CCNA Voice Interview Questions: 25 Voice and VoIP Answers (Exam Retired in 2020)

Key takeaways

• The CCNA Voice certification no longer exists. Voice topics now sit inside CCNA 200-301 (QoS, IP services, network fundamentals) and in Cisco's professional-level tracks.
• Interviews for voice and collaboration roles still test VoIP fundamentals, so the questions below remain useful.
• Know the three numbers behind good call quality: one-way delay, jitter and packet loss.
• Be ready to explain a voice VLAN, a DSCP marking and a dial peer in plain words, and to draw the packet path of a call.
• Check Cisco's current certification pages for the up-to-date collaboration track instead of preparing for a retired exam.

What happened to CCNA Voice?

Cisco retired CCNA Voice, together with CCNA Security, Wireless, Routing and Switching and Data Center, in February 2020 and folded the CCNA family into a single exam, CCNA 200-301. Voice content did not disappear. Core topics such as QoS, DHCP, VLANs and IP services are in CCNA 200-301, and deeper voice and collaboration material moved to Cisco's professional-level tracks. Cisco restructures its tracks from time to time, so confirm the current names on the Cisco certifications page.

If you are preparing for an interview rather than an exam, none of this changes the questions. A hiring manager for a voice engineer or network engineer role will still ask how a call is set up and how you keep it clear under load.

Voice fundamentals

1. What is VoIP and how does it work?

VoIP (Voice over IP) carries phone calls as IP packets. The phone or gateway digitises speech, compresses it with a codec, puts it in RTP packets and sends it across the network. The receiver reassembles the packets and plays the sound. Signalling, such as SIP, sets up and ends the call; RTP carries the audio.

2. What is the difference between signalling and media?

Signalling controls the call: registering, ringing, answering, hanging up. Media is the actual voice stream. They use different protocols and often different paths. A phone can register fine (signalling works) and still have one-way audio (media blocked), which is a classic firewall or NAT symptom.

3. How does SIP work?

SIP is a text-based signalling protocol. A caller sends an INVITE; the other side replies with provisional responses (100 Trying, 180 Ringing) and then 200 OK; the caller confirms with ACK; and BYE ends the call. SDP, carried inside SIP messages, describes the codec and the IP address and port for the RTP stream.

4. SIP, SCCP (Skinny) and H.323: what is the difference?

SIP is an open standard (IETF) and the most widely used. SCCP is Cisco's older proprietary protocol for its IP phones; many Cisco phones support SIP as well. H.323 is an older ITU standard suite, still found on some legacy gateways. Interviewers want you to know SIP best, and to recognise the other two.

5. What does RTP do?

Real-time Transport Protocol carries voice samples over UDP. It adds a sequence number and timestamp so the receiver can reorder packets and measure jitter. RTCP, its companion, reports loss and delay statistics. Voice uses UDP because a late packet is worthless: retransmitting it, as TCP would, only adds delay.

6. What are codecs and which ones should I know?

A codec converts voice to digital data and back. G.711 gives toll-quality audio at 64 kbps of payload and is common on LANs. G.729 compresses to 8 kbps of payload, saving WAN bandwidth at some cost in quality and with licensing history. At the IP layer, with 20 ms packetisation, G.711 needs about 80 kbps per call and G.729 about 24 kbps. Layer 2 headers add more. Wideband codecs such as G.722 and Opus give better audio.

Quality of service

7. Which three network problems damage voice?

Delay (latency), jitter (variation in delay) and packet loss. Cisco's commonly quoted design targets for voice are one-way delay of 150 ms or less, jitter of 30 ms or less and loss of 1 percent or less. Treat these as design guidelines to quote with their source, not as laws.

8. What is QoS and why does voice need it?

QoS (quality of service) is a set of tools that classify traffic and treat important packets better when links are busy. Voice is small, steady and sensitive to delay, so it must not queue behind a large file download.

9. What is DSCP and what value does voice use?

DSCP is a 6-bit marking in the IP header that tells routers how to treat the packet. Voice media is normally marked EF (Expedited Forwarding, decimal 46). Call signalling is commonly marked CS3 (decimal 24). Marking must be done at the edge, and trusted only from devices you control.

10. What is LLQ?

Low Latency Queueing adds a strict-priority queue to class-based weighted fair queueing (CBWFQ). Voice goes in the priority queue and is sent first, but it is policed to a set rate so it cannot starve other traffic.

11. What is the difference between policing and shaping?

Policing drops or re-marks traffic above a rate. Shaping buffers excess traffic and sends it later, which smooths bursts but adds delay. Voice queues are policed; bulk data is often shaped.

12. What is a jitter buffer?

The receiving phone holds incoming packets briefly and plays them out at an even rate, hiding small variations in arrival time. A bigger buffer absorbs more jitter but adds delay.

Switching and phone connectivity

13. Why use a voice VLAN?

A voice VLAN separates phone traffic from PC traffic on the same switch port. This lets you apply QoS and security policy to voice, keeps broadcast traffic away from phones, and lets a PC plug in behind the phone. A minimal port looks like this:

interface GigabitEthernet0/5
 switchport mode access
 switchport access vlan 10
 switchport voice vlan 110
 spanning-tree portfast

VLAN 10 carries the PC's data; VLAN 110 carries the phone. The switch tells the phone its voice VLAN using CDP or LLDP-MED.

14. How does a phone get its IP address and configuration?

The phone boots, learns its voice VLAN from the switch, requests an address by DHCP and receives the address of its configuration server. On Cisco networks this is usually DHCP option 150 (a list of TFTP server addresses). Option 66 is the standard, single-name alternative. The phone then downloads its configuration and registers with call control.

15. How are phones powered?

Through Power over Ethernet. 802.3af supplies up to 15.4 W per port, 802.3at (PoE+) up to 30 W and 802.3bt more. Plan the switch's total PoE budget, not just the per-port limit.

Call control and components

16. What are the main parts of a Cisco Unified Communications system?

Cisco Unified Communications Manager (CUCM) handles call control. IP phones and soft clients are the endpoints. Cisco Unity Connection provides voicemail. Cisco Unified Contact Center Express (UCCX) handles call-centre features. The IM and Presence Service (earlier called Unified Presence) handles messaging and presence. Voice gateways connect to the public phone network.

17. What does CUCM do?

CUCM is the call-processing server. Phones register with it, and it handles call routing, dial plan, call admission, features such as transfer and conference, and configuration of devices. A cluster has a publisher and subscribers for redundancy.

18. What is a voice gateway?

A gateway links the IP voice network to other telephony networks such as the PSTN. It converts between IP voice and traditional signalling. FXS ports connect analogue phones and fax machines, FXO ports connect to an analogue phone line, and digital ports connect to ISDN PRI or T1/E1 circuits.

19. What is a dial peer?

On a Cisco IOS gateway, a dial peer matches a dialled number pattern and tells the router where to send the call. A VoIP dial peer points to an IP address; a POTS dial peer points to a physical voice port.

dial-peer voice 100 voip
 destination-pattern 2...
 session protocol sipv2
 session target ipv4:10.1.1.10
 codec g711ulaw

This sends any four-digit number starting with 2 to 10.1.1.10 over SIP using G.711.

20. What is SRST?

Survivable Remote Site Telephony lets a branch router keep basic calling alive when the WAN or CUCM is unreachable. Phones register with the local router, so staff can still make local and PSTN calls. It is a fallback with fewer features.

21. What is a trunk in voice networks?

In voice, a trunk is a connection that carries many calls between systems, for example between CUCM and a gateway, or between two call-control clusters. SIP trunks are common. Do not confuse them with 802.1Q trunk links between switches.

Troubleshooting questions

22. Calls connect but you hear one-way audio. Where do you look?

Suspect the media path. Check NAT and firewall rules for the RTP port range, whether SDP carries a private address that the far side cannot reach, and routing in both directions. Use packet captures to see whether RTP flows both ways.

23. Voice is choppy during working hours only. What next?

That points to congestion. Check interface utilisation and drops, confirm DSCP markings survive across the path, check LLQ counters, and look for a bulk transfer or backup starting at that time.

24. A phone shows "Registering" and never completes. What do you check?

Work from the bottom up: link and PoE, voice VLAN assignment, DHCP lease and option 150, reachability of the TFTP and CUCM addresses, then the device's configuration and MAC address in CUCM.

25. What tools do you use to troubleshoot voice?

Packet captures in Wireshark (which can decode SIP and RTP and show call flows and RTP stream statistics), switch and router show commands, and the call-control server's own logs and reports. The Wireshark documentation covers VoIP analysis.

How to prepare for a voice or network interview now

• Be able to draw a call from phone boot to a connected conversation, naming each protocol.
• Know the QoS chain end to end: classify and mark at the edge, queue with LLQ on congested links.
• Practise explaining things aloud in two sentences. Interviewers test clarity as much as recall.
• Set up a small lab if you can, using a free softphone and an open-source or trial call server, and capture a real SIP call.

Next steps

If you are early in your networking career, start with the current CCNA: our CCNA 200-301 course covers QoS and the IP services these questions depend on. Practise more with our CCNA interview questions on QoS and CCNA exam preparation questions.

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