0% found this document useful (0 votes)
6 views8 pages

Section 2

The document covers fundamental concepts of IP Telephony, including definitions, differences between VOIP and traditional telephony, and key terms like codecs, jitter, and endpoints. It explains technical aspects such as packetization, Quality of Service, and NAT traversal, as well as operational elements like dial plans and call admission control. Additionally, it addresses the importance of protocols like SRTP and T.38 for secure and reliable communication.
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
0% found this document useful (0 votes)
6 views8 pages

Section 2

The document covers fundamental concepts of IP Telephony, including definitions, differences between VOIP and traditional telephony, and key terms like codecs, jitter, and endpoints. It explains technical aspects such as packetization, Quality of Service, and NAT traversal, as well as operational elements like dial plans and call admission control. Additionally, it addresses the importance of protocols like SRTP and T.38 for secure and reliable communication.
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd

2.

IP Telephony Fundamentals (Questions 36–65)

36. What is IP Telephony?

Answer:
IP Telephony is the use of Internet Protocol networks to transmit
voice, fax, and related services instead of the traditional public
switched telephone network (PSTN).

37. How does VOIP differ from traditional telephony?

Answer:
VOIP converts voice into digital packets and transmits them over
IP networks, whereas traditional telephony uses analog signals
over dedicated circuits.

38. Name three common VOIP codecs and their


bitrates.

Answer:

 G.711 (64kbps)

 G.729 (8kbps)

 G.722 (typically 48–64kbps)

39. What is jitter and why does it matter in VOIP?


Answer:
Jitter is the variation in packet arrival times. High jitter can cause
voice distortion or dropouts in VOIP calls.

40. Define an endpoint in IP Telephony.

Answer:
An endpoint is any device capable of sending or receiving VOIP,
such as an IP phone, softphone, or conference unit.

41. Explain the term “dial plan.”

Answer:
A dial plan determines how call routing is managed by defining
which digit sequences or patterns correspond to local, national, or
international calls.

42. What is echo in VOIP and how is it mitigated?

Answer:
Echo is the return of a caller's voice. It's often controlled by echo
cancellation algorithms in endpoints and gateways.

43. What is DTMF and why is it important?

Answer:
Dual Tone Multi-Frequency—used for dialing and telephony
signaling. Reliable DTMF transport is crucial for IVR and other
interactive systems.

44. What is packetization in VOIP?

Answer:
Dividing voice signals into packets for transport over IP networks.

45. How is Quality of Service (QoS) typically


implemented for VOIP?

Answer:
By tagging VOIP packets at Layer 2/3 (e.g., using DSCP or
802.1p) to prioritize them over other types of data.

46. What are voice VLANs?

Answer:
Separate logical networks to isolate VOIP traffic from regular data,
improving QoS and security.

47. What is the typical port range for RTP?

Answer:
UDP ports 1024–65535 (commonly 16384–32767 for Avaya).

48. How does POE benefit IP Telephony?


Answer:
Power over Ethernet delivers electrical power along the same
cables as data, allowing phones to operate without separate power
adapters.

49. Explain the SIP REGISTER process.

Answer:
An endpoint sends a REGISTER message to the SIP registrar to
announce its location (IP address), updates, and validity.

50. What is NAT traversal in VOIP?

Answer:
Methods to enable devices behind Network Address Translation
(NAT) to communicate with external endpoints, often using
STUN/TURN/ICE.

51. Why is call admission control important?

Answer:
It limits the number of concurrent calls to prevent congestion and
ensure call quality.

52. What is SRTP?


Answer:
Secure Real-time Transport Protocol—encrypts and authenticates
voice streams.

53. What is “Early Media” in SIP?

Answer:
Audio (such as ringback tones or messages) sent before a call is
fully established, often via 183 Session Progress responses.

54. Explain codec negotiation.

Answer:
Endpoints agree on the audio format to use (e.g., G.711 or G.729)
during call setup using SDP in SIP messages.

55. What is “One-way audio” and a typical cause?

Answer:
Audio flows in only one direction. Common causes: firewalls or
NAT blocking RTP streams.

56. What is the benefit of using DHCP for IP phones?

Answer:
Automatic IP assignment and provisioning of configuration
(TFTP, HTTP) server addresses.
57. What is a softphone?

Answer:
Software-based phone application that enables VOIP functionality
on a computer or mobile device.

58. Explain the purpose of a TFTP/HTTP server in


VOIP deployments.

Answer:
Used for supplying firmware, configuration files, and settings to IP
telephony endpoints.

59. What is RTP payload type?

Answer:
A numeric code identifying the codec used in the RTP stream (e.g.,
0 for G.711-uLaw).

60. Why do VOIP endpoints re-register periodically?

Answer:
To maintain their active status and update location with the SIP
registrar or H.323 gatekeeper.

61. How do you manage clock synchronization for


VOIP devices?
Answer:
Via NTP (Network Time Protocol) servers to ensure accurate
timestamps and event logging.

62. What is a “hunt group”?

Answer:
A group of endpoints or agents configured to receive incoming
calls in a pre-defined pattern.

63. How do call parks and pickup groups improve


productivity?

Answer:
Allow calls to be temporarily held (“parked”) or “picked up” by
any group member, aiding team collaboration.

64. What happens when phones are misaligned with


their VLAN?

Answer:
Phones may not reach their provisioning servers, fail to register, or
obtain wrong IP addresses, resulting in outages.

65. How can fax be supported over VOIP?


Answer:
Using T.38 protocol for reliable fax transmission, or by falling
back to G.711 pass-through for legacy devices.

You might also like