352-001 · Question #688
A retail company has 250 branches connected to the core via MPLS Layer 3 VPN. The company is migrating from traditional telephony services to a VoIP solution. Which two primary concerns when…
The correct answer is A. jitter B. available bandwidth. VoIP over a WAN requires careful management of jitter and available bandwidth to maintain acceptable voice quality across the MPLS network.
Question
A retail company has 250 branches connected to the core via MPLS Layer 3 VPN. The company is migrating from traditional telephony services to a VoIP solution. Which two primary concerns when implementing a VoIP solution are true? (Choose two.)
Options
- Ajitter
- Bavailable bandwidth
- Ccall routing design
- Dcall routing design
- EMTU
- FSRST
How the community answered
(45 responses)- A80% (36)
- C11% (5)
- D4% (2)
- E2% (1)
- F2% (1)
Why each option
VoIP over a WAN requires careful management of jitter and available bandwidth to maintain acceptable voice quality across the MPLS network.
Jitter - the variation in packet inter-arrival delay - directly degrades voice quality by causing choppy or distorted audio at the receiving endpoint. It must be kept below 30ms for acceptable VoIP quality, requiring QoS mechanisms like priority queuing and LLQ on the MPLS network.
Sufficient available bandwidth must be reserved for voice traffic to prevent packet loss and queuing delays that degrade call quality. Without proper bandwidth allocation using mechanisms like CBWFQ or LLQ, voice packets compete with data traffic and experience unacceptable delay.
Call routing design is an application-layer telephony concern related to dial plans and call admission control, not a primary network transport quality parameter for VoIP.
Call routing design is a duplicate of choice C and is an application-layer telephony concern, not a primary network transport quality parameter for VoIP.
MTU size can cause fragmentation overhead but is a secondary, easily addressed concern through interface configuration and does not continuously impact call quality the way jitter and bandwidth do.
SRST (Survivable Remote Site Telephony) is a Cisco failover feature providing local call processing when WAN connectivity is lost, not a primary network transport quality concern for active VoIP traffic.
Concept tested: VoIP QoS primary network transport requirements
Source: https://www.cisco.com/c/en/us/support/docs/voice/voice-quality/14081-voip-qos.html
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