EEET2330 Voice & Video over IP Question Bank

Lecture 1:
1. What were the main purposes of PSTN? The Main purposes of PSTN are: 1) To carry the Analog voice signal sent by telephones. 2) Fax machines to send/receive Analog signals over the PSTN. 3) Dial-up internet services where data was converted into voice with the help of a modem for transmission over the copper wires. 4) Pulse Code Modulation (PCM) is used to convert an Analog telephone signal into a digital telephone signal. 2. Describe the three categories PSTN can be broadly divided. 1) Local Exchange Networks provide network access to the users 2) Inter-Exchange Networks provide network connectivity between exchanges 3) International Networks provide network connectivity between international gateways. 3. What is signalling? List different PSTN Signalling and describe them. Signalling is used for call setup, call management, and call termination. 1) User-to-Network signalling is used by users devices to communicate with the local telephone exchange (CO). 2) Network-to-Network signalling - is used by a local telephone exchange (CO) to communicate with another exchange. 4. How User-to-Network signalling works? User-to-Network signalling (DTMF) used by the (CO) works by utilizing a low and high frequency to determine which key/number has been pressed by the user. 5. Give examples of some Network-to-network signalling and describe. 1) Multi-Frequency (MF) is similar to DTMF, however it uses a different set of frequencies for signalling between switches installed at the (CO). 2) Robbed Bit Signalling (RBS) - the least significant bit of information is robbed from the channels that carry voice without affecting voice quality. 3) Signalling System 7 (SS7) - sends call setup and control messages between switches on a separate dedicated signalling network. 6. List some of the services provided by PSTN. 1) Local/National/International phone calls 2) Fax service 3) Dial-up internet 4) Message bank 5) Caller ID 6) Call diversion 7) Call waiting 8) Call back

7. Where is an E.164 number used? Give an example of an E.164 number mentioning its different components. An E.164 number is used in PSTN. Example: +61-3-99255647 +61 - Is the Country Code 3 - Is the Area Code 99255647 - Is the Subscriber Number 8. What role ITU play in telecommunication? Write its three different sectors. ITU is an international body that works for the development of telecommunication technology and provides regulatory and standards information. There are three (3) sectors: 1) ITU-R: Radio Communications 2) ITU-T: Telecom Standardization 3) ITU-D: Telecom Development 9. Define switching. Describe what different switching network would be suitable based on the size of the message and distance. 1) Switching is a communication network that can use a dedicated or shared network depending on the service required by a user. 2) Dedicated Line (Point-to-point) is used when users want to send long messages all the time from one location to another. 3) Switched or Shared Network is used when users want to send short messages over short or long distances. 10. List different types of switching. 1) Circuit Switching 2) Packet Switching 3) Message Switching 4) Cell Switching 11. Write the characteristics of circuit switching, packet switching, message switching, cell switching. 1) Circuit Switching - provides a type of communication in which a dedicated circuit is established only for the duration of a transmission. 2) Packet Switching - is a WAN technology that involves splitting the messages into smaller packets before transmission. These packets may follow different routes to the destination, where they are re-assembled into the original message. 3) Message Switching - involves receiving, formatting, storing and forwarding the message from source to the destination. 4) Cell Switching - combines the best features of circuit and packet switching. It is used in ATM networks. 12. Describe the Need for Voice over IP Network.

VOIP is needed because standard phone charges are very expensive for international and long-distance calls compared to internet charges. Therefore

VOIP uses the internet to place phone calls at cheaper cost. Lecture 2:
1. What is Voice over IP (VoIP)? VOIP is the packetisation and transport of classic public switched telephone system audio over an IP network. 2. How VoIP Technology works? The analog audio stream is encoded in a digital format, with possible compression, and encapsulation in IP for transport over a LAN, WAN or the public internet. 3. What are the Business Cases for VoIP? 1) Cost savings 2) Flexibility 3) Improved Billing System 4) Advanced features 4. Write different Categories of VoIP. 1) Amateur 2) Enterprise 3) ISP/Carrier 4) Telco Grade 5. List VoIP Components. 1) IP Networks 2) Gateways 3) Call Agents or Servers 4) User Devices 7. Describe gateway. A Gateway is a software or hardware based device that converts voice or fax calls (Analog signals) into packets (Digital signals) for delivery over IP networks and vice versa. 8. What are the jobs of CODEC? CODEC is used to digitise an Analog voice signal and vice versa. 9. Explain VAD. Voice Activity Detection (VAD) - detects the absence of voice signal and conserve bandwidth by preventing the transmission of these silent packets over the VoIP network. 10. Write VoIP Functions. 1) Signalling

2) Bearer Control 3) Codecs 11. Explain the Steps involved in digitising Analog voice signal. 1) Sampling - samples the Analog voice signal at periodic intervals 2) Quantisation - maps samples of a continuous range of the analog signals to a number of digital intervals. 3) Encoding - converts the quantised signals into binary numbers. 4) Compression - (Optional) 12. What are the important characteristics of the codec? 1) Bandwidth used by the digitised sound 2) Packet interval 13. What do you need to consider providing VoIP service? 1) Latency 2) Jitter 3) Bandwidth 4) Packet loss 5) Reliability 6) Security 15. Describe Major VoIP Protocols. 1) H.323 - ITU standard protocol for interactive conferencing. 2) MGCP - Emerging IETF standard for PSTN gateway control. 3) Megaco / H.248 - Joint IETF and ITU standard for gateway control with support for multiple gateway types. 4) SIP - IETF protocol for interactive and non-interactive conferencing. 5) RTP - IEFT standard media streaming protocol. 6) RTCP - IEFT protocol that provides out of band control information for an RTP flow. 16. Explain Bandwidth Implications of Codec. Different codecs have different bandwidth requirements. 17. Explain Impact of Voice Samples considering bandwidth requirement. The greater the Voice sample size the greater the bandwidth required. 18. What different types of Overhead you need to consider for VoIP? 1) Layer 2 Headers overhead 2) Security and Tunnelling overhead 3) Total bandwidth required 4) Effects of VAD

Lecture 3:
1. Describe some common features of any Cisco IP phone.

1) Placing a call 2) Answering a call 3) Ending a call 4) Parking a call 5) Transferring a call 6) Forwarding all calls 7) Placing a call on hold 8) Placing a conference call 9) Call history 10) Voice mail

Lecture 4:
1. List different types of voice ports a router must have for Analog phones and describe them. FXS (Foreign eXchange Subscriber) - is the port that actually delivers the Analog line to the subscriber. FXO (Foreign eXchange Office) - is the port that receives the Analog line. 3. How FXS/ FXO technically work? OUTBOUND CALL - When you wish to place a call: 1) You pick up the phone (the FXO device). 2) The FXS port detects that you have gone off hook. 3) You dial the phone number, which is passed as (DTMF) digits to the FXS port. INBOUND CALL: 1) The FXS port receives a call, and then sends a ring voltage to the attached FXO device. 2) The phone rings. 3) As soon as you pick up the phone you can answer the call. 4) Ending the call normally the FXS port relies on either of the connected FXO devices to end the call. 4. Describe different Configuration Parameters for FXS and FXO. 1) Signal - Sets the signalling type for the FXS port. 2) Cptone - Configures the appropriate call-progress tone for the local region. 3) Description - Configures a description for the voice port. 4) Ring frequency - Configures a specific ring frequency (in Hz) for an FXS voice port. 5) Ring cadence - Configures the ring cadence for an FXS port. 6) Busy-Out - Configures the ability to busy out an Analog port, perhaps for maintenance purposes. 7) Station ID name - Provides the station name associated with the voice port. 8) Station ID number - Provides the station number that is to be used as the calling number associated with the voice port. 5. Explain E&M Ports. E&M ports provide signalling that is used generally for switch-to-switch or switch-tonetwork trunk connections.

6. What is ECHO cancellation? Echo cancellation is the process of removing echo from a voice communication in order to improve the voice call quality. 7. Define different types of call with diagram: Local Calls, On-Net Calls, Off-Net Calls, Private Line, Automatic Ringdown (PLAR), PBX-to-PBX Calls, Cisco Call Manager to Cisco Call Manager, Off-Net to Off-Net Calls, On-Net to Off-Net Call. Local Calls: Occur between two telephones connected to one Cisco voice-enabled router. On-net calls: Occur between two telephones on the same data network Off-net calls: used to gain access to the public switched telephone network (PSTN) PLAR calls: automatically connect a telephone to a second telephone when the first telephone goes off hook. PBX-to-PBX calls: Originate at a PBX at one site and terminate at a PBX at another site while using the network as the transport between the two locations. Cisco Call Manager to Cisco Call Manager: Cisco Call Manager assesses whether the call is destined for another IP phone under its control or whether the call must be routed through a remote Cisco Call Manager for call completion. On-Net to Off-Net Call: Originates on an internal network and is routed to an external network, usually to the PSTN. Off-Net to Off-Net Calls: is when a call is made from a PSTN to VoIP network (off-Net) and forwarded to another PSTN number (off-Net) through a VoIP network.

8. Describe Dial Plans and Dial Peers Dial Plan - allows people to call each other by dialling a number on the telephone. Dial Peers - are used to configure dial plans and to identify call source and destination endpoints. 9. List different types of Dial Peers. Explain them. 1) POTS dial peers - are used to connect a physical voice port to a PBX, a telephone, or the PSTN. 2) VOIP dial peers - are used to define packet voice network attributes and map dial strings to a remote router or device. 10. What is Destination Patterns Destination pattern - is used to associate a dial string with a particular telephony device. 11. Write Common Wildcard Symbols Used in Destination Patterns 1) . 2) [ ] 3) T 13. Define - Port, Session target, Call leg. Port - Command associates a dial peer to a physical router port. Session target - command associates a dial peer to a network IP address of a

remote router or device through which the call should be routed. Call Leg - is a logical connection between two routers or between a router and a telephony-capable device. 14. How can you verify and Troubleshoot voice Ports 1) Check for dial tone (FXS only) 2) Check for DTMF tones (FXS only) 3) Use the show voice port command 4) PBX configuration and voice port compatibility 5) Check the physical installation of the hardware 15. How destination pattern is matched in router? Destination pattern is matched based on the longest number match.

Lecture 5:
1. How router or gateway matches inbound dial peers? The router or gateway matches inbound dial peers by using the following in the following order: 1) Incoming called number 2) Answer address parameter 3) Destination pattern 4) Port parameter 5) Default dial peer 0 2. Describe the impact of Echo Cancellation. Echo cancellation stops voice echoes from being played back at the source and saves bandwidth. 3. How Echo Cancellation works? Echo cancellation keeps a certain-sized sample of the outbound voice and calculates what that same signal looks like when it returns as an echo. Echo cancellation then attenuates the inbound signal by that amount to cancel the echo signal. 4. How router or gateway matches outbound dial peers? The router or gateway matches outbound dial peers on a digit by digit basis, and routes the call when a full match is made. 5. List hunt-group commands and explain their purpose. 1) Preference - sets the priority peers. 2) Huntstop - disables dial peer hunting. 3) Dial-peer hunt - changes the default selection order for hunting. 4) show dial-peer voice summary - shows the current settings for dial-peer hunt. 6. How router selects dial peer in hunt group? 1) The router matches the most specific telephone number

2) The router matches according to the preference setting 3) The router matches randomly 7. Explain how Digit Consumption and Forwarding happens? For POTS dial peers - the router consumes the left-justified digits that explicitly match the destination pattern and forwards wildcarded digits. For VOIP dial peers - the router forwards all digits collected. 8. What is Digit Collection? How it works? 1) The router collects digits, one at a time, until it can match an outbound dial peer 2) After a match is made, router immediately places the call 3) No further digits are collected 9. Write the Digit Manipulation Commands and explain. 1) prefix Configured as a Dial-peer command Adds digits to the front of the dial string before it is forwarded to the telephony interface 2) forward-digits Configured as a Dial-peer command Controls the number of digits forwarded to the telephony interface 3) num-exp Configured as a Global command Expands an extension into a full telephone number or replaces one number with another 4) translation-rule Configured as a Global and dial-peer command Digit translation rules used to manipulate the calling number digits for a voice call 10. Give example of sample configuration using the prefix command, forward-digits command, number expansion table (num-exp) command and translation-rule command. Prefix command example: dial-peer voice 1 pots destination-pattern 555. prefix 555 port 1/0/0 Forward-digits command example: dial-peer voice 1 pots destination-pattern 555. forward-digits 7 port 1/0/0 Number expansion table (num-exp) command example: num-exp 2 5552 dial-peer voice 1 pots destination-pattern 5552... port 1/0/0

Translation-rule command example: translation-rule 5 rule 1 2401 5552401 dial-peer voice 1 pots translate-outgoing called-number 5

Lecture 6:
1. Write an Overview of Signalling. Signalling and call control are fundamental to the call establishment, management, and administration of voice communication in an IP network. 2. What is Endpoint? Give examples of some Endpoints. Endpoints are typically simple single-user devices, such as terminals, that support either a voice process or a gateway. 3. Define Common control. What different services are provided by Common control? Common control components provide call administration and accounting. 1) Call status 2) Address registration and resolution 3) Admission control 4. List several call control models and their corresponding protocols. 1) H.323 2) SIP 3) MGCP 4) H.248/Megaco 5) SAP 6) RTSP 7) Cisco Call Manager 5. Briefly describe some call control models. 1) H.323 - describes the architecture to support multimedia communications over networks without quality of service (QoS) guarantees. 2) SIP - call control model for creating, modifying, and terminating multimedia sessions or calls. 3) MGCP - is a call control model that controls VoIP gateways from an external call control element or call agent. 4) H.248/Megaco - used in environments in which a media gateway consists of distributed subcomponents, and communication is required between the gateway subcomponents. 5) SAP - describes a multicast mechanism for advertising the session characteristics of a multimedia session 6) RTSP - describes a model for controlled, on-demand delivery of real-time audio and video. 7) Cisco Call Manager - is a proprietary Cisco Systems implementation of a call control environment that provides basic call processing, signaling, and connection services to configured devices, such as IP phones, VoIP gateways, and software applications.

6. Write about the standardization organizations who work for improving VoIP. International Telecommunications Union (ITU) - provides regulatory and standards information about Telecommunications. Internet Engineering Task Force (IETF) - work to accelerate the evolution of the internet architecture. 7. List three different series of standard developed by ITU-T. 1) G - Series 2) H - Series 3) T Series 8. What are the two main categories of protocols used to provide VoIP communications? Describe them. Transport Protocols - are responsible for transmission of digitised voice from one end to another over the VoIP networks. Signalling Protocols - are required for call management 9. Describe different transport Protocols: Transmission Control Protocol (TCP), User Datagram Protocol (UDP), Real-time Transport Protocol (RTP), and Real-time Transport Control Protocol (RTCP). TCP - offers a guaranteed delivery of data by providing connection-oriented, reliable packet delivery through an IP network. UDP - is an unreliable, connectionless protocol that offers no guarantee of data delivery. RTP - provides network transport functions required for multimedia applications to be transmitted in real time. RTCP - controls RTP and monitors data delivery. 10. What are the four signalling protocols popularly used in VoIP? 1) H.323 2) SIP 3) MGCP 4) H.248 11. Describe different signalling Protocols: H.323, Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), Media Gateway Controller (MEGACO) H.248 1) H.323 - is used to provide end-to-end call signalling in VoIP communications 2) SIP is an alternative end-to-end signalling protocol for VoIP communications 3) MGCP - is used for signalling between gateways and call control elements 4) H.248 - is also used for signalling between gateways and call control elements (Preferred by VSPs) 12. Describe VoIP Network Operation. When a user makes a VoIP phone call, the first step preformed is the digitisation of the users voice by a codec. Silence suppression and compression is done as well. VOIP protocols are added to these packets to prepare them for transmission over

the IP network. When these packets arrive at the destination, they are arranged in a sequence. 13. Why Quality of Service (QoS) must be ensured over VoIP? QoS must be ensured over VOIP because if voice traffic flows in a disorderly/delayed fashion, the receiver of the traffic will not be able to understand what is said. 14. What are the four main factors (in addition to noise) that can influence the QoS in VoIP communications? Describe them. Bandwidth - the more bandwidth available, the better the QoS and vice versa Packet Loss - less packet loss results in better QoS and vice versa Delay - less delay experienced by the digitised voice packets during end-to-end transmission, results in better QoS and vice versa. Jitter - less jitter experienced by different digitised voice packets during end-to-end transmission, results in better QoS and vice versa.

Lecture 7:
1. Write an overview of H.323. H.323 was originally created to provide a mechanism for transporting multimedia applications over LANs. 2. What protocols are defined by H.323? 1) H.245 (Capabilities exchange) 2) H.225 (Call setup) 3) H.225 (RAS control for call routing) 3. What are the functions of H.245? 1) Logical channel signalling 2) Capabilities exchange 3) Master determination 4) Mode request 5) Timer and counter values 6) RAS signalling function 5. What different terminals exist in H.323 network? 1) H.323 terminal 2) Multipoint Control Unit (MCU) 6. H.323 gateway performs what services? 1) Translation between audio, video, and data formats 2) Conversion between call setup signals and procedures 3) Conversion between communication control signals and procedures 7. Why do we need IP-to-IP Gateways? We need IP-to-IP Gateways because it facilitates easy and cost-effective connectivity

between independent VoIP service provider networks.

9. Where do we need a gatekeeper? We need a gatekeeper between our private network and the public network. 10. When a gatekeeper is included, what functions it must perform? 1) Address translation 2) Admission control 3) Bandwidth control 4) Zone management 11. Describe Multipoint Conference Components. 1) Multipoint Controller (MC) - provides the functions that are necessary to support conferences involving three or more endpoints 2) Multipoint processor (MP) - adds functionality to multipoint conferences 3) Multipoint Control Unit (MCU) - provides support for multipoint conferences by incorporating one MC and zero or more MPs. 12. Explain the component relationships for call establishment and management. 1) Endpoint to endpoint - endpoints locate other endpoints through non-standard mechanisms and initiate direct communication between the endpoints. 2) Endpoint to gatekeeper - When a gatekeeper is added to the network, endpoints interoperate with the gatekeeper using the RAS channel. 3) Gatekeeper to gatekeeper - In the presence of multiple gatekeepers, gatekeepers communicate with each other on the RAS channel. 13. Write various RAS message types and explain them. 1) Gatekeeper discovery 2) Terminal and gateway registration 3) Terminal and gateway un-registration 4) Location request 5) Call admission 6) Bandwidth change 7) Disengage 8) Status queries 14. Draw the block diagram of H.323 Basic Call Setup and describe the procedure.
1) The originating gateway initiates an H.225.0 session with the destination gateway on registered TCP port 1720. The gateway determines the IP address of the destination gateway internally. The gateway has the IP address of the destination endpoint in its configuration or it knows a Domain Name System (DNS) resolvable domain name for the destination. 2) Call setup procedures based on Q.931 create a call-signalling channel between the endpoints. 3) The endpoints open another channel for the H.245 control function. The H.245 control function negotiates capabilities and exchanges logical channel descriptions. 4) The logical channel descriptions open Real-Time Transport Protocol (RTP) sessions. 5) The endpoints exchange multimedia over the RTP sessions, including exchanging call quality statistics using Real-Time Transport Control Protocol (RTCP).

15. Draw the block diagram of H.323 Fast Connect Call Setup and describe the procedure.
1) The originating gateway initiates an H.225.0 session with the destination gateway on registered TCP port 1720. 2) Call setup procedures based on Q.931 create a combined call-signalling channel and control channel for H.245. Capabilities and logical channel descriptions are exchanged within the Q.931 call setup procedure. 3) Logical channel descriptions open RTP sessions. 4) The endpoints exchange multimedia over the RTP sessions.

16. How call flows with a Gatekeeper? 1) The gateway sends an Admission ReQuest (ARQ) to the gatekeeper to initiate the procedure. The gateway is configured with the domain or address of the gatekeeper. 2) The gatekeeper responds to the ARQ with an Admission ConFirmation (ACF). In the Confirmation, the gatekeeper provides the IP address of the remote endpoint. 3) When the originating endpoint identifies the terminating endpoint, it initiates a basic call setup. 4) Before the terminating endpoint accepts the incoming call, it sends an ARQ to the gatekeeper to gain permission. 5) The gatekeeper responds affirmatively, and the terminating endpoint proceeds with the call setup procedure. 17. How Gatekeeper-Routed Call Signaling occurs? 1) The gatekeeper responds to an ARQ and advises the endpoint to perform the call setup procedure with the gatekeeper, not with the terminating endpoint. 2) The endpoint initiates the setup request with the gatekeeper. 3) The gatekeeper sends its own request to the terminating endpoint and incorporates some of the details acquired from the originating request. 4) When a connect message is received from the terminating endpoint, the gatekeeper sends a connect message to the originating endpoint. 5) The two endpoints establish an H.245 control channel between them. The call procedure continues normally from this point. 18. Explain the three types of conferences defined by H.323. 1) Centralized multipoint conference - The endpoints must have their audio, video, or data channels connected to an MP. 2) Distributed multipoint conference - The endpoints do not have a connection to an MP. Instead, endpoints multicast their audio, video, and data streams to all participants in the conference. 3) Ad hoc multipoint conference - Any two endpoints in a call can convert their relationship into a point-to-point conference. If neither of the endpoints has a collocated MC, the services of a gatekeeper are used. 19. Describe H.323 replication strategies. 1) Hot Standby Router Protocol (HSRP): HSRP allows two gatekeepers to share both an IP address and access to a common LAN; however, at any time, only one gatekeeper is active. Endpoints are configured with the name of the gatekeeper, which they can resolve using the Domain Name System (DNS) or the IP address of the gatekeeper. 2) Virtual Router Redundancy Protocol (VRRP): VRRP allows a group of

gatekeepers on a multiaccess link to use the same virtual IP address, with one device acting as the master virtual router and one or more devices configured to be available as backup virtual routers. Endpoints are configured with the name or virtual IP address of the gatekeeper. VRRP is defined by the Internet Engineering Task Force (IETF) as RFC 3768. 3) Multiple gatekeepers with gatekeeper discovery: Deployment of multiple gatekeepers reduces the probability of the total loss of gatekeeper access. However, adding new gatekeepers presents a new challenge. Each gatekeeper creates a unique H.323 zone. Because an H.323 endpoint is associated with only one gatekeeper at a time (in only one zone at a time), endpoints are configured to find only one of several working gatekeepers. Fortunately, a gateway can be configured with an ordered list of gatekeepers or to use IP multicast to locate a gatekeeper. 4) Multiple gatekeepers configured for the same prefix: Gatekeepers send location request (LRQ) messages to other gatekeepers when locating an endpoint. By supporting the same prefix on multiple gatekeepers, the LRQ can be resolved by multiple gatekeepers. This strategy makes the loss of one gatekeeper less significant. 5) Multiple gateways configured for the same prefix: Survivability is enhanced at the gateway with multiple gateways that are configured to reach the same switched circuit network (SCN) destination. By configuring the same prefix of destinations in multiple gateways, the gatekeeper sees the same prefix more than once as each gateway registers with its gatekeeper. 20. Why would you need a H.323 proxy server? You would need an H.323 proxy server because it can avoid the shortcomings of a direct path in cases where the direct path between two H.323 endpoints is not the most appropriate. Lecture 8: 1. Describe SIP and its associated standards. SIP is a simple extensible protocol that creates, modifies, and terminates multimedia sessions with one or more participants. SIP leverages various IETF standards including RTP, RTCP, HTTP, SDP, DNS, SAP, and RTSP. 2. How Multimedia sessions are established and terminated using SIP? Multimedia sessions are established and terminated by these services: 1) User location services 2) User capabilities services 3) User availability services 4) Call setup services 5) Call handling services 3. Explain the functional components of UAs (User Agents). 1) User agent client (UAC) - A client application that initiates a SIP request 2) User agent server (UAS) - A server application that contacts the user when a SIP invitation is received and then returns a response on behalf of the user to the invitation originator. 4. List some SIP UA devices.

1) IP telephone 2) Gateway 5. What different types of SIP servers exist? Describe them. 1) Proxy server - is an intermediate component that receives SIP requests from a client and forwards those requests on behalf of the client to the next SIP server in the network. 2) Redirect server - provides a user agent (UA) with information about the next server that the UA should contact. 3) Registrar server - makes requests from UACs for registration of their current location. 4) Location server - is an abstraction of a service providing address resolution services to SIP proxy or redirect servers. 6. Write about SIP headers. There are four (4) SIP types of headers: 1) General header 2) Entity header 3) Request header 4) Response header. The first two types of headers appear on both message types. The latter two types of headers are specific to request and response, respectively. 7. Give example of SIP address format. 1) Fully qualified domain names (example: sip:jdoe@cisco.com) 2) E.164 addresses (example: sip:14085551234@gateway.com; user=phone 3) Mixed addresses (example: sip:14085551234; password=changeme@10.1.1.1 sip:jdoe@10.1.1.1) 8. Describe with diagram how Direct Call Setup happens in SIP. 1) The originating UAC sends an invitation to the UAS of the recipient. 2) If the UAS of the recipient determines that the call parameters are acceptable, it responds positively to the originator UAC. 3) The originating UAC issues an ACK. 9. Describe with diagram how Call Setup Using a Proxy Server occurs in SIP. 1) The originating UAC sends an invitation to the proxy server. 2) The proxy server, if required, consults the location server to determine path to the recipient & its IP address. 3) The proxy server sends the INVITE to the UAS of the recipient. 4) If the UAS of the recipient determines that the call parameters are acceptable, it responds positively to the proxy server. 5) The proxy server responds to the originating UAC. 6) The originating UAC issues an ACK. 7) The proxy server forwards the ACK to recipient UAS. 10. Describe with diagram how Call Setup Using a Redirect Server happens in SIP. 1) The originating UAC sends an invitation (INVITE) to the redirect server. 2) The redirect server, if required, consults the location server to determine the path

to the recipient and its IP address. 3) The redirect server returns a moved response to the originating UAC with the IP address obtained from the location server. 4) The originating UAC acknowledges the redirection. 5) The originating UAC sends an INVITE to the remote UAS. 6) If the UAS of the recipient determines that the call parameters are acceptable, it responds positively to the UAC. 7) The originating UAC issues an ACK 8) The UAC and UAS now have all the information that is required to establish RTP sessions between them. 11. Explain the Survivability Strategies for SIP In a SIP environment, the failure of a network server cripples UAs that are dependent on that server. The most obvious way to preserve access to the critical components is to implement multiple instances of access. Lecture 9: 1. List the basic Requirements for Broadband Multimedia Services. 1) DVB Transcoding 2) Encoding - Compression Technologies 3) Video Streaming Servers - Unicast / Multicast 4) Digital Rights Management (DRM) 5) Multiple Specialized Media Servers 6) Middleware 7) The IP Set top Box 2. Who are the prime business drivers for IP/TV? 1) US 2) Canada 3) Europe 4) UK 5) Australia 3. What are the significant business opportunities for IP/TV? 1) Broadcast or Multicast TV 2) Video on Demand 3) Pay-Per-View 4) Internet services 5) T-Commerce 6) Horse Racing 7) Closed Members Group Entertainment 8) Karaoke 9) Financial Services for Stock Trading and Portfolio 10) Management 4. Describe the IP/TV business segments. 1) TELCOS / ISP 2) IPTV on Public Internet Networks 3) HOSPITALITY

5. How IP/TV on Public Internet Networks is emerging? IP/TV on the Public Internet Networks is emerging rapidly, and aggregated projections of this segment looks equal to or even bigger than the TELCOS and ISPs.