Telecommunications Voice over IP (VoIP) 1 — Questions and Answers
Question 1: What protocol is most commonly used to initiate and terminate VoIP calls?
- RTP
- SIP (Correct answer)
- H.323
- MGCP
Correct answer: SIP
SIP (Session Initiation Protocol) is the dominant signaling protocol for establishing, modifying, and terminating VoIP sessions.
Question 2: What protocol carries the actual voice media in a VoIP call?
- SIP
- RTP (Correct answer)
- TCP
- SCTP
Correct answer: RTP
RTP (Real-time Transport Protocol) carries the encoded audio and video media streams during an active VoIP call.
Question 3: What is jitter in VoIP communications?
- Packet loss percentage
- Variation in packet arrival times (Correct answer)
- Total one-way delay
- Echo on the voice line
Correct answer: Variation in packet arrival times
Jitter is the variation in latency between consecutive voice packets, which causes audio quality issues if not buffered.
Question 4: Which codec is known for narrowband voice quality at 64 kbps?
- G.711 (Correct answer)
- G.729
- G.722
- Opus
Correct answer: G.711
G.711 is the standard PCM codec used in traditional telephony and VoIP, operating at 64 kbps with toll-quality audio.
Question 5: What is the recommended maximum one-way latency for acceptable VoIP call quality?
- 50 ms
- 150 ms (Correct answer)
- 300 ms
- 500 ms
Correct answer: 150 ms
ITU-T G.114 recommends a maximum one-way delay of 150 ms for acceptable conversational quality in VoIP.
Question 6: What does a SIP proxy server do?
- Convert analog to digital signals
- Route SIP requests between endpoints (Correct answer)
- Encode voice into packets
- Provide DHCP for VoIP phones
Correct answer: Route SIP requests between endpoints
A SIP proxy server routes SIP signaling messages between clients, enabling call setup across different network segments.
Telecommunications Voice over IP (VoIP) 1
What protocol is most commonly used to initiate and terminate VoIP calls?