SSCA VoIP Technologies & Network Integration 3 — Questions and Answers
Question 1: What is the purpose of a jitter buffer in VoIP?
- To compress audio before transmission
- To store incoming packets temporarily and smooth out arrival time variations (Correct answer)
- To retransmit lost voice packets
- To convert between G.711 and G.729 codecs
Correct answer: To store incoming packets temporarily and smooth out arrival time variations
A jitter buffer holds arriving packets for a brief period to reorder and smooth out delivery before passing audio to the decoder.
Question 2: Which SIP method is used by a UA to de-register from a SIP registrar?
- UNREGISTER
- CANCEL
- REGISTER with Expires: 0 (Correct answer)
- BYE
Correct answer: REGISTER with Expires: 0
A UA sends a REGISTER request with an Expires header value of 0 to remove its binding from the registrar.
Question 3: In SIP, what does the Contact header field indicate?
- The final destination of the SIP message
- The URI where the UA can be reached directly for subsequent requests (Correct answer)
- The address of the SIP proxy
- The email address of the call originator
Correct answer: The URI where the UA can be reached directly for subsequent requests
The Contact header provides a direct URI (typically including IP and port) that can be used to send future requests directly to the UA.
Question 4: Which technology allows a PBX to connect to the PSTN over an IP network instead of traditional T1/E1 circuits?
- H.323 gatekeeper
- SIP trunking (Correct answer)
- MGCP endpoint
- ISDN BRI
Correct answer: SIP trunking
SIP trunking replaces traditional PSTN circuits by carrying voice calls over an IP connection to an ITSP.
Question 5: What is the maximum acceptable one-way mouth-to-ear latency recommended by ITU-T G.114 for voice quality?
- 50 ms
- 100 ms
- 150 ms (Correct answer)
- 300 ms
Correct answer: 150 ms
ITU-T G.114 recommends that one-way latency should not exceed 150 ms for acceptable conversational voice quality.
Question 6: Which SIP header is used to prevent SIP request loops by tracking the proxies a message has traversed?
- Via (Correct answer)
- Route
- Max-Forwards
- Record-Route
Correct answer: Via
The Via header is added by each proxy in the path, allowing loop detection and providing the return path for responses.
Question 7: A G.729 codec call uses approximately how much bandwidth including IP/UDP/RTP headers?
- 8 kbps
- 32 kbps
- 24 kbps (Correct answer)
- 64 kbps
Correct answer: 24 kbps
G.729 produces 8 kbps of audio payload, but with IP/UDP/RTP overhead the total bandwidth per call is approximately 24 kbps.
SSCA VoIP Technologies & Network Integration 3
What is the purpose of a jitter buffer in VoIP?