
To improve network performance.

Correct answer: To maintain network availability during failures

Network redundancy in VoIP systems involves duplicating critical network components and paths. Its primary role is to provide failover capabilities, ensuring that if one component or link fails, another can immediately take over. This minimizes downtime, maintains continuous communication, and prevents service interruptions, which is essential for business continuity.

Question 16: AudioCodes Mediant SBC 'Teams Direct Routing' mode enables:

Microsoft Teams Phone System to use AudioCodes SBC as a certified gateway to PSTN SIP trunks (Correct answer)
Teams to perform transcoding without an SBC
Direct RTP media between Teams and PSTN bypassing the SBC
AudioCodes to act as a Microsoft Teams client

Correct answer: Microsoft Teams Phone System to use AudioCodes SBC as a certified gateway to PSTN SIP trunks

Teams Direct Routing allows enterprises to connect their own PSTN SIP trunks through a Microsoft-certified SBC (such as AudioCodes Mediant) to the Teams Phone System cloud.

Question 17: In AudioCodes Mediant, what is the purpose of configuring 'Media Realms'?

To group media gateway channels by capacity
To assign SRTP profiles to call legs
To set codec priorities per geographic region
To define the network interfaces and IP address ranges used for RTP media streams, enabling media routing across multiple NICs (Correct answer)

Correct answer: To define the network interfaces and IP address ranges used for RTP media streams, enabling media routing across multiple NICs

Media Realms specify which network interfaces and IP subnets are used for RTP media, allowing the device to correctly route media on the right interface in multi-homed deployments.

Question 18: Which AudioCodes SBC security feature rate-limits SIP messages to protect against DoS and flooding attacks?

SNMP ACL
SIP Message Policy (SIP Whitelist/Blacklist and rate limiting) (Correct answer)
IP Group Bandwidth Limit
Firewall Rule Sets

Correct answer: SIP Message Policy (SIP Whitelist/Blacklist and rate limiting)

AudioCodes SBC includes SIP message classification and rate-limiting policies that block or throttle excessive SIP messages from unknown or suspicious sources, mitigating DoS attacks.

Question 19: On a Mediant gateway, what does a 'proxy set' configuration define?

The media codec priority list
A group of SIP proxy servers with load balancing and redundancy settings (Correct answer)
The VLAN tagging rules for voice traffic
The TLS certificate binding

Correct answer: A group of SIP proxy servers with load balancing and redundancy settings

A proxy set defines one or more SIP proxy server addresses along with load balancing mode, keep-alive settings, and redundancy behavior.

Question 20: In AudioCodes Mediant, 'Voice Activity Detection (VAD)' when enabled will:

Amplify quiet speech to improve intelligibility
Suppress RTP transmission during silence to conserve bandwidth using comfort noise generation (Correct answer)
Detect caller ID tones from the PSTN
Monitor for echo and trigger echo cancellation

Correct answer: Suppress RTP transmission during silence to conserve bandwidth using comfort noise generation

VAD detects silence periods and suspends RTP transmission, replacing it with Comfort Noise Generation (CNG) signals, reducing bandwidth usage during pauses in speech.

Question 21: What is the main advantage of using a VoIP system over traditional phone lines?

Higher call quality.
Better security.
Lower cost and greater flexibility (Correct answer)
Better sound clarity.

Correct answer: Lower cost and greater flexibility

The main advantage of using a VoIP system over traditional phone lines is the significant reduction in communication costs, especially for long-distance and international calls, as calls are routed over the internet. Additionally, VoIP offers greater flexibility with features like remote work capabilities, easy scalability, and integration with other business applications, which are often more complex or expensive with traditional systems.

Question 22: In a Microsoft Teams Direct Routing deployment with AudioCodes SBC, what causes a 'TLS Certificate Not Trusted' error during SIP trunk establishment?

The SIP proxy port is blocked by a firewall
The SRTP profile does not match
The SBC certificate is self-signed or its CA is not in Microsoft's trusted root certificate store (Correct answer)
The SIP trunk FQDN contains an underscore

Correct answer: The SBC certificate is self-signed or its CA is not in Microsoft's trusted root certificate store

Microsoft Teams requires SBC certificates signed by a CA in its trusted root store; self-signed or private CA certificates will cause TLS handshake failures during trunk establishment.

Question 23: What is the main advantage of using SIP (Session Initiation Protocol) in VoIP communication?

It enhances video quality.
It ensures a secure connection.
It enables communication between different devices and networks (Correct answer)
It increases bandwidth usage.

Correct answer: It enables communication between different devices and networks

SIP (Session Initiation Protocol) is a signaling protocol used to initiate, maintain, and terminate real-time communication sessions, including voice and video calls. Its main advantage is its ability to facilitate interoperability, allowing various VoIP devices and applications from different vendors to communicate seamlessly across diverse networks. This open standard ensures flexibility and broad compatibility in unified communications environments.

Question 24: How does AudioCodes handle PSTN/TDM trunk state during an HA failover event?

PSTN traffic is rerouted to a third backup gateway until the HA pair recovers
The standby unit has pre-configured identical trunk parameters and reconnects trunks automatically after taking the active role (Correct answer)
PSTN trunks are temporarily bridged between active and standby using a shared TDM bus
All PSTN channels are dropped and must be re-established by the PSTN carrier after switchover

Correct answer: The standby unit has pre-configured identical trunk parameters and reconnects trunks automatically after taking the active role

Because the standby unit has synchronized trunk configurations, it can re-establish PSTN connections after assuming the active role, minimizing recovery time.

Question 25: What is the role of AudioCodes Mediant media gateways?

They provide security for SIP communications.
They connect VoIP networks to traditional telephony systems (Correct answer)
They connect mobile phones to the internet.
They monitor network traffic.

Correct answer: They connect VoIP networks to traditional telephony systems

AudioCodes Mediant media gateways are designed to bridge the gap between modern VoIP networks and traditional telephony systems, such as the Public Switched Telephone Network (PSTN) or legacy PBXs. They convert voice signals between IP packets and circuit-switched TDM (Time Division Multiplexing) formats. This allows organizations to integrate their existing phone infrastructure with new VoIP deployments.

Question 26: Which AudioCodes tool provides real-time SIP message tracing directly from the web GUI without needing an external packet capture?

SIP Ladder Diagram (Message Log / Syslog viewer) (Correct answer)
SNMP MIB browser
CDR report generator
Wireshark plugin

Correct answer: SIP Ladder Diagram (Message Log / Syslog viewer)

The AudioCodes web GUI's Message Log displays a real-time SIP ladder diagram showing INVITE, 100 Trying, 180 Ringing, and other messages with timestamps for quick troubleshooting.

Question 27: Which AudioCodes command-line utility can be used to test IP connectivity from the device to a remote SIP server?

ping (via CLI 'ping' command) (Correct answer)
nslookup sip
sip-test utility
traceroute sip

Correct answer: ping (via CLI 'ping' command)

AudioCodes Mediant CLI supports the standard 'ping' command to test basic ICMP reachability from the device's IP interface to a remote address.

Question 28: AudioCodes 'Syslog' messages are categorized by severity. Which severity level indicates a critical failure requiring immediate attention?

Debug (7)
Informational (6)
Emergency (0) or Critical (2) (Correct answer)
Warning (4)

Correct answer: Emergency (0) or Critical (2)

Syslog severity levels 0 (Emergency) and 2 (Critical) represent the most severe conditions — system failures or imminent failures that require immediate operator action.

Question 29: What does 'SIP Trunking' over an AudioCodes SBC fundamentally replace in legacy enterprise telephony?

H.323 video conferencing trunks
Analog POTS lines to fax machines only
Internal IP phone-to-PBX signaling
Traditional ISDN/PRI/BRI physical PSTN trunks with IP-based SIP connections to a service provider (Correct answer)

Correct answer: Traditional ISDN/PRI/BRI physical PSTN trunks with IP-based SIP connections to a service provider

SIP trunking replaces physical ISDN, PRI, or BRI circuits with IP-based SIP connections to a service provider, delivering PSTN calling over the enterprise IP network.

Question 30: AudioCodes 'Survivability' feature on a branch SBC ensures:

Local calls continue to function during WAN outage by the SBC registering endpoints and routing calls locally (Correct answer)
CDR records are buffered and sent to OVOC when WAN restores
Remote firmware upgrades proceed during WAN outages
SRTP keys are refreshed automatically on WAN recovery

Correct answer: Local calls continue to function during WAN outage by the SBC registering endpoints and routing calls locally

Survivability mode activates when the SBC loses connectivity to the central call control system, allowing it to register local endpoints and route intra-branch calls independently.

Question 31: What does 'DTLS-SRTP' key exchange provide compared to SDES?

DTLS-SRTP provides higher audio quality
DTLS-SRTP exchanges keys directly in the media plane using DTLS without depending on SIP TLS, providing independent media-layer security (Correct answer)
DTLS-SRTP is only used for video streams
DTLS-SRTP uses a pre-shared key so no certificate is needed

Correct answer: DTLS-SRTP exchanges keys directly in the media plane using DTLS without depending on SIP TLS, providing independent media-layer security

DTLS-SRTP negotiates SRTP keys via a DTLS handshake on the media port itself, making the encryption independent of SIP signaling security and aligning with WebRTC requirements.

Question 32: What is the purpose of a VoIP gateway?

It converts voice signals from analog to digital (Correct answer)
It encrypts VoIP calls.
It secures VoIP networks.
It converts digital signals to text.

Correct answer: It converts voice signals from analog to digital

A VoIP gateway serves as a bridge between traditional analog telephony systems and modern IP networks. Its primary purpose is to convert analog voice signals from conventional phones or PBXs into digital packets that can be transmitted over an IP network, and vice versa. This conversion allows legacy equipment to integrate with VoIP systems, facilitating hybrid communication environments.

Question 33: Which AudioCodes tool provides centralized management, firmware upgrades, and configuration for multiple devices?

AudioCodes SNMP Manager
AudioCodes EMS
AudioCodes SysLog Server
AudioCodes One Voice Operations Center (OVOC) (Correct answer)

Correct answer: AudioCodes One Voice Operations Center (OVOC)

OVOC (One Voice Operations Center) is AudioCodes' centralized network management platform for monitoring, configuring, and upgrading devices at scale.

Question 34: During an AudioCodes HA switchover, how is configuration consistency between the active and standby maintained?

Configuration is stored only on the EMS server and downloaded by whichever unit becomes active
Administrators must manually push configuration updates to both units separately
Configuration is continuously synchronized from the active unit to the standby unit over the HA link (Correct answer)
The standby unit polls the active unit for configuration changes every 24 hours

Correct answer: Configuration is continuously synchronized from the active unit to the standby unit over the HA link

AudioCodes HA continuously replicates the active unit's configuration and call state to the standby unit via the HA link, ensuring immediate readiness.

Question 35: What does 'Alternative Routing' (fallback routing) configuration in AudioCodes do when a primary SIP trunk call fails?

Holds the call and alerts an administrator
Plays a recorded announcement and terminates the call
Automatically retries the call through a secondary route or trunk group based on configured failure reason codes (Correct answer)
Sends a SIP redirect to the calling party

Correct answer: Automatically retries the call through a secondary route or trunk group based on configured failure reason codes

Alternative Routing allows the gateway to detect specific SIP failure responses (e.g., 503, 408) and automatically retry the call via a configured secondary route without caller intervention.

Question 36: Which AudioCodes CLI command shows active calls and their current state in real time?

show rtp streams
debug sip call
show voip calls active (Correct answer)
list sessions current

Correct answer: show voip calls active

The 'show voip calls active' command displays all currently active call legs with their SIP state, codec, and media information on AudioCodes Mediant devices.

Question 37: In AudioCodes SBC for Teams Direct Routing, which SIP transport is mandated by Microsoft for the SBC-to-Teams signaling connection?

UDP
TCP without encryption
SCTP
TLS (TCP with TLS encryption) (Correct answer)

Correct answer: TLS (TCP with TLS encryption)

Microsoft requires TLS for all SIP signaling between the certified SBC and the Microsoft Teams infrastructure to ensure signaling security and meet compliance requirements.

Question 38: What is the purpose of using VLANs (Virtual Local Area Networks) in VoIP networks?

To connect multiple networks.
To reduce the number of devices in the network.
To increase security.
To prioritize VoIP traffic and reduce congestion (Correct answer)

Correct answer: To prioritize VoIP traffic and reduce congestion

VLANs (Virtual Local Area Networks) segment a physical network into multiple logical networks. In VoIP, creating a dedicated VLAN for voice traffic isolates it from other data traffic, preventing congestion and ensuring consistent performance. This isolation allows for easier application of Quality of Service (QoS) policies, prioritizing voice packets and improving call quality.

Question 39: AudioCodes Mediant supports which type of jitter buffer that automatically adjusts its depth based on network conditions?

Static jitter buffer
Adaptive jitter buffer (Correct answer)
Buffered playout jitter
Fixed jitter buffer

Correct answer: Adaptive jitter buffer

The adaptive jitter buffer dynamically increases or decreases its depth in real time based on measured network jitter, balancing latency against packet loss concealment.

Question 40: Which of the following is a core product offered by AudioCodes?

Audio equipment.
Mobile phone applications.
Content delivery networks.
VoIP gateways and session border controllers (Correct answer)

Correct answer: VoIP gateways and session border controllers

AudioCodes is particularly well-known for its core products, which include VoIP gateways and Session Border Controllers (SBCs). VoIP gateways enable connectivity between traditional telephony networks and IP networks, while SBCs provide security, interoperability, and quality assurance for VoIP communications at the network border. These products are fundamental to modern unified communications deployments.

Question 41: What is the primary purpose of a jitter buffer in VoIP media processing?

To absorb network delay variation and deliver audio packets to the codec at a steady rate (Correct answer)
To detect and suppress background noise
To compress audio samples before transmission
To reorder out-of-sequence TCP segments

Correct answer: To absorb network delay variation and deliver audio packets to the codec at a steady rate

A jitter buffer temporarily stores arriving RTP packets to smooth out variable network delays, providing a steady stream of audio samples to the decoder.

Question 42: What is the primary purpose of the AudioCodes High Availability (HA) configuration?

To enable geographic routing of SIP calls across multiple data centers
To increase call processing capacity by load-balancing across two units
To synchronize configuration changes across a cluster of gateways
To ensure continuous service by providing automatic failover to a standby unit (Correct answer)

Correct answer: To ensure continuous service by providing automatic failover to a standby unit

AudioCodes HA provides continuous service availability by automatically switching traffic to a standby unit when the active unit fails.

Question 43: What is the purpose of an AudioCodes session border controller (SBC)?

It connects devices to the internet.
It manages Wi-Fi networks.
It provides call routing between networks.
It ensures secure and efficient VoIP communications (Correct answer)

Correct answer: It ensures secure and efficient VoIP communications

An AudioCodes Session Border Controller (SBC) is a critical component that sits at the border of an enterprise or service provider network. Its purpose is to ensure secure and efficient VoIP communications by providing functions such as network security (firewall, denial-of-service protection), interoperability between different SIP networks, and quality of service (QoS) management. It acts as a gatekeeper for VoIP traffic.

Question 44: What attack does 'SIP Digest Authentication' (using 401/407 challenges) protect against in VoIP environments?

DNS poisoning of SIP proxy records
BGP route hijacking of SIP trunks
Unauthorized call origination by requiring valid credentials before processing INVITE or REGISTER requests (Correct answer)
Eavesdropping on RTP media streams

Correct answer: Unauthorized call origination by requiring valid credentials before processing INVITE or REGISTER requests

SIP Digest Authentication uses a challenge-response mechanism (MD5/SHA hash of credentials) to verify that the calling entity is authorized before allowing call setup or registration.

Question 45: An AudioCodes gateway shows 'Registration Failed' in the status page. Which troubleshooting step should be performed first?

Re-flash the firmware
Reboot the gateway
Check the SIP proxy address, port, credentials, and network reachability from the gateway (Correct answer)
Reload the factory default configuration

Correct answer: Check the SIP proxy address, port, credentials, and network reachability from the gateway

Registration failures are most commonly caused by incorrect proxy address/port, wrong credentials, or network connectivity issues, all of which should be verified before more disruptive steps.

Question 46: When troubleshooting AudioCodes OVOC connectivity to a managed device, which port must be open from the device to the OVOC server?

TCP 5060 (SIP) only
UDP 5004 (RTP) only
UDP 161 (SNMP) and TCP 443 (HTTPS) (Correct answer)
TCP 8080 (HTTP proxy)

Correct answer: UDP 161 (SNMP) and TCP 443 (HTTPS)

OVOC uses SNMP (UDP 161) for monitoring/traps and HTTPS (TCP 443) for configuration push and device communication, so both ports must be allowed through firewalls.

Question 47: What type of network does AudioCodes' solutions optimize?

Wireless local area networks.
Wired local area networks.
IP-based networks, especially for VoIP and unified communications (Correct answer)
Cloud-based networks.

Correct answer: IP-based networks, especially for VoIP and unified communications

AudioCodes is a leading vendor in the VoIP and unified communications space. Their solutions, such as Session Border Controllers (SBCs) and Media Gateways, are specifically designed to manage, secure, and optimize voice and video traffic over IP networks. This focus ensures high-quality, reliable communication for businesses adopting modern communication platforms.

Question 48: In AudioCodes cloud SBC deployments, 'High Availability (HA)' is typically achieved through:

Multiple independent SBCs with no shared state
Load balancing RTP across all available SBCs randomly
An active/standby pair of SBC instances with automatic failover and shared configuration synchronization (Correct answer)
A single SBC with RAID storage redundancy

Correct answer: An active/standby pair of SBC instances with automatic failover and shared configuration synchronization

AudioCodes HA pairs run one active and one standby SBC instance that synchronize configuration and call state, with the standby automatically taking over if the active node fails.

Question 49: What is 'SPIT' (Spam over Internet Telephony) and which AudioCodes feature helps mitigate it?

A SIP header spoofing technique; mitigated with TLS
A type of RTP injection attack; mitigated with SRTP
A codec negotiation exploit; mitigated with SDP filtering
Unsolicited bulk VoIP calls; mitigated via SIP blacklists, CAPTCHA challenges, and call rate limiting (Correct answer)

Correct answer: Unsolicited bulk VoIP calls; mitigated via SIP blacklists, CAPTCHA challenges, and call rate limiting

SPIT refers to automated spam calls over VoIP; AudioCodes SBC counters it using SIP source blacklisting, registration rate limiting, and anomaly detection policies.

Question 50: Which AudioCodes gateway feature allows automatic configuration retrieval from a provisioning server on boot?

Zero-Touch Provisioning (ZTP) (Correct answer)
Auto-Config Push
Bulk Config Loader
TFTP Boot Manager

Correct answer: Zero-Touch Provisioning (ZTP)

Zero-Touch Provisioning allows AudioCodes devices to automatically retrieve their configuration from a designated provisioning server upon first boot without manual intervention.

Question 51: When an AudioCodes Mediant gateway returns SIP response '503 Service Unavailable' to outbound calls, the most likely cause is:

An expired TLS certificate on the local device
A DTMF relay mismatch
The configured SIP proxy is unreachable or all proxy servers in the proxy set are down (Correct answer)
An incorrect codec configuration

Correct answer: The configured SIP proxy is unreachable or all proxy servers in the proxy set are down

A 503 response typically indicates the gateway cannot reach any of the configured SIP proxies, either because they are down or because network connectivity to them is lost.

Question 52: Which AudioCodes feature can detect when an incoming SIP REGISTER request uses a spoofed 'From' header and block it?

SIP Sanity Checks / Malformed SIP detection (Correct answer)
Topology Hiding
SRTP Enforcement
DTLS-SRTP negotiation

Correct answer: SIP Sanity Checks / Malformed SIP detection

AudioCodes SBC performs SIP sanity checking to detect malformed, inconsistent, or spoofed SIP headers, dropping messages that fail validation before they can affect call processing.

AudioCodes Certified Associate (ACA) Exam

The AudioCodes Certified Associate (ACA) certification validates foundational knowledge of AudioCodes products and solutions, focusing on voice over IP (VoIP) technologies.

Questions

60 min

Time Limit

70%

To Pass

Checking for saved progress...

Exam Rules

You can skip questions and return to them later
Flag questions for review before submitting
No feedback shown until you submit the entire exam
Unanswered questions count as wrong — answer everything
10 pretest questions are mixed in and don't affect your score
Timer auto-submits when time runs out
Your progress is auto-saved every 30 seconds