Advanced Technologies: Mobile, NGN and VoIP

Mobile Communication Evolution

Mobile systems progressed from analog voice to digital voice and messaging, then packet data and broadband multimedia. Higher generations add radio and core-network capabilities; the rate, delay, and services a user receives depend on the actual deployment, coverage, and load.

Gen	Main technology	Services / key feature
1G	Analog cellular (FM)	Voice; low security
2G	GSM, IS-95 CDMA	Digital voice, SMS; encryption, more capacity
2.5G	GPRS, EDGE	Always-on low-speed packet data
3G	UMTS, CDMA2000	Voice, video call, mobile internet; multimedia
4G	LTE, LTE-A	All-IP broadband; OFDMA, MIMO, low latency
5G	5G NR	eMBB/URLLC/mMTC; massive MIMO, slicing

Evolution of mobile communication from 1G to 5G

UMTS

The Universal Mobile Telecommunications System (UMTS) is a 3G system standardised by 3GPP and evolved from GSM/GPRS. Its widely deployed FDD radio interface uses WCDMA for voice, data, and multimedia; UMTS also includes TDD radio variants.

• Radio access: WCDMA uses nominal $5\,\mathrm{MHz}$ carriers and a $3.84\,\mathrm{Mchip/s}$ chip rate. UTRAN comprises Node B base stations and radio network controllers (RNCs).

• Core domains: Traditional UMTS supports circuit-switched voice through the MSC domain and packet data through SGSN/GGSN. The RNC connects through Iu-CS and Iu-PS respectively.

• Identity and mobility: The USIM supports subscriber identity and authentication. Roaming additionally requires compatible networks, subscription support, and operator agreements.

UMTS network architecture.

Element	Function
Node B	Radio transmission/reception (base station)
RNC	Controls Node B; radio resource management, handover
MSC/VLR	Circuit-switched switching; visited-area subscriber data
HLR/AuC	Permanent subscriber database; authentication keys
SGSN	Packet mobility and session management
GGSN	Gateway to external packet networks

Important UMTS elements

Note

UMTS vs GSM

• GSM: Uses $200\,\mathrm{kHz}$ carrier spacing and TDMA slots, primarily for digital voice and SMS with data extensions. Handover normally breaks the old radio connection before using the new one.

• WCDMA: Multiple users share a wider carrier using spreading and scrambling codes. Power control limits near-far interference, and a Rake receiver can combine resolvable multipath components.

• Handover: WCDMA supports soft handover between compatible same-frequency cells, maintaining multiple radio links during the transition. Inter-frequency and inter-system changes can require hard handover; not every UMTS transfer is a soft handover.

IMT-2000

IMT-2000 is the ITU framework for third-generation mobile systems. It defines requirements and a family of radio interfaces, including UMTS and CDMA2000, rather than a single universal air interface.

• Service goals: High-quality mobile voice, packet data, and multimedia with improved spectrum efficiency and support for roaming across suitable networks.

• Compatibility: Membership in the IMT-2000 family does not make different radio interfaces directly compatible. Multimode terminals and network interworking are needed when technologies differ.

The commonly quoted initial 3G target rates depend on mobility and environment; they are not guaranteed user throughput or ceilings for later 3G enhancements.

Environment	Target rate
High mobility	$\approx144\,kbit/s$
Pedestrian/urban	$\approx384\,kbit/s$
Indoor/low mobility	up to $\approx2\,Mbit/s$

Interface	System
IMT-DS	Direct spread WCDMA/UMTS
IMT-MC	Multi-carrier CDMA/CDMA2000
IMT-TC	Time-code CDMA, including UTRA TDD and TD-SCDMA
IMT-SC	Single carrier, UWC-136/EDGE
IMT-FT	Frequency time, DECT-based

Next Generation Network (NGN)

A Next Generation Network (NGN) is a packet-based network that separates service control from transport and delivers voice, data and video over a common IP infrastructure, with QoS support and generalised mobility.

• Functional separation: Packet transport forwards traffic; session control authenticates requests, routes sessions, and applies policy; application servers implement service logic. Defined interfaces allow these functions to evolve independently.

• Convergence: Fixed and mobile access can reach common voice, data, and video services. Media and signalling gateways interwork with legacy circuit networks.

• Transport requirements: QoS, security, and availability require provisioned capacity, scheduling, access controls, monitoring, and redundancy. They are not supplied merely by carrying packets over IP/MPLS.

Layered architecture of an NGN.

Component	Function
Softswitch/MGC	Software-based call/session control; controls gateways
Media Gateway (MGW)	Converts media between circuit (TDM/PCM) and packet (RTP)
Signalling Gateway	Interworks SS7 signalling with IP-based control
Application Server	Voicemail, conferencing, IN, IPTV logic
Session Border Ctrl.	Border security, NAT traversal, admission, topology hiding

Main NGN components

Note

NGN advantages

• Shared infrastructure: Statistical multiplexing and common transport and management can reduce duplicate equipment and operating costs across voice, video, and data services.

• Service changes: Software-based control and application servers allow new services without redesigning every transport node. Gateways permit staged migration while legacy subscribers remain connected.

VoIP

Voice over IP (VoIP) carries coded speech in packets over an IP network.

• Signalling path: Exchanges session requests, responses, and media descriptions to establish, modify, and release a call.

• Media path: Carries the speech packets between media endpoints, sometimes through a gateway or relay. A signalling proxy need not be on that path.

VoIP signalling and media paths.

Protocol Stack

• SIP: Session Initiation Protocol carries call signalling over transports such as UDP, TCP, or TLS over TCP. SDP offers and answers describe media formats, receiving addresses, and ports. H.323 is an alternative call-control framework.

• RTP and RTCP: RTP normally carries speech over UDP/IP; RTCP supplies reception-quality reports and timing information.

• Protection: TLS protects a SIP transport connection, not the associated RTP stream. Media protection requires a mechanism such as SRTP and suitable key establishment.

SIP-based VoIP protocol stack.

Note

H.323 note

• H.225.0 is call signalling, normally over TCP.

• H.245 is call control, normally over TCP.

• RAS denotes registration, admission and status, commonly over UDP.

Voice Coding and Quality

Codec	Notes	Bit rate
G.711	PCM, high quality, high bandwidth	$64\,kbit/s$
G.729	Compressed, common in VoIP	$8\,kbit/s$
G.723.1	Low bit rate	$5.3$/$6.3\,kbit/s$
AMR	Mobile networks	variable

• Processing chain: Speech coding $\to$ packetisation $\to$ network transport $\to$ jitter buffering $\to$ decoding $\to$ playout.

• Quality trade-offs: Longer packetisation intervals reduce header overhead but add delay and make each lost packet remove more speech. A larger jitter buffer tolerates greater arrival variation but also delays playout. Codec distortion, packet loss, late arrivals, and echo affect perceived quality.

Note

PSTN vs VoIP

• Traditional PSTN: Reserves bearer circuits, commonly $64\,\mathrm{kbit/s}$ PCM channels, controlled by SS7 or CAS. A basic wired analog telephone may receive operating power from the exchange.

• VoIP: Shares IP capacity and commonly uses SIP or H.323 signalling with RTP media. Codecs may be uncompressed PCM such as G.711 or compressed speech. Customer modems, routers, and IP phones need local power or backup even if the provider network remains operational.

PC-to-Ordinary-Phone Call Flow

A PC-to-ordinary-phone call crosses two bearer domains. The IP side carries packetised speech using RTP/UDP/IP, whereas the PSTN side carries speech on a circuit bearer, normally PCM/TDM in the digital network. The VoIP gateway, together with its associated call-control and signalling functions, forms the boundary between them.

Figure separates call setup from the bidirectional speech path and call release.

PC-to-ordinary-phone VoIP call flow across an IP–PSTN gateway.

1. Capture and packetisation: The PC microphone captures speech; an ADC samples it, a codec such as G.711 or G.729 encodes it, and the softphone places the coded frames in RTP/UDP/IP packets.

2. IP-side setup: When the user dials the telephone number, the softphone sends SIP or H.323 call-control signalling through the Internet toward the gateway or its call server.

3. Gateway interworking: The gateway control function interprets the destination number, selects a PSTN route, and maps the IP session state to PSTN signalling such as SS7/ISUP.

4. PSTN setup and alerting: The PSTN establishes a circuit toward the terminating exchange, which applies ringing to the called telephone and returns an alerting indication.

5. Answer: Going off-hook produces an answer indication. The gateway completes the SIP/H.323 setup toward the PC, and the conversation phase begins.

6. PC-to-phone speech: The gateway receives RTP packets, removes IP/UDP/RTP headers, compensates for jitter, decodes or transcodes the speech, and supplies PCM samples to the PSTN bearer.

7. Phone-to-PC speech: In the reverse direction, the gateway encodes the incoming PSTN speech and packetises it into RTP/UDP/IP. Speech therefore flows simultaneously in both directions.

8. Release: When either party hangs up, SIP BYE or the corresponding H.323 procedure is interworked with PSTN release signalling; the IP session and the PSTN circuit are cleared.

Note

Exam memory chain

• Route: PC $\to$ Internet $\to$ VoIP gateway $\to$ PSTN $\to$ ordinary phone.

• Four scoring labels: SIP/H.323 = IP call setup; RTP/UDP/IP = IP speech; SS7/ISUP = PSTN call setup; PCM/TDM = PSTN speech bearer.

SIP Call Establishment and Release

In a basic offer-in-INVITE exchange, the caller sends INVITE with SDP, the destination accepts with 200 OK and an SDP answer, and the caller sends ACK. Provisional responses report progress. Media uses the negotiated endpoints, not necessarily the proxy route; some call flows also establish early media before the final answer.

SIP call establishment and release.

Message	Function
INVITE	Requests a session; the SDP offer proposes media formats and receiving addresses/ports
100 Trying	Reports request processing; this provisional response is not required in every successful exchange
180 Ringing	Indicates that the called user is being alerted
200 OK to INVITE	Accepts the session; in this exchange it carries the SDP answer
ACK	Acknowledges the successful final response to INVITE
BYE	Terminates an established session; either endpoint may initiate it
200 OK to BYE	Completes the release transaction; no further ACK is sent

Real-time Transport Protocol (RTP)

• Media framing: RTP identifies the payload format, packet sequence, media sampling time, and source so the receiver can detect missing or reordered packets and schedule playout.

• Control reports: RTCP reports loss and interarrival jitter, identifies sources, and relates RTP media timestamps to a reference clock. These reports inform applications; RTCP does not reserve network capacity.

RTP packet header.

Field	Width	Function
V	bits	Protocol version, normally 2
P	bit	Indicates padding at the end of the packet
X	bit	Indicates a header extension after the CSRC list
CC	bits	Number of CSRC identifiers, from 0 to 15
M	bit	Profile-defined marker, such as a significant media boundary
PT	bits	Payload format identifier, interpreted through the applicable profile or negotiated mapping
Sequence number	bits	Increases by one per transmitted RTP packet; supports loss and reordering detection
Timestamp	bits	Sampling instant in the payload’s clock units
SSRC	bits	Identifies the synchronisation source within the RTP session
CSRC list	$32\times\mathrm{CC}$ bits	Identifies contributing sources, typically added by a mixer

The fixed RTP header is 12 bytes. With $\mathrm{CC}$ contributing sources and no extension, its length is $12+4\mathrm{CC}$ bytes; a header extension adds its own header and data. Padding, if present, is at the end of the packet, not part of this header length.

For G.711 with an 8 kHz timestamp clock and 20 ms packetisation, the timestamp increment is

$$\Delta T=8000\times0.020=160.$$

Note

RTP timing and delivery

• Sequence numbers advance per packet; timestamps advance in the media clock’s units. Packets carrying different parts of one video frame can share a timestamp while having different sequence numbers.

• RTP itself guarantees neither delivery nor bounded delay. Loss concealment, congestion control, retransmission where appropriate, and network QoS require additional application or network mechanisms.

DECT

DECT (Digital Enhanced Cordless Telecommunications) is a digital cordless technology for short-range wireless voice (cordless phones, wireless PBX), using TDMA/TDD.

• Access: Portable handsets share time slots with a fixed base station, using time-division duplexing for the two directions. Cordless PBX deployments can support handover between coordinated base stations.

• Capabilities and limits: Low-power handsets suit building-scale voice coverage. Authentication and encryption are available, but actual protection depends on the equipment and configuration; digital transmission alone does not ensure security.

GSM

GSM (Global System for Mobile Communications) is a 2G digital cellular system based on FDMA carriers and TDMA slots, normally using GMSK.

GSM network architecture.

Element	Function
BTS	Radio transmission with mobiles
BSC	Controls multiple BTSs; handover, frequency management
MSC	Call switching and mobility management
HLR / VLR	Permanent / temporary (roaming) subscriber database
AuC	Authentication and ciphering keys
EIR	IMEI-based equipment status database

Important GSM elements

Note

GSM radio access and channels

• Radio resources: Conventional GSM uses $200\,\mathrm{kHz}$ carrier spacing, 8 TDMA slots per carrier, GMSK modulation, and paired uplink/downlink frequencies (FDD). Control channels occupy some radio resources; eight slots do not always mean eight available full-rate voice calls.

• Logical channels:

  • Traffic channels (TCH) carry speech or user data.

  • Broadcast channels supply frequency correction, synchronisation, and system information.

  • Common control channels include RACH for access requests, AGCH for access grants, and PCH for paging.

  • Dedicated control channels include SDCCH for setup signalling and SACCH/FACCH for associated control, measurements, and urgent signalling during a connection.

• Interfaces: Um connects the mobile station to BTS; Abis connects BTS to BSC; A connects BSC to MSC. The mobile station comprises mobile equipment and SIM.

GSM mobile-originated call sequence.

1. The mobile requests access on RACH. The network returns an immediate assignment on AGCH, normally assigning an SDCCH for call setup.

2. The mobile sends a service request and identity. The network performs authentication and enables ciphering according to policy; equipment checks may also occur.

3. The mobile supplies the called number. The MSC selects a route and sends inter-exchange setup signalling, such as ISUP IAM for a PSTN destination.

4. The BSC assigns a TCH and forwards call progress. Answer establishes the conversational phase; the exact TCH-assignment timing depends on the call procedure.

5. Disconnect and release signalling clear the radio resources and associated core-network connection.

GSM supports telephony, emergency calls, SMS, circuit-switched data, and supplementary services such as forwarding and call waiting. GPRS and EDGE add packet-data access.

CDMA2000-1x

• System: CDMA2000-1x evolved from IS-95 and uses nominal $1.25\,\mathrm{MHz}$ carriers to support voice, SMS, and packet data.

• Code-domain sharing: Multiple users occupy the same carrier at the same time. Spreading and channelisation codes, timing, and receiver correlation separate the desired signal from other users.

• Soft handoff: A mobile can maintain compatible same-frequency radio links with more than one base station during a transition, reducing interruption and providing diversity.

Note

Processing gain

For direct-sequence spreading with chip rate $R_c$ and bit rate $R_b$, the chip-to-bit processing gain is

$$G_p=\frac{R_c}{R_b}\approx\frac{B_s}{B_d},\qquad G_{p,\mathrm{dB}}=10\log_{10}G_p.$$

The bandwidth ratio is approximate because occupied bandwidth depends on pulse shaping and modulation. State whether $R_b$ denotes coded or information bits when coding is present.

• Interference rejection: Despreading correlates the desired signal with its code while spreading uncorrelated narrowband interference over a wider band. A larger processing gain can improve interference tolerance, but does not create extra signal energy or guarantee secrecy.

• Capacity and power control: Reuse of the same carrier across cells increases interference. A strong nearby user can overwhelm weaker users unless transmit powers are controlled; capacity is limited by the resulting interference and required signal quality.

• Receiver cost: Code acquisition, tracking, multipath processing, and power-control loops add complexity compared with separating users into independent narrowband channels.

DS-SS vs FH-SS

Direct-sequence and frequency-hopping spread spectrum.

Feature	DS-SS	FH-SS
Instantaneous signal	Wideband, continuous	Narrowband on one hop
Rejection	Correlation / processing gain	Avoidance across hops
Near-far	Sensitive; power control in multiuser systems	Hop collisions and strong interferers still matter
Multipath	Rake can combine resolvable paths	Frequency diversity; equalisation may still be needed
Complexity	Wideband Rx, power control	Fast frequency synthesizer
Applications	CDMA (IS-95, CDMA2000, UMTS), GPS	Bluetooth, military, GSM slow FH

Direct-sequence vs frequency-hopping spread spectrum

DS-CDMA Synchronization

• Code alignment: The receiver must reproduce the incoming spreading sequence at the correct chip rate and phase. A misaligned code gives weak correlation and loses despreading gain.

• Acquisition: A serial/sliding correlator or matched filter searches candidate delays and, where necessary, frequency offsets. A sufficiently strong correlation peak identifies a candidate lock point, which must be confirmed against noise and false peaks.

• Tracking: A delay-lock loop compares early and late correlators around the prompt estimate. An error such as $e_\tau=|E|^2-|L|^2$ adjusts the local code clock to keep the peak centred. Carrier and symbol synchronisation are also required for demodulation.

DS-CDMA code acquisition and tracking.

Note

CDMA performance limits

• Diversity: A Rake receiver combines resolvable paths, and soft handover can supply cell diversity. Their benefit depends on the path quality and receiver implementation.

• Load: Additional users raise interference even when code identifiers remain available. Admission control may block new calls to protect existing ones; uncontrolled overload can reduce coverage or destabilise power control rather than degrade gracefully.

WiFi and WiMAX

WiFi provides wireless LAN access using IEEE 802.11, commonly in 2.4, 5, or 6 GHz bands according to version and regulation. Stations use contention-based access, with scheduled multiuser transmissions in newer versions. WiMAX uses IEEE 802.16 for broadband access, with a base station scheduling subscriber service flows.

Standard	Band	Key feature
802.11b	2.4 GHz	up to $11\,Mbit/s$
802.11a	5 GHz	OFDM, up to $54\,Mbit/s$
802.11g	2.4 GHz	up to $54\,Mbit/s$
802.11n	2.4/5 GHz	MIMO, higher throughput
802.11ac	5 GHz	wider channels, MU-MIMO
802.11ax	2.4/5/6 GHz	WiFi 6/6E, OFDMA

Feature	WiFi	WiMAX
Standard	IEEE 802.11	IEEE 802.16
Coverage	LAN / hotspot	MAN / metropolitan
Main use	Local networking	Broadband access / backhaul
QoS	WMM priorities; newer versions add scheduled multiuser access	Scheduled service flows with traffic parameters
Mobility	Roaming between access points	Fixed (802.16-2004) and mobile (802.16e) profiles
Deployment	Homes, offices, campus	ISP access, rural broadband

WiFi vs WiMAX

Technology Comparison

Technology	Access	Main use
DECT	TDMA/TDD	Cordless telephony
GSM	FDMA/TDMA	Digital voice, SMS (2G)
UMTS	WCDMA	Mobile multimedia (3G)
CDMA2000-1x	CDMA	Voice and packet data (3G)
WiFi	CSMA/CA, OFDM(A)	Local wireless internet (WLAN)
WiMAX	OFDM/OFDMA	Broadband wireless access (WMAN)
NGN	IP/MPLS packet	Converged services
VoIP	RTP over IP	Voice over packet network

Advanced technology comparison

Solved Examples

Note

Example: GSM carrier capacity

One GSM carrier has 8 TDMA slots per frame. Ignoring control-channel reservations and assuming full-rate speech, it can support up to 8 traffic channels. If one slot is reserved entirely for control, at most $8-1=7$ full-rate traffic channels remain on that carrier. Actual capacity depends on the configured logical channels and speech mode.

[!NOTE]

Example: CDMA processing gain

For chip rate $R_c=1.2288\,\mathrm{Mchip/s}$ and information rate $R_b=9.6\,\mathrm{kbit/s}$, the chip-to-information-bit processing gain is

$$G_p=\frac{1.2288\times10^6}{9.6\times10^3}=128,\quad G_{p,\text{dB}}=10\log_{10}128\approx21\,dB.$$

[!NOTE]

Example: VoIP packet rate

During continuous speech, a sender places $20\,\mathrm{ms}$ of audio in each RTP packet. Its packet rate in that direction is

$$\text{packets/s}=\frac{1}{20\times10^{-3}}=50\ \text{packets/s}.$$

Model Answers

Note

Example: NGN architecture, components, advantages and services (10 marks)

An NGN delivers multiple services over shared packet transport while separating application logic and session control from bearer forwarding.

• Functional layers:

  • Access connects users through DSL, FTTH, Ethernet, WiFi, or cellular systems.

  • Transport forwards packets over IP/MPLS and physical links, applying configured traffic treatment.

  • Control authenticates and routes sessions and coordinates admission and resource policy, using systems such as a softswitch or IMS.

  • Application servers provide telephony features, messaging, conferencing, and video-service logic. OSS/BSS and network management supply provisioning, charging, performance monitoring, and fault handling across the layers.

• Interworking and border components:

  • A media gateway converts TDM/PCM bearers to packet media under media-gateway-controller control.

  • A signalling gateway transports or interworks legacy signalling toward IP control systems. An SBC enforces session policies and can provide media relaying and NAT traversal at a border.

• Benefits: Shared links and management reduce duplicate service networks; software-based application logic permits service changes independently of transport upgrades. Services include VoIP/IP-PBX, conferencing, messaging, IPTV, and video-on-demand.

• Requirements: Redundant power and paths, admission control, security, emergency-call handling, and legacy interoperability must be explicitly designed. IP convergence alone guarantees neither availability nor QoS.

[!NOTE]

Example: IP-based convergence in NGN (10 marks)

IP convergence allows voice, video, and data to share access and core transport while retaining their distinct service requirements.

• Service transport:

  • Voice commonly uses SIP/IMS for session control, SDP for media negotiation, and RTP/UDP for speech.

  • Video uses real-time media transport or buffered streaming, with multicast or content-distribution systems where appropriate.

  • Data applications use transport protocols over the same IP forwarding infrastructure. Statistical multiplexing lets unused capacity from one service carry another service’s traffic.

• QoS and control:

  • Classification and DiffServ marking identify traffic classes; queue scheduling and policing apply their treatment. Admission control limits new real-time sessions to available resources.

  • MPLS can steer labelled traffic along engineered paths, but labels alone do not reserve capacity or guarantee delay. Forwarding, scheduling, and resource policies must work together.

  • Service control can change call policy independently of routers. Media gateways convert TDM and packet media, while signalling interworking maps SS7/ISUP call states to the IP call-control system.

• Interoperability constraints:

  • Codec mismatch can require transcoding; SIP extensions, IPv4/IPv6 differences, and NAT can affect session or media reachability.

  • E.164 numbers, SIP URIs, portability lookups, and emergency-location requirements must identify the correct destination and user location.

  • QoS and security policies must continue across operator boundaries. Charging, lawful-interception obligations, power backup, and failure recovery must be maintained during migration.

[!NOTE]

Example: GSM architecture and mobile-originated call flow (10 marks)

GSM is a 2G digital cellular system. Its conventional radio interface uses FDMA with $200\,\mathrm{kHz}$ carrier spacing, eight TDMA slots per carrier, GMSK, and FDD.

• Architecture:

  • The mobile station comprises equipment and SIM. BTSs provide radio transmission; a BSC manages radio channels and handovers for its BTSs. Together they form the BSS.

  • The MSC switches calls and controls mobility; the GMSC provides gateway call routing. HLR and VLR hold permanent and visited-area subscriber data, AuC supplies authentication material, and EIR stores equipment status.

  • OSS/OMC supports configuration, fault handling, and performance monitoring.

• Interfaces: Um is MS–BTS, Abis is BTS–BSC, and A is BSC–MSC. Core mobility and subscriber operations use MAP over the SS7 transaction stack.

For a mobile-originated circuit call:

1. The mobile sends a channel request on RACH; an AGCH immediate assignment normally gives it an SDCCH.

2. It sends a service request and identity. Authentication, ciphering activation, and optional equipment checking follow according to network policy.

3. The called number is sent in call setup. The MSC chooses the outgoing route and sends appropriate inter-exchange signalling, such as ISUP IAM.

4. The network assigns a TCH and reports call progress and alerting. Assignment timing can vary; answer places the call in its conversational phase and normally starts charging.

5. Disconnect and release procedures free radio resources, the A-interface bearer, and the outgoing circuit.

[!NOTE]

Example: VoIP operation and RTP header (5 marks)

• Media chain: Speech is coded, packetised, transmitted over IP, buffered against jitter, decoded, and played out. SIP commonly controls the session, while SDP negotiates the media formats, addresses, and ports used by RTP/UDP.

• SIP exchange: In a basic call, INVITE carries an SDP offer, 200 OK returns the answer, and ACK completes the successful INVITE exchange. Provisional responses can report processing or ringing. Either endpoint releases an established call with BYE, answered by 200 OK without another ACK.

• RTP format: The 12-byte fixed header contains V, P, X, CC, M, PT, a 16-bit sequence number, a 32-bit timestamp, and a 32-bit SSRC. A CSRC list and header extension may follow.

• Receiver use: Sequence numbers detect missing or reordered packets; timestamps identify media sampling time. G.711 with an $8\,\mathrm{kHz}$ clock and $20\,\mathrm{ms}$ packets advances its timestamp by 160. PT is interpreted using the session’s payload-format mapping; SSRC and CSRC identify sources.

• Reports and guarantees: RTCP reports loss, jitter, and timing. RTP alone supplies no bandwidth reservation, delivery guarantee, or encryption; suitable congestion control, QoS, and media protection such as SRTP are separate requirements.

[!NOTE]

Example: PC-to-ordinary-phone VoIP call flow (8 marks)

The route is PC/softphone $\to$ Internet $\to$ VoIP gateway $\to$ PSTN $\to$ ordinary telephone. Figure shows the required signalling and speech paths.

1. The microphone captures speech; the PC digitises it with an ADC, encodes or compresses it with a codec, and packetises it as RTP/UDP/IP.

2. The softphone sends SIP or H.323 call-setup signalling, including the called number, through the Internet to the gateway or associated call controller.

3. The gateway selects a PSTN route and interworks the IP call control with SS7/ISUP signalling.

4. The PSTN establishes a circuit to the terminating exchange, which rings the ordinary telephone.

5. When the called party answers, the answer indication returns through the PSTN and the gateway confirms connection to the PC.

6. In the forward direction, the gateway depacketises and decodes the RTP stream and converts it to PCM/TDM for the PSTN circuit.

7. In the reverse direction, it encodes and packetises PSTN speech into RTP; the established voice path is bidirectional.

8. On hang-up, SIP/H.323 and PSTN release signalling clear the packet session and the circuit resources.

Key distinction: SIP/H.323 and SS7/ISUP control the call; RTP and the PCM/TDM circuit carry the speech.

[!NOTE]

Example: WiFi and WiMAX comparison (5 marks)

• WiFi: IEEE 802.11 WLANs connect stations through access points, commonly in 2.4, 5, or 6 GHz bands. CSMA/CA manages contention; newer versions add OFDMA and multiuser MIMO. WMM supplies traffic priorities, while WPA2/WPA3 provide security when appropriately configured.

• WiMAX: IEEE 802.16 broadband access uses a base station to schedule subscriber service flows with specified QoS parameters. OFDM/OFDMA profiles support fixed access in 802.16-2004 and mobility through 802.16e.

• Deployment: WiFi is widely used in homes, offices, and hotspots. WiMAX targeted operator last-mile and metropolitan access, requiring suitable spectrum, sites, and customer equipment; LTE largely overtook it in mobile broadband deployment.

• Performance: WiFi contention and shared-band interference affect delay and throughput. WiMAX scheduling provides more explicit resource control, but neither technology guarantees a particular range or rate without a link budget, sufficient capacity, and suitable propagation conditions.

[!NOTE]

Example: DS-SS and FH-SS (4 marks)

• DSSS: Multiplies data symbols by a faster pseudonoise chip sequence, producing a continuously wideband signal. A synchronised correlator despreads the desired signal; uncorrelated narrowband interference is spread by that operation. Processing gain is $G_p=R_c/R_b\approx B_s/B_d$ under consistent bandwidth definitions.

• DSSS requirements: Code acquisition and tracking are essential. In multiuser CDMA, power control limits the near-far problem; a Rake receiver can exploit resolvable multipath. DSSS is used in CDMA systems and satellite navigation.

• FHSS: Moves a narrowband carrier among frequencies according to a shared hopping sequence. The receiver follows the same hop timing. A narrowband interferer affects only the hops overlapping its band, giving frequency diversity, but collisions and strong interference can still cause loss.

• FHSS requirements: Fast frequency synthesis and hop synchronisation replace the continuous wideband despreading operation. Bluetooth adaptive frequency hopping is an example. Neither hopping nor spreading codes alone provide cryptographic secrecy.

[!NOTE]

Example: DS-CDMA synchronization and importance (6 marks)

• Alignment: The receiver’s spreading code must match the incoming chip sequence in rate and phase. Incorrect alignment produces little correlation even when the correct code identity is known.

• Acquisition: A sliding correlator or matched-filter search tests possible code delays and carrier-frequency offsets. Integrated correlation energy is compared with a threshold, and a candidate peak is verified before tracking begins.

• Tracking: Early, prompt, and late correlators sample around $\hat\tau-\Delta$, $\hat\tau$, and $\hat\tau+\Delta$. A delay-lock discriminator such as $e_\tau=|E|^2-|L|^2$ adjusts the local code clock until the early and late powers balance around the correlation peak.

• Data and multipath: The prompt branch supplies despread data. Rake fingers can track different resolvable paths and combine them for diversity; carrier recovery and symbol timing are also needed.

• Capacity: Processing gain and near-unity frequency reuse allow many users to share a carrier, but interference and imperfect code separation limit capacity. Power control and admission control prevent strong or excessive users from degrading other connections.