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BAI613 Module 2 - Voice over IP Technology. Module Objectives 1. Describe the benefits of IP Telephony/Packet Telephony/VoIP over traditional telephone.

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Presentation on theme: "BAI613 Module 2 - Voice over IP Technology. Module Objectives 1. Describe the benefits of IP Telephony/Packet Telephony/VoIP over traditional telephone."— Presentation transcript:

1 BAI613 Module 2 - Voice over IP Technology

2 Module Objectives 1. Describe the benefits of IP Telephony/Packet Telephony/VoIP over traditional telephone systems 2. Analyze some VoIP Applications such as Packet Telephony Call Centers, Service Provider Calling Card applications

3 Module Objective 1 Describe the benefits of IP Telephony/Packet Telephony/VoIP over traditional telephone systems

4 Separate Voice and Data Networks

5 Benefits of IP Telephony/Packet Telephony/VoIP over traditional telephone systems  Monetary Savings  TCO is lower (over a longer period of time).  Short term savings might not be substantial  Consolidation of network infrastructure, staffing, and Service Provider  New or Next-Generation Applications

6 Key Benefits of VoIP (contd.)  Easier Moves, Adds and Changes (MACs)  Dynamic Host Configuration Protocol (DHCP) based IP configuration  E.G. on CCM, profiles setup for phones based on MAC addresses

7 Key Benefits (contd..)  One Information Services (IS) department for both voice and data networks  Common infrastructure  Additional tools not needed  Unified Messaging – Key benefit, Voic , Videomail and in the same inbox. One location for checking your messages.

8 IP Telephony Advantages

9 Converged VoIP Network – A more efficient solution

10 Difference Between Converged Technologies  Voice Over Packet  Transporting circuit voice over another technology  Could be VoIP, VoFR, VoATM  LAN Telephony  Telephony on a data network using Layer 2 only (Ethernet)

11 Voice over Frame Relay

12 Voice over ATM

13 Voice over IP (IP Telephony)  VoIP, also referenced as IP telephony when integrated with IP-based signaling and call control, is how almost all new deployments are being implemented.  VoIP provides transport of voice over the IP protocol family. IP makes voice globally available regardless of the data link (Ethernet, ATM, Frame Relay) protocol in use.

14 Enterprise IP Telephony vs Internet Telephony  Internet Telephony uses IP but uses a public infrastructure (i.e. Internet)  The Internet is unmanaged and not engineered for voice and therefore cannot offer any type of SLA.  The common result is poor voice quality  Enterprise IP Telephony is managed and engineered for voice offering a high quality network in which to transport voice. (How to do this will be learned in Topic 4)  However, industry has and continues to re- engineer the Internet in order to carry converged applications

15 Packet Telephony Call centers Challenges faced by Circuit Switched Call centers: Unable to grow in smaller chunks Telecommuting expensive Many toll-free numbers Misrouted/rerouted calls Multiple centers Percentage distribution/overflow routing Employee turnover

16 Module Objective 2: Analyze some VoIP Applications such as Packet Telephony Call Centers and Service Provider Calling Card Application

17 CSCC Challenges (Contd.)  Seasonal staffing needs  Inconvenient busy hours  Regional call-center talent - The CSCC technology addresses some of the challenges above but Packet based call centers is the solution

18 CSCC Setup

19 Packet Telephony Based Call Centers

20 Click to talk

21 Service Provider Calling Card Case Study The following list breaks down the call-flow of a pre- or post-paid call through an IP network:  Subscriber dials local phone number of service provider (call leg A).  Subscriber gets a second dial tone and is prompted to enter the destination phone number, account number, and password if calling away from home.  Call is completed to the destination phone (call legs B and C).

22 Calling Card Study

23 Calling Card Component Details

24 Calling Card Call Flow 1.The subscriber calls the local access number for the gateway. 2.The gateway queries the RADIUS server with the Automatic Number ID (ANI, or caller ID) of the caller. 3.The RADIUS server looks up the ANI to verify that the caller is a subscriber and then sends a message to the gateway to authenticate the user.

25 Calling Card Call Flow(1)

26 Calling Card Call Flow (2) 4.The user enters the destination phone number. 5.The gateway consults the gatekeeper on ways he can route the call. 6.The gatekeeper looks up the E.164 address against a table and sends the gateway the IP address of the destination gateway.

27 Calling card Call Flow(3)

28 Calling card call flow (4) 7.The originating gateway places an H.323 call across the IP network to the destination gateway. 8.The destination gateway places a PSTN call to the destination phone. 9.The gateways send start/stop records to the RADIUS server for billing.

29 Calling Card Call flow (5)

30 ACME Case Study (1)

31 ACME – Voice and Data Network

32 ACME Case Study (Present Network)

33 ACME Converged Network

34 ACME Cost Savings  Phones use DHCP and keep phone numbers regardless of physical location.  Cabling to the desktop is easier (everything is Ethernet).  Call appearance remains the same whether the user is at home or at work. This enables fully transparent telecommuting.  The call-processing engine is now on a standard platform, which provides the enterprise network with greater flexibility.  Cisco's Call Manager is actually configurable through Hypertext Markup Language (HTML), which simplifies administrative overhead as well as PBX administration.  Cisco's Call Manager can also support other standards-based interfaces such as Station Message Desk Interface (SMDI) for an interface into a legacy PBX. As an example, you can use this interface to illuminate the message-waiting light.

35 Reading  Chapter 6 of textbook


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