SIP Interoperability and Integration 2 — Questions and Answers
Question 1: Which SIP header is used to indicate the protocol version and transport when a request traverses multiple proxies?
- Via (Correct answer)
- Route
- Contact
- Record-Route
Correct answer: Via
The Via header records each hop a SIP request traverses, including the protocol version and transport (UDP, TCP, TLS).
Question 2: When integrating SIP with H.323, which gateway function translates between the two signaling protocols?
- Media gateway controller
- Interworking function (IWF) (Correct answer)
- Border element
- Session Border Controller
Correct answer: Interworking function (IWF)
An Interworking Function (IWF) handles the translation of signaling and media between SIP and H.323 endpoints.
Question 3: Which IETF RFC defines the SIP-T framework for carrying ISUP information within SIP messages?
- RFC 3261
- RFC 3372 (Correct answer)
- RFC 4474
- RFC 3711
Correct answer: RFC 3372
RFC 3372 defines SIP-T, which enables ISUP signaling to be encapsulated within SIP for PSTN interworking.
Question 4: A SIP UA sends an INVITE to a PSTN gateway that supports only G.711. The UA's SDP offer includes G.729 and G.711. What determines which codec is used?
- The UA always wins the codec negotiation
- The gateway selects the first codec in the offer
- SDP answer from the gateway selects the mutually supported codec (Correct answer)
- RTCP negotiates the codec after the call is established
Correct answer: SDP answer from the gateway selects the mutually supported codec
The gateway's SDP answer selects one or more codecs from the offer that it supports, finalizing the codec choice.
Question 5: In a SIP-to-WebRTC interworking scenario, which protocol adaptation is typically required for media?
- RTP to SRTP with DTLS-SRTP keying (Correct answer)
- RTP to RTMP
- G.711 to opus transcoding only
- SDP to JSEP conversion only
Correct answer: RTP to SRTP with DTLS-SRTP keying
WebRTC mandates DTLS-SRTP for media encryption, so a SIP-WebRTC gateway must convert plain RTP to SRTP with DTLS-SRTP keying.
Question 6: Which SIP mechanism allows a third-party call controller to set up calls between two endpoints without being in the media path?
- SIP REFER
- 3PCC (Third Party Call Control) (Correct answer)
- SIP NOTIFY
- SIP UPDATE
Correct answer: 3PCC (Third Party Call Control)
Third Party Call Control (3PCC) allows a controller to establish a session between two UAs by sending separate INVITEs and coordinating SDP.
Question 7: When a SIP trunk interconnects two enterprises with overlapping private numbering plans, which element typically resolves number translation?
- DNS SRV record
- Session Border Controller with dial plan translation (Correct answer)
- SIP Registrar
- STUN server
Correct answer: Session Border Controller with dial plan translation
A Session Border Controller (SBC) with dial plan translation rewrites numbers to avoid conflicts between overlapping private numbering plans.
SIP Interoperability and Integration 2
Which SIP header is used to indicate the protocol version and transport when a request traverses multiple proxies?