SSCA VoIP Technologies & Network Integration 2 — Questions and Answers
Question 1: Which codec is most commonly used in G.711 VoIP calls and what is its bitrate?
- G.711 uses ADPCM at 32 kbps
- G.711 uses PCM at 64 kbps (Correct answer)
- G.711 uses CELP at 8 kbps
- G.711 uses OPUS at 128 kbps
Correct answer: G.711 uses PCM at 64 kbps
G.711 uses Pulse Code Modulation (PCM) and operates at 64 kbps, providing toll-quality audio.
Question 2: What is the primary function of a Session Border Controller (SBC) in a VoIP network?
- To assign IP addresses to VoIP phones
- To control signaling and media at the border between networks (Correct answer)
- To convert analog signals to digital
- To manage DNS resolution for SIP URIs
Correct answer: To control signaling and media at the border between networks
An SBC sits at the network edge to protect, control, and monitor SIP signaling and RTP media flows between networks.
Question 3: In VoIP, what does jitter refer to?
- The total one-way delay of a voice packet
- Variation in packet arrival times (Correct answer)
- Packet loss percentage over a call
- The difference in codec bitrates between endpoints
Correct answer: Variation in packet arrival times
Jitter is the variation in the delay of received packets, which can cause choppy audio if not compensated by a jitter buffer.
Question 4: Which transport protocol does RTP typically run over in VoIP implementations?
- TCP
- SCTP
- UDP (Correct answer)
- IPSec
Correct answer: UDP
RTP runs over UDP because the low overhead and real-time delivery of UDP are more suitable for voice than TCP's reliability mechanisms.
Question 5: What is the role of RTCP in a VoIP call?
- To encrypt RTP media streams
- To carry voice payload between endpoints
- To provide quality statistics and synchronization for RTP sessions (Correct answer)
- To negotiate codec selection before the call starts
Correct answer: To provide quality statistics and synchronization for RTP sessions
RTCP sends periodic control packets alongside RTP to report statistics like packet loss, jitter, and round-trip time.
Question 6: A VoIP call is experiencing one-way audio. Which of the following is the MOST likely cause?
- Mismatched codecs between endpoints
- Asymmetric NAT preventing RTP from reaching one endpoint (Correct answer)
- An incorrect SIP INVITE method
- A missing DNS SRV record
Correct answer: Asymmetric NAT preventing RTP from reaching one endpoint
Asymmetric NAT commonly causes one-way audio because RTP packets from one side cannot reach the NATed endpoint behind the firewall.
Question 7: Which QoS marking is most commonly used to prioritize VoIP traffic on a network?
- DSCP EF (Expedited Forwarding) (Correct answer)
- DSCP AF11
- DSCP CS1
- DSCP BE (Best Effort)
Correct answer: DSCP EF (Expedited Forwarding)
DSCP EF (decimal 46) is the standard marking for real-time voice traffic to receive low-latency, low-loss forwarding.
SSCA VoIP Technologies & Network Integration 2
Which codec is most commonly used in G.711 VoIP calls and what is its bitrate?