VoIP & SIP test (Intermediate)

Sample questions

5 sample questions from the VoIP & SIP test, with the answers. Samples are taken out of the live test, so you will not meet these when you sit it.

1. How is the audio level detected in WebRTC during a call?

   1. The audio level is detected by monitoring the frequency of the audio signal using the RTP timestamp.
   2. The audio level is detected using the RTCP receiver report (RR) and the volume is derived from the audio stream's energy levels.
   3. The audio level is automatically detected by the browser using the getUserMedia API.
   4. The audio level is determined by the RTP payload type and is calculated based on the codec used.

   Answer: B. The audio level is detected using the RTCP receiver report (RR) and the volume is derived from the audio stream's energy levels.

   WebRTC uses RTCP receiver reports (RR) to monitor audio levels by analyzing the energy levels in the audio stream. This allows for detecting changes in volume and ensuring consistent audio quality. Other options may involve incorrect methods, such as using RTP headers or relying on browser APIs like getUserMedia, which are not directly related to detecting audio levels during a call.

2. In a contact centre ACD system, which method ensures efficient call distribution by matching caller needs with agent skills, availability, and priority?

   1. Call queueing without agent skill matching
   2. Agent-initiated call pickup only
   3. Skill-based routing with agent availability and priority
   4. Random distribution to any available agent

   Answer: C. Skill-based routing with agent availability and priority

   In contact centre ACD (Automatic Call Distribution) systems, calls are distributed based on agent availability, skill sets, and priority to ensure efficient handling. This ensures that calls are routed to the most appropriate agent, improving customer satisfaction and operational efficiency. Other options are either incorrect or not relevant to the primary method of ACD call distribution.

3. In a contact centre IVR system, users are unable to navigate the menu using DTMF inputs. The call logs indicate that the DTMF inputs are not being recognized by the IVR. What is the most likely cause of this issue?

   1. The SBC is not normalizing SIP messages, causing a mismatch in the IVR's processing of DTMF inputs.
   2. The ACD is not routing calls properly, and the issue lies in the call routing configuration.
   3. The call recording system is not capturing audio correctly, leading to missing DTMF inputs in the logs.
   4. The IVR system is not properly configured to handle DTMF inputs, which are being sent as SIP INFO messages instead of being processed by the IVR.

   Answer: D. The IVR system is not properly configured to handle DTMF inputs, which are being sent as SIP INFO messages instead of being processed by the IVR.

   In a contact centre IVR system, DTMF inputs (such as key presses) must be processed correctly to navigate the IVR menu. If the IVR is configured to expect DTMF inputs as in-band audio or RTP events (RFC 4733) but receives SIP INFO messages instead, the IVR will not recognize the input, leading to a failure in menu navigation. The other options are unrelated to the issue of DTMF handling in the IVR architecture.

4. Which of the following statements accurately describes the robustness of the iLBC codec compared to G.729 in terms of packet loss resilience?

   1. iLBC is less robust to packet loss than G.729 because it lacks support for Forward Error Correction (FEC).
   2. iLBC is more robust to packet loss than G.729 due to its use of a robust speech coding algorithm and built-in Forward Error Correction (FEC).
   3. iLBC is more robust to packet loss than G.729 because it uses a higher bit rate.
   4. iLBC and G.729 have the same level of robustness to packet loss because they are both designed for VoIP networks.

   Answer: B. iLBC is more robust to packet loss than G.729 due to its use of a robust speech coding algorithm and built-in Forward Error Correction (FEC).

   iLBC (Internet Low Bitrate Codec) is designed to be more robust to packet loss compared to other codecs like G.729. This is due to its use of a robust speech coding algorithm and the inclusion of Forward Error Correction (FEC), which allows the codec to reconstruct lost packets without requiring retransmission. This makes iLBC particularly suitable for environments with high network loss, such as wireless or VoIP networks with unstable connections. G.729, while efficient in bandwidth usage, does not include FEC and is therefore less robust to packet loss.

5. Which statement accurately compares SIP dialog establishment with TCP connection establishment?

   1. SIP dialog establishment uses the Call-ID and CSeq to identify a session, whereas TCP connection establishment uses source and destination IP addresses and port numbers to identify a connection.
   2. SIP dialog establishment uses the Record-Route header and Contact header to identify a session, whereas TCP connection establishment uses source and destination IP addresses and port numbers to identify a connection.
   3. SIP dialog establishment uses the Via header and CSeq to identify a session, whereas TCP connection establishment uses source and destination IP addresses and port numbers to identify a connection.
   4. SIP dialog establishment uses the Call-ID, From tag, and To tag to uniquely identify a session, whereas TCP connection establishment uses source and destination IP addresses and port numbers to identify a connection.

   Answer: D. SIP dialog establishment uses the Call-ID, From tag, and To tag to uniquely identify a session, whereas TCP connection establishment uses source and destination IP addresses and port numbers to identify a connection.

   SIP dialog establishment uses the Call-ID, From tag, and To tag to uniquely identify a session, as this allows SIP to manage multiple sessions between the same endpoints. TCP connection establishment, on the other hand, uses source and destination IP addresses and port numbers to identify a connection, which is a lower-level protocol that does not concern itself with sessions or user agents.