3CX Intermediate Certification Practice Exam

3CX INTERMEDIATE

CERTIFICATION PRACTICE

EXAM

Question 1: What is the recommended method to connect a remote extension to a

3CX PBX hosted in the cloud?

Choices:

1) Port forwarding SIP ports directly to the IP phone 2) Using a 3CX Session Border Controller (SBC) or the 3CX App with tunnel 3) Setting up a site-to-site VPN for every individual home user 4) Using standard unencrypted UDP SIP trunking Correct Answer: Using a 3CX Session Border Controller (SBC) or the 3CX App with tunnel Explanation: The 3CX SBC or 3CX Tunnel is the recommended and most secure method to connect remote IP phones or clients, bypassing complex �rewall con�gurations.Page 1

Question 2: Which ports must be open on the �rewall for 3CX WebRTC and video conferencing functionality?

Choices:

1) TCP 5060 and UDP 5060 2) TCP 443 and UDP 5090 3) HTTPS 443 and TCP/UDP 5000-5004 (or speci�c WebRTC media ports as con�gured) 4) TCP 22 and TCP 80 only Correct Answer: HTTPS 443 and TCP/UDP 5000-5004 (or speci�c WebRTC media ports as con�gured) Explanation: WebRTC and video conferencing rely heavily on HTTPS and designated media/audio/video ports for real-time transport.Question 3: What is the primary function of a 3CX Session Border Controller (SBC)?

Choices:

1) To act as a �rewall router for the entire o�ce network 2) To multiplex SIP tra�c from multiple remote phones through a single encrypted tunnel to the 3CX PBX 3) To convert analog signals to digital VoIP packets 4) To replace the need for an active SIP trunk provider Correct Answer: To multiplex SIP tra�c from multiple remote phones through a single encrypted tunnel to the 3CX PBX Explanation: The 3CX SBC aggregates remote extension tra�c and tunnels it securely over a single port to the PBX, solving NAT and audio issues.Page 2

Question 4: When con�guring a VoIP Provider / SIP Trunk in 3CX, what is the purpose of the 'Re-invite' or 'PBX delivers audio' setting?

Choices:

1) It determines whether media streams �ow directly through the PBX or peer-to-peer between phone and trunk 2) It restarts the SIP service automatically every night 3) It forces all calls to be recorded locally 4) It changes the codec to G.711 automatically Correct Answer: It determines whether media streams �ow directly through the PBX or peer-to-peer between phone and trunk Explanation: 'PBX delivers audio' forces the audio stream to pass through the 3CX server, which is often required to resolve one-way audio issues behind complex NAT.Question 5: Which audio codec is universally mandated and preferred for standard PSTN calls in 3CX?

Choices:

1) G.722

2) G.711 A-law / U-law 3) Opus

4) G.729

Correct Answer: G.711 A-law / U-law

Explanation: G.711 (A-law in Europe/international, U-law in North America) is the standard uncompressed narrowband codec used for PSTN interconnects.Page 3