SSCA VoIP Architecture 4 — Questions and Answers
Question 1: What is the G.711 codec's bandwidth requirement per call channel in VoIP?
- 8 kbps
- 64 kbps (Correct answer)
- 128 kbps
- 32 kbps
Correct answer: 64 kbps
G.711 (PCMU/PCMA) samples audio at 8000 Hz with 8-bit samples, producing a 64 kbps uncompressed audio stream.
Question 2: Which SIP method is used by a UA to deregister from a SIP registrar?
- BYE with Contact header
- REGISTER with Expires: 0 (Correct answer)
- CANCEL with Contact header
- UNREGISTER method
Correct answer: REGISTER with Expires: 0
Sending a REGISTER request with an Expires header value of 0 removes the binding from the registrar, effectively deregistering the UA.
Question 3: What VoIP architecture technique allows a provider to offer multiple tenants on shared infrastructure with isolated configurations?
- Load balancing
- Multi-tenancy (Correct answer)
- SIP forking
- Media anchoring
Correct answer: Multi-tenancy
Multi-tenancy allows a single VoIP platform to serve multiple customers with logically isolated numbering plans, routing, and features.
Question 4: In VoIP, what is 'codec negotiation' and when does it occur?
- The process of selecting audio encoding during SDP offer/answer exchange in call setup (Correct answer)
- Changing the codec mid-call to adapt to bandwidth changes
- The ISP selecting the codec for the customer
- Compressing SIP headers during registration
Correct answer: The process of selecting audio encoding during SDP offer/answer exchange in call setup
Codec negotiation happens during the SDP offer/answer exchange (in INVITE/200 OK) where endpoints agree on a common audio encoding format.
Question 5: What is the primary advantage of using G.729 over G.711 in a bandwidth-constrained VoIP deployment?
- G.729 provides higher audio quality
- G.729 uses approximately 8 kbps versus G.711's 64 kbps, saving bandwidth (Correct answer)
- G.729 eliminates the need for jitter buffers
- G.729 supports HD voice up to 16 kHz
Correct answer: G.729 uses approximately 8 kbps versus G.711's 64 kbps, saving bandwidth
G.729 uses CS-ACELP compression to achieve approximately 8 kbps per call versus G.711's 64 kbps, an 8:1 bandwidth reduction.
Question 6: What does 'SIP trunking' replace in the traditional telephony architecture?
- Internal PBX extensions
- Analog or ISDN PRI lines connecting the PBX to the PSTN (Correct answer)
- Voicemail servers
- Conference bridges
Correct answer: Analog or ISDN PRI lines connecting the PBX to the PSTN
SIP trunking replaces traditional analog or ISDN PRI circuits by connecting an IP PBX to the PSTN via a SIP provider over an IP network.
Question 7: In VoIP architecture, what problem does ICE (Interactive Connectivity Establishment) solve?
- Encrypting RTP streams end-to-end
- Finding the optimal media path between two endpoints across various NAT/firewall topologies (Correct answer)
- Compressing SIP signaling traffic
- Managing SIP dialog state on proxies
Correct answer: Finding the optimal media path between two endpoints across various NAT/firewall topologies
ICE combines STUN and TURN to systematically discover and select the best connectivity path for media between endpoints behind NATs and firewalls.
SSCA VoIP Architecture 4
What is the G.711 codec's bandwidth requirement per call channel in VoIP?