SSCA VoIP Technologies & Network Integration 4 — Questions and Answers
Question 1: What is the function of STUN (Session Traversal Utilities for NAT) in VoIP?
- To encrypt SIP signaling between UACs and UASs
- To allow a UA behind NAT to discover its public IP address and port (Correct answer)
- To transcode audio between incompatible codecs
- To register a UA with a SIP proxy server
Correct answer: To allow a UA behind NAT to discover its public IP address and port
STUN enables a UA behind NAT to learn its externally visible IP and port so it can include correct contact information in SIP and SDP.
Question 2: Which SIP response code indicates that a request has been forwarded to another URI and the client should retry there?
- 301 Moved Permanently
- 302 Moved Temporarily (Correct answer)
- 305 Use Proxy
- 380 Alternative Service
Correct answer: 302 Moved Temporarily
A 302 Moved Temporarily response instructs the UAC to retry the request at the Contact URI provided in the response.
Question 3: In SDP, which field specifies the media type, port, transport protocol, and codec payload types for a media stream?
- c= (connection)
- o= (origin)
- m= (media) (Correct answer)
- a= (attribute)
Correct answer: m= (media)
The m= line in SDP describes the media type (audio/video), port, transport (RTP/AVP), and the list of supported payload types.
Question 4: What is the primary difference between a SIP B2BUA and a SIP proxy?
- A B2BUA only handles audio; a proxy handles both audio and video
- A B2BUA terminates and re-originates both legs of a call; a proxy only forwards requests (Correct answer)
- A proxy supports multiple calls simultaneously; a B2BUA handles only one at a time
- A B2BUA uses TCP; a proxy uses UDP only
Correct answer: A B2BUA terminates and re-originates both legs of a call; a proxy only forwards requests
A B2BUA (Back-to-Back User Agent) acts as a UAS on the incoming leg and a UAC on the outgoing leg, giving it full control over the call.
Question 5: Which protocol is used to carry DTMF tones out-of-band in SIP/RTP calls?
- G.711 inband signaling
- RFC 2833 / RFC 4733 (RTP Named Telephone Events) (Correct answer)
- H.245 user input indication
- SIP INFO with DTMF body
Correct answer: RFC 2833 / RFC 4733 (RTP Named Telephone Events)
RFC 2833 (updated by RFC 4733) defines RTP Named Telephone Events as the standard method for carrying DTMF tones out-of-band in RTP streams.
Question 6: What does the SIP OPTIONS method primarily accomplish?
- It initiates a media session between two UAs
- It queries the capabilities of a SIP server or UA without establishing a call (Correct answer)
- It modifies an existing session's media parameters
- It cancels a pending INVITE request
Correct answer: It queries the capabilities of a SIP server or UA without establishing a call
OPTIONS is used to query what SIP methods, extensions, and codecs a server or UA supports without starting an actual call.
Question 7: Which VoIP security threat involves an attacker intercepting and replaying a valid SIP registration to hijack a user account?
- SIP flood attack
- Registration hijacking via replay attack (Correct answer)
- Toll fraud via SIP scanning
- Man-in-the-middle codec downgrade
Correct answer: Registration hijacking via replay attack
Registration hijacking occurs when an attacker captures a valid REGISTER request and replays it to overwrite the legitimate binding with their own address.
SSCA VoIP Technologies & Network Integration 4
What is the function of STUN (Session Traversal Utilities for NAT) in VoIP?