SSCA SIP Troubleshooting & Diagnostic Tools

Question 1

What is the primary function of SIP OPTIONS method?

Accepted Answer

To query capabilities of a server or endpoint

Answer

The SIP OPTIONS method is used to query a server or endpoint about its capabilities and supported features without establishing a call. It allows a user agent to discover what methods, codecs, and extensions a remote party supports. This is useful for pre-call negotiation and ensuring compatibility before attempting to set up a session.

Question 2

Which tool is commonly used to capture and analyze SIP traffic?

Accepted Answer

Wireshark

Answer

Wireshark is a powerful and widely used network protocol analyzer that can capture and analyze network traffic, including SIP packets. It allows administrators to inspect the contents of individual SIP messages, trace call flows, and identify issues like malformed packets or signaling errors. Its detailed packet inspection capabilities make it invaluable for troubleshooting VoIP problems.

Question 3

What does SIP response code 486 indicate?

Accepted Answer

Busy Here

Answer

SIP response code 486 indicates 'Busy Here,' meaning the called party is currently busy and cannot accept another call. This response is sent by the user agent or server to inform the caller that the recipient is unavailable. It's a standard way to communicate that the intended recipient is engaged in another conversation.

Question 4

Why is SIP ladder diagram useful in diagnostics?

Accepted Answer

They help visualize SIP message flow

Answer

A SIP ladder diagram is a graphical representation that illustrates the sequence and flow of SIP messages between different entities during a call setup or teardown. This visual aid is extremely useful in diagnostics as it helps administrators quickly understand the order of events, identify where a call failed, or pinpoint delays in the signaling process. It provides a clear timeline of interactions.

Question 5

What is a common reason for one-way audio in SIP calls?

Accepted Answer

Improper NAT/firewall configuration

Answer

One-way audio in SIP calls is frequently caused by improper Network Address Translation (NAT) or firewall configurations. NAT devices and firewalls can block or incorrectly translate the Real-time Transport Protocol (RTP) media streams, preventing one of the parties from receiving audio. This typically happens when the firewall doesn't correctly open the necessary ports for the RTP traffic.

(SSCA) Sip School Certified Associate Practice Test

(SSCA) Sip School Certified Associate Practice Test

SSCA SIP Troubleshooting & Diagnostic Tools