B.Eng. (Hons.) Telecommunications. Examinations for / Semester 1

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B.Eng. (Hons.) Telecommunications Cohort: BTEL/12/FT Examinations for 2014-2015 / Semester 1 MODULE: IP TELEPHONY MODULE CODE: TELC 3107 Duration: 3 Hours Instructions to Candidates: 1.

1 B.Eng. (Hons.) Telecommunications Cohort: BTEL/12/FT Examinations for / Semester 1 MODULE: IP TELEPHONY MODULE CODE: TELC 3107 Duration: 3 Hours Instructions to Candidates: 1. Answer all questions. 2. Questions may be answered in any order but your answers must show the question number clearly. 3. Always start a new question on a fresh page. 4. All questions carry equal marks. 5. Total marks 100. This question paper contains 5 questions and 10 pages. Page 1 of 10

2 QUESTION 1: (20 MARKS) SIP ANSWER ALL QUESTIONS (a) State the transport protocols used by the SIP protocol. (4 marks) (b) Assume that your boss ask you to set up a low cost VoIP infrastructure solution within the premises of your company (around 200 employees) that will enable employees to use their smartphones to access VoIP services without going through the 3G cellular network. Describe briefly how this objective can be achieved. (6 marks) (c) Consider the VoIP setting illustrated in figure 1.1. Figure 1.1 Page 2 of 10

3 (i) List the messages labels from (1) to (12). (ii) Differentiate between requests and responses messages in (i). The message (1) code is detailed as follows: INVITE SIP/2.0 Via: SIP/2.0/UDP ; branch=z9hg4bk776asdhds Max-Forwards: 70 To: John From: Mark Call-ID: CSeq: INVITE Content-Type: application/sdp Content-Length: 228 v = 0 o = mark IN IP s =session SDP c = IN IP t =0 0 m = audio RTP/AVP 0 a = rtpmap: 0 PCMU/8000 (iii) What is the difference between the Call-ID and the CSeq fields? (iv) Explain the purpose of the parameter branch=z9hg4bk776asdhds in this message? (v) In the last 2 lines of the message what do and PCMU/8000 stand for? (10 marks) Page 3 of 10

4 QUESTION 2: (20 MARKS) H.323 (a) Figure 2.1 depicts the protocol stack for H.323. You will notice that H.3.23 splits into H.225 RAS Signalling, Call signalling and Control Signalling. (i) Why is it important to separate call signalling from control signalling in IP Telephony? Figure 2.1 (b) Amongst the components in an H.323 architecture, there are the Gatekeepers and Control Units. (i) What is the difference between the Gatekeeper (GK) and the Multipoint Control Unit (MCU) regarding their functionalities? (ii) How is the CODEC information negotiated in H.323? Page 4 of 10

5 (c) Draw a call flow diagram to show the different message exchanges in a direct call signalling with 2 terminals and one Gatekeeper as portrayed in figure 2.2. Figure 2.2 (4 marks) (d) Consider now a Gatekeeper routed call signalling. (i) Draw a call flow diagram to show the different message exchanges in a GK routed call signalling with 2 terminals and one Gatekeeper. (ii) Use a different colour on the same diagram answered in (i) to show message exchanges assuming there is request for bandwidth change. (6 marks) Page 5 of 10

6 QUESTION 3: (20 MARKS) MGCP (a) What would be the role for gateways if MGCP can sustain signalling between different types of Networks, for example a PSTN and a VoIP? (b) MGCP is meant to interconnect networks with different signalling systems. Explain why neither H.323 nor SIP are efficient to allow this through extensions. (c) Is the MGCP close to the IP philosophy? Discuss. (d) In what ways is the use of MGCP particular attractive to operators and hardware manufacturers? Page 6 of 10

7 QUESTION 4: (20 MARKS) RTP/RTCP (a) What are the quality issues in IP telephony that prompted the need for another Transport protocol for voice? (b) Compare the functionalities between the RTP and RTCP protocols (c) What are the functions of a translator and a mixer in a VoIP infrastructure? Illustrate your answer with the help of a diagram. (d) For a 16 kbps DVI4 audio conference with 100 participants (10 senders, 90 receivers) with an average packet size 100 bytes. (i) Calculate the period between 2 RTCP packets for a sender. (ii) Calculate the period between 2 RTCP packets for a receiver. Page 7 of 10

8 QUESTION 5: (20 MARKS) CODEC / Dial Plan (a) Voice quality is quite subjective. The 2 popular voice quality scales set by the ITU are the Mean Option Score (MOS) and the Perceptual Speech Quality Measurement (PSQM). Describe the MOS and the PSQM systems. (b) Assume the following protocols headers format. 40 bytes for IP (20 bytes) / User Datagram Protocol (UDP) (8 bytes) / Real-Time Transport Protocol (RTP) (12 bytes) headers. Compressed Real-Time Protocol (crtp) reduces the IP/UDP/RTP headers to 2 or 4 bytes (crtp is not available over Ethernet). 6 bytes for Multilink Point-to-Point Protocol (MP) or Frame Relay Forum (FRF).12 Layer 2 (L2) header. 1 byte for the end-of-frame flag on MP and Frame Relay frames. 18 bytes for Ethernet L2 headers, including 4 bytes of Frame Check Sequence (FCS) or Cyclic Redundancy Check (CRC). Page 8 of 10

9 CODEC information Table 5.1 Bandwidth requirements CODEC Bit rate(kbps) Codec sample size(b) Code sample interval (ms) MOS Voice payload size (B) Voice payload size(ms) PPS Bandwidth MP or FRF.12 (kbps) Bandwidth w/crtp MP or FRF.12 (kbps) Bandwidth Ethernet (kbps) G G Calculate the bandwidths for G.711 and G.729 as per the table 5.1. (c) Whenever we deal with external route configuration in a VoIP setting, we need to consider the following: Route patterns Route lists Route groups Route groups devices (i) Define the above routes. (ii) In Cisco Call Manager, what is the difference between On-Cluster Calls and Off-Cluster Calls? (d) There are actually different dial plan design guidelines according to the geographical location of the VoIP infrastructure as follows. Single Site Enterprise Multisite with distributed call processing Multisite with centralised call processing Page 9 of 10

10 (i) How do the above site configurations impact on the dial plan design? (ii) One of the advanced tools in Call Manager Dial Plan Tool Kit is the Automated Alternate Routing (AAR). Briefly describe the mechanism of the AAR. ***END OF QUESTION PAPER*** Page 10 of 10

Voice over IP (VoIP)

Voice over IP (VoIP) David Wang, Ph.D. UT Arlington 1 Purposes of this Lecture To present an overview of Voice over IP To use VoIP as an example To review what we have learned so far To use what we have