hardMultiple Choice
300-410 Practice Question: Is troubleshooting NAT for a VoIP phone that uses…
A network engineer is troubleshooting NAT for a VoIP phone that uses SIP. The phone is at 192.168.2.10, and the router performs PAT to the outside interface 198.51.100.1. The phone can register with the SIP server, but calls fail after 30 seconds. The engineer notices that the SIP signaling includes the phone's private IP in the SDP body. What is the most likely cause?
⚠ Common exam trap
Cisco often tests the distinction between control plane (SIP signaling) and data plane (RTP media) failures, and the trap here is that candidates assume registration success means NAT is working correctly, overlooking the need for ALG to translate embedded IPs in the SDP body.
Answer choices
Why each option matters
Answer the question above first, then reveal the full breakdown to understand why each option is right or wrong.
Correct answer & explanation
✓
The router's SIP ALG is disabled, so the private IP in the SDP is not translated.
The SIP signaling includes the phone's private IP in the SDP body, which is used for media negotiation. When the router's SIP Application Layer Gateway (ALG) is disabled, it does not inspect and translate the private IP addresses embedded in the SDP payload. As a result, the SIP server sends media (RTP) to the private IP 192.168.2.10, which is unreachable from outside, causing the call to fail after the initial registration and signaling succeed.
Answer analysis
Option-by-option breakdown
For each option: why learners choose it and why it is or isn't the right answer here.
- ✗
The PAT port range is exhausted.
Why it's wrong here
Port exhaustion would prevent new PAT translations, breaking registration and all concurrent sessions, not just calls after 30 seconds. It is tempting because PAT limits genuinely cause intermittent VoIP failures, and exhaustion would be the cause if many endpoints shared the single outside address.
- ✓
The router's SIP ALG is disabled, so the private IP in the SDP is not translated.
Why this is correct
SIP embeds the phone's private address inside the SDP body of the signalling payload. PAT rewrites only packet headers, not payload contents, so without the SIP ALG inspecting and translating that embedded address, the remote party cannot return media.
- ✗
The phone's default gateway is misconfigured.
Why it's wrong here
A wrong default gateway would prevent the phone reaching the SIP server at all, so registration would fail, not succeed then drop. The gateway is tempting because it is the first thing checked when endpoints cannot communicate, and it would be the cause if the phone had no network reachability whatsoever.
- ✗
The outside interface has a firewall blocking UDP ports.
Why it's wrong here
A firewall blocking UDP would stop SIP registration and RTP entirely, yet registration succeeds here. Firewall rules are tempting because they commonly break VoIP media, and blocking UDP would be the cause if signalling never completed or no audio flowed from the outset.
Visual reference
Quick reference
OSI Model Reference
| Layer | Name | PDU | Key Protocols / Devices |
|---|---|---|---|
| 7 | Application | Data | HTTP, HTTPS, DNS, SMTP, FTP, SSH |
| 6 | Presentation | Data | TLS / SSL, JPEG, ASCII encoding |
| 5 | Session | Data | NetBIOS, RPC, SIP |
| 4 | Transport | Segment / Datagram | TCP, UDP |
| 3 | Network | Packet | IP, ICMP, OSPF — Routers |
| 2 | Data Link | Frame | Ethernet, Wi-Fi, PPP — Switches, Bridges |
| 1 | Physical | Bits | Cables, NICs, Hubs, Repeaters |
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Written by Johnson Ajibi, MSc IT Security
Senior Network & Security Engineer · founder of Courseiva
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