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Question 1. Which protocol is primarily responsible for establishing, modifying, and terminating VoIP calls? A) RTP B) SIP C) DHCP D) ICMP Answer: B Explanation: SIP (Session Initiation Protocol) handles call setup, management, and teardown in VoIP environments. Question 2. In VoIP, what does the acronym RTP stand for? A) Real-Time Transfer Protocol B) Remote Telephony Protocol C) Real-Time Transport Protocol D) Routing Transfer Procedure Answer: C Explanation: RTP carries the media (audio/video) streams after a call is established. Question 3. Which protocol is used to monitor the quality of RTP streams? A) RTCP B) SIP C) TCP D) SNMP Answer: A Explanation: RTCP (Real-Time Control Protocol) provides feedback on packet loss, jitter, and delay.
Question 4. Which codec typically provides the highest voice quality at the expense of bandwidth? A) G. B) G. C) G. D) G. Answer: C Explanation: G.722 is a wideband codec offering superior audio quality but uses more bandwidth than narrowband codecs. Question 5. What is the primary difference between traditional PSTN telephony and VoIP? A) PSTN uses analog signals, VoIP uses digital packets. B) PSTN requires routers, VoIP does not. C) PSTN supports video, VoIP does not. D) PSTN uses SIP, VoIP uses SS7. Answer: A Explanation: PSTN transmits analog voice over circuit-switched networks, while VoIP packetizes voice for IP networks. Question 6. Which layer of the OSI model does SIP operate at? A) Physical B) Data Link C) Transport D) Application Answer: D Explanation: SIP is an application‑layer protocol that runs over TCP or UDP.
A) <20 ms B) <150 ms C) <500 ms D) <1 s Answer: B Explanation: Delays under 150 ms are generally imperceptible to users; higher delays degrade conversation quality. Question 11. Jitter in a VoIP network refers to: A) Packet loss rate B) Variation in packet arrival time C) Bandwidth consumption D) Encryption strength Answer: B Explanation: Jitter is the variability in packet inter‑arrival times, which can cause audio distortion if not buffered. Question 12. Which DHCP option provides the IP address of the SIP proxy server? A) Option 3 B) Option 6 C) Option 120 D) Option 150 Answer: D Explanation: Option 150 is commonly used in Cisco environments to deliver the SIP proxy address to VoIP phones. Question 13. In SIP URIs, the “sip:” scheme indicates:
A) Secure SIP over TLS B) Standard SIP over UDP/TCP C) SIP over HTTP D) SIP over SCTP Answer: B Explanation: “sip:” denotes standard SIP signaling; “sips:” would indicate SIP over TLS. Question 14. The E.164 numbering plan is used to: A) Define IP address ranges for VoIP. B) Standardize international telephone numbers. C) Assign MAC addresses to VoIP devices. D) Encode audio codecs. Answer: B Explanation: E.164 defines the format for global telephone numbers, essential for SIP address normalization. Question 15. Which component in a SIP network is responsible for protecting the network perimeter and handling NAT traversal? A) Proxy Server B) Registrar C) Session Border Controller (SBC) D) Redirect Server Answer: C Explanation: SBCs manage security, media anchoring, and NAT traversal for SIP traffic. Question 16. A Biamp Tesira SVC‑2 card supports which of the following? A) Only analog PSTN lines
Question 19. When configuring Acoustic Echo Cancellation (AEC) on a Biamp VoIP endpoint, the Far‑End reference path should be sourced from: A) The local microphone input B) The VoIP Receive block output C) The system’s line‑out jack D) The network switch port Answer: B Explanation: AEC requires the remote (far‑end) audio to reference echo; the Receive block provides this signal. Question 20. Which codec is mandatory for SIP compliance according to RFC 3261? A) G. B) G. C) G. D) Opus Answer: A Explanation: G.711 is the default mandatory codec for SIP; other codecs are optional. Question 21. In a Biamp VoIP web interface, the “Registration Expiration” field defines: A) How long the device stays logged in without re‑registering. B) The maximum call duration. C) The time to wait for DNS resolution. D) The interval for firmware updates. Answer: A Explanation: Registration Expiration sets the lease time for SIP registration before the device must renew.
Question 22. Which of the following is NOT a typical reason for a SIP registration failure on a Biamp device? A) Incorrect SIP password B) Wrong VLAN ID C) Mismatched codec list D) Invalid SIP domain name Answer: C Explanation: Codec mismatches affect media negotiation, not registration; registration failures stem from authentication or network settings. Question 23. In a Wireshark capture of a Biamp VoIP device, a “401 Unauthorized” SIP response indicates: A) Network congestion B) Authentication failure C) Codec incompatibility D) Media path error Answer: B Explanation: A 401 response signals that the server requires authentication credentials, which may be missing or incorrect. Question 24. Which Biamp product is specifically designed for multi‑site conferencing with SIP support? A) Vocia MS‑ 1 B) Tesira FORTÉ VT C) Audia VoIP‑ 2 D) Tesira SVC‑ 2 Answer: D
Explanation: The Console block provides user‑interface functions such as answering or ending calls. Question 28. Which of the following QoS parameters is most critical for minimizing VoIP jitter? A) Packet size B) Buffer size on the receiving device C) IP TTL D. MAC address filtering Answer: B Explanation: Properly sized jitter buffers smooth out packet arrival variations, reducing audible distortion. Question 29. In Biamp’s VoIP checklist, the “SIP User Name” typically corresponds to: A) The device’s MAC address B) The extension number or SIP ID assigned by the PBX C) The VLAN tag D) The DHCP lease identifier Answer: B Explanation: SIP User Name is the identifier (often the extension) used for authentication and call routing. Question 30. Which protocol is used by Biamp devices to obtain IP addresses automatically? A) RARP B) DHCP C) ARP
D) BOOTP
Answer: B Explanation: DHCP provides dynamic IP addressing, subnet mask, gateway, and DNS information. Question 31. When a Biamp VoIP endpoint experiences “high latency,” the most likely network cause is: A) Low CPU utilization on the device B) Excessive routing hops or congestion C) Incorrect SIP URI format D) Use of G.711 codec Answer: B Explanation: More hops or congested links increase round‑trip time, leading to latency. Question 32. Which Biamp product can be used to add VoIP capability to an existing analog audio system without replacing the DSP? A) Tesira SVC‑ 2 B) Audia VoIP‑ 2 C) Vocia MS‑ 1 D) Tesira FORTÉ VT Answer: B Explanation: Audia VoIP‑2 adds SIP/VoIP interfaces to an Audia system while keeping the existing DSP configuration. Question 33. In SIP, what does the “Contact” header field convey? A) The IP address of the proxy server B) The address where the user can be reached for future requests
A) One‑way audio B) High CPU usage on the PBX C) Incorrect SIP registration time D) Duplicate MAC addresses Answer: A Explanation: NAT can block RTP streams in one direction, resulting in audio flowing only from one side. Question 37. The “sips:” URI scheme is used when: A) SIP traffic must be encrypted with TLS. B) SIP is carried over UDP. C) The device uses IPv6 only. D) The call is a video session. Answer: A Explanation: “sips:” indicates that the SIP signaling should be protected with TLS encryption. Question 38. Which Biamp software tool is used to design and configure DSP signal flow, including VoIP blocks? A) SageVue B) Audia Designer C) Tesira Designer D) Vocia Configurator Answer: C Explanation: Tesira Designer provides a graphical environment for inserting and routing VoIP blocks. Question 39. In a VoIP network, the term “packet loss” most directly impacts:
A) Call setup time B) Audio quality (e.g., gaps or choppiness) C) IP address allocation D) VLAN tagging Answer: B Explanation: Lost RTP packets result in missing audio samples, causing perceptible gaps. Question 40. Which SIP method is used to cancel a pending INVITE request before the call is answered? A) BYE B) CANCEL C) OPTIONS D) NOTIFY Answer: B Explanation: CANCEL aborts an outstanding INVITE transaction. Question 41. When configuring a Biamp VoIP endpoint for Cisco CallManager, which parameter is typically set to “cisco.com” as the SIP domain? A) Proxy Address B) SIP Domain Name C) Authentication User Name D) SIP User Name Answer: B Explanation: Cisco CallManager often uses “cisco.com” as the SIP domain to route calls correctly.
Question 45. The “SIP Proxy” in a VoIP system primarily performs which role? A) Generates audio tones B) Routes SIP requests to the appropriate destination C) Provides DHCP leases D) Encrypts media streams Answer: B Explanation: The SIP Proxy forwards SIP messages, acting as an intermediary between user agents. Question 46. In Biamp’s VoIP web interface, setting the “Transport” to “TLS” ensures: A) Media encryption only B) SIP signaling encryption C) Automatic QoS tagging D) VLAN isolation Answer: B Explanation: TLS secures SIP signaling, protecting credentials and call setup data. Question 47. During a VoIP call, if the RTP sequence numbers are not monotonic, what is the likely cause? A) DNS failure B) Packet reordering due to network congestion C) Incorrect SIP URI D) Faulty codec selection Answer: B Explanation: Network congestion can cause packets to arrive out of order, breaking the expected sequence.
Question 48. Which Biamp product includes a built‑in Far‑End Echo Reference for AEC? A) Tesira SVC‑ 2 B) Vocia MS‑ 1 C) Audia VoIP‑ 2 D) Tesira FORTÉ VT Answer: A Explanation: The SVC‑2 provides a dedicated Far‑End reference path to improve echo cancellation. Question 49. In a VoIP environment, “SIP trunking” refers to: A) Connecting two PBXs via analog lines B) Using SIP to transport multiple voice channels over a single IP link C) Providing video conferencing over SIP D) Encrypting SIP traffic with IPSec Answer: B Explanation: SIP trunking aggregates many voice calls onto one IP connection, reducing PSTN costs. Question 50. Which of the following is a recommended practice for securing Biamp VoIP devices? A) Disable all QoS settings B) Use default SIP passwords C) Enable TLS for SIP signaling and strong passwords D) Open all firewall ports Answer: C Explanation: TLS and strong authentication protect the device from eavesdropping and unauthorized access.
Explanation: SDP outlines the media session parameters, such as supported codecs and transport ports. Question 54. Which of the following is a typical symptom of a misconfigured DNS server for a Biamp VoIP device? A) Continuous ringing without answer B) Failure to resolve the SIP proxy address, leading to registration errors C) Audio echo on both ends D) Random reboot of the device Answer: B Explanation: If DNS cannot resolve the proxy hostname, the device cannot register, resulting in errors. Question 55. When using Biamp’s Tesira Designer, the “Signal Flow” view primarily helps engineers to: A) Configure VLANs B) Visualize audio routing and processing blocks C) Set SIP passwords D) Monitor network latency Answer: B Explanation: Signal Flow displays how audio signals travel through DSP blocks, including VoIP components. Question 56. Which SIP response code indicates that the request was successful and the call is now established? A) 100 Trying B) 180 Ringing C) 200 OK
D) 403 Forbidden Answer: C Explanation: 200 OK confirms that the INVITE was accepted and the media session can begin. Question 57. In a Biamp VoIP deployment, “QoS Precedence” is used to: A) Assign priority levels to different types of traffic (voice vs. data) B) Determine which codec to use C) Set VLAN IDs automatically D) Enable NAT traversal Answer: A Explanation: Precedence bits in the IP header help routers prioritize voice packets over less‑time‑critical data. Question 58. Which of the following is a primary advantage of using SIP over H.323 for VoIP? A) H.323 supports more codecs B) SIP is text‑based and easier to debug C) H.323 provides built‑in encryption D) SIP requires less bandwidth for media Answer: B Explanation: SIP’s plain‑text messages simplify troubleshooting and integration with web technologies. Question 59. A Biamp VoIP device shows “SIP registration: Failed – 404 Not Found.” The most likely cause is: A) The device’s MAC address is duplicated. B) The proxy address is incorrectly entered.
