SSCA RTP & Media Transport 2 — Questions and Answers
Question 1: What does RTCP stand for?
- Real-Time Control Protocol
- RTP Control Protocol (Correct answer)
- Reliable Transport Control Protocol
- Remote Transfer Control Protocol
Correct answer: RTP Control Protocol
RTCP stands for RTP Control Protocol and provides out-of-band statistics and control information for an RTP flow, defined alongside RTP in RFC 3550.
Question 2: What percentage of total session bandwidth is typically allocated to RTCP per RFC 3550?
- 1%
- 5% (Correct answer)
- 10%
- 20%
Correct answer: 5%
RFC 3550 recommends that RTCP consume no more than 5% of the total session bandwidth, keeping overhead minimal relative to the media stream.
Question 3: Which RTCP packet type provides statistics from an active media sender?
- RR (Receiver Report)
- SDES (Source Description)
- SR (Sender Report) (Correct answer)
- BYE
Correct answer: SR (Sender Report)
The SR (Sender Report) packet contains an NTP timestamp, RTP timestamp, and packet/byte counts sent, and may also include reception report blocks.
Question 4: Which quality metric included in an RTCP Receiver Report measures variability in packet arrival times?
- Codec bitrate
- Round-trip time
- Interarrival jitter (Correct answer)
- Encryption overhead
Correct answer: Interarrival jitter
Interarrival jitter, expressed in timestamp units, measures the statistical variance in RTP packet arrival spacing and is a key indicator of network consistency.
Question 5: How is 'jitter' defined in the context of VoIP media transport?
- The total number of packets lost during a session
- Variation in the delay between successive RTP packet arrivals (Correct answer)
- The difference between original and decoded audio quality
- The one-way propagation delay from sender to receiver
Correct answer: Variation in the delay between successive RTP packet arrivals
Jitter is the variation in packet interarrival delay; high jitter causes choppy audio because packets do not arrive at the consistent intervals needed for smooth playback.
Question 6: What is the primary purpose of a jitter buffer in a VoIP endpoint?
- To compress audio packets before transmission
- To encrypt RTP streams at the endpoint
- To hold incoming packets briefly so late arrivals can be played in correct order (Correct answer)
- To retransmit lost RTP packets automatically
Correct answer: To hold incoming packets briefly so late arrivals can be played in correct order
A jitter buffer absorbs network jitter by introducing a small playback delay, allowing late-arriving packets to be inserted in sequence before the audio is rendered.
Question 7: Which RTCP packet type signals that a participant is leaving an RTP session?
- SR (Sender Report)
- RR (Receiver Report)
- SDES (Source Description)
- BYE (Correct answer)
Correct answer: BYE
The RTCP BYE packet indicates that one or more sources are no longer active in the session and may include an optional reason string.
