Phone Service Business VoIP Technology & Protocols 2 — Questions and Answers
Question 1: What is latency in a VoIP call, and what is considered an acceptable one-way delay?
- Echo caused by hardware; under 10ms
- The delay between speaking and hearing; under 150ms is ideal per ITU-T (Correct answer)
- Packet loss percentage; under 5%
- Background noise level; under 30dB
Correct answer: The delay between speaking and hearing; under 150ms is ideal per ITU-T
Latency is the one-way delay in a VoIP call; the ITU-T recommends keeping it under 150ms for acceptable call quality.
Question 2: What does QoS (Quality of Service) do in a network supporting VoIP?
- Encrypts all voice packets
- Prioritizes voice traffic over other data to reduce latency and jitter (Correct answer)
- Increases internet bandwidth automatically
- Converts analog signals to digital
Correct answer: Prioritizes voice traffic over other data to reduce latency and jitter
QoS settings on a network router prioritize voice packets to ensure consistent, high-quality VoIP calls even during high data usage.
Question 3: What is packet loss in VoIP and what percentage is generally considered problematic?
- Calls being dropped; any loss above 0%
- Lost audio data packets; above 1-3% noticeably degrades call quality (Correct answer)
- Encryption failures; above 5%
- Codec mismatches; above 10%
Correct answer: Lost audio data packets; above 1-3% noticeably degrades call quality
Packet loss above 1-3% in VoIP causes audible gaps or clipping because lost audio packets cannot be retransmitted in real time.
Question 4: Which port does SIP most commonly use for signaling traffic?
- Port 80
- Port 443
- Port 5060 (Correct answer)
- Port 25
Correct answer: Port 5060
SIP signaling traffic typically uses UDP or TCP port 5060, and port 5061 for TLS-encrypted SIP.
Question 5: What is SRTP used for in a VoIP deployment?
- Compressing audio files
- Encrypting voice media streams to prevent eavesdropping (Correct answer)
- Routing calls between SIP servers
- Managing bandwidth allocation
Correct answer: Encrypting voice media streams to prevent eavesdropping
SRTP (Secure Real-time Transport Protocol) encrypts VoIP audio streams to protect them from interception.
Question 6: What is a codec in the context of VoIP?
- A type of VoIP phone handset
- Software that compresses and decompresses audio for transmission (Correct answer)
- A network routing device
- A billing management system
Correct answer: Software that compresses and decompresses audio for transmission
A codec (coder-decoder) encodes audio signals for transmission and decodes them at the receiving end during a VoIP call.
Phone Service Business VoIP Technology & Protocols 2
What is latency in a VoIP call, and what is considered an acceptable one-way delay?