Voice over IP (VoIP) Flashcards
6 cards from real Telecommunications practice questions. Tap to flip, then mark Knew It or Still Learning — missed cards come back until you master them.
Read the first 6 Voice over IP (VoIP) flashcards as text
What protocol is most commonly used to initiate and terminate VoIP calls?
Answer: SIP
SIP (Session Initiation Protocol) is the dominant signaling protocol for establishing, modifying, and terminating VoIP sessions.
What protocol carries the actual voice media in a VoIP call?
Answer: RTP
RTP (Real-time Transport Protocol) carries the encoded audio and video media streams during an active VoIP call.
What is jitter in VoIP communications?
Answer: Variation in packet arrival times
Jitter is the variation in latency between consecutive voice packets, which causes audio quality issues if not buffered.
Which codec is known for narrowband voice quality at 64 kbps?
Answer: G.711
G.711 is the standard PCM codec used in traditional telephony and VoIP, operating at 64 kbps with toll-quality audio.
What is the recommended maximum one-way latency for acceptable VoIP call quality?
Answer: 150 ms
ITU-T G.114 recommends a maximum one-way delay of 150 ms for acceptable conversational quality in VoIP.
What does a SIP proxy server do?
Answer: Route SIP requests between endpoints
A SIP proxy server routes SIP signaling messages between clients, enabling call setup across different network segments.