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  • 1. Voice Over Internet Protocol (VoIP) CSC550 Term Project Group Members: Alice Miller, Bill Smith and Cathy Davis Advisor: Dr. Frank Lee 10/8/2005
  • 2. OUTLINE
    • INTRODUCTION
    • ADVANTAGES OF VoIP
    • POPULAR VoIP PROTOCOLS
      • H.323
      • SIP
      • MGCP
    • SUPPORTING PROTOCOLS
    • TECHNICAL ISSUES
    • HARWARE REQUIREMENTS
    • SOFTWARE REQUIREMENTS
    • PRODUCTS
    • SERVICES
    • FUTURE DEVELOPMENTS
    • CONCLUSION
  • 3. INTRODUCTION
    • VoIP - The ability to carry toll quality voice using compression techniques and packet switching over the IP packet network.
    Voice CODEC: Analog to Digital Compress Create Voice Datagram Add Header (RTP, UDP, IP etc) CODEC: Digital to Analog Decompress Re-Sequence and Buffer-Delay Process Header Voice analog analog digital digital
  • 4. INTRODUCTION (contd)
    • Real time voice traffic can be carried over IP networks in three different ways
      • PC to PC
      • PC to Phone
      • Phone to Phone
  • 5. INTRODUCTION (contd)
    • Protocols commonly implemented by Voice over IP
      • H.323
      • SIP ( Session Initiation Protocol)
      • MGCP (Media Gateway Control Protocol)
      • RSVP (Resource Reservation Protocol)
  • 6. ADVANTAGES OF VoIP
    • INTEGRATION OF VOICE AND DATA: Web servers capable of interacting with voice, data and images.
    • SIMPLIFICATION: Allows more standardization and less equipment management.
    • NETWORK EFFICIENCY: Provides bandwidth consolidation.
    • COST REDUCTION: Slashes high charges for long distance calls.
    • ADVANCED APPLICATIONS: To be derived from multimedia and multi-service applications.
  • 7. H.323
    • H.323 is a standard that specifies the components, protocols and procedures that provide multimedia communication services such as real-time audio, video, and data communications over packet networks, including Internet Protocol (IP) based networks.
  • 8. COMPONENTS OF H.323
      • TERMINALS : Can be either a personal computer or a stand-alone device
      • GATEWAYS : A H.323 gateway provides connectivity between an H.323 network and a non-H.323 network.
      • GATEKEEPERS : Provide call control services such as address translation, bandwidth management, admission control and zone management.
      • MULTIPOINT CONTROL UNITS (MCU) : Manage conference resources, negotiate between terminals for the purpose of determining the audio or video coder/decoder to use, and may handle the media stream.
  • 9. Layout of H.323-enabled inter-network
  • 10. H.323 PROTOCOL ARCHITECTURE
    • An integrated set of software programs that follows the ITU (Intl Telecomm Union) H.323 recommendation and all associated recommendations.
    • CALL CONTROL LAYER
      • Signaling for call setup and capability exchange
      • Signaling of commands, indications and messages to open
      • Describes the content of logical channels.
      • Formats the data streams into messages for output
      • Performs logical framing, sequence numbering, and error detection and correction for each media type.
  • 11. H.323 PROTOCOL ARCHITECTURE (contd)
    • CALL SIGNALING
      • The H.225 standard defines a layer that formats the transmitted video, audio, data, and control streams for output to the network, and retrieves the corresponding streams from the network.
      • Q.931 resides within H.223 and it is a link layer protocol for establishing connections and framing data.
  • 12. H.323 PROTOCOL ARCHITECTURE (contd)
    • REGISTRATION, ADMISSION, AND STATUS
      • The H.225 also includes registration, admission, and status (RAS) control.
      • RAS is the protocol between endpoints and gatekeepers that makes connections available between them.
    • CONTROL SIGNALING
      • The H.245 standard provides the call control mechanism that allows H.323-compatible terminals to connect to each other.
      • The control messages that it carries relate to: Opening and closing of logical channels used to carry media streams, preference requests, flow-control messages and general commands and indications.
  • 13. H.323 Protocol Stack
  • 14. SIP
    • The Session Initiation Protocol (SIP) is an application layer signaling protocol that defines initiation, modification and termination of interactive multimedia communication sessions between users.
    • It was developed by the IETF and is explained in RFC 2453. It was approved in early 1999.
    Physical I P U D P T C P S I P S D P RTP/RTCP CODEC D N S Signaling Media Utility
  • 15. HOW DOES SIP MAKE A CALL?
    • User Registering and Location - determination of the end system to be used for communication
    • User Availability - determination of the willingness of the called party to engage in communications
    • User Capabilities - determination of the media and media parameters to be used
    • Call Setup - ringing and establishing call parameters at both called and calling party
    • Call Modification change of media, call forward etc
    • Call Handling - the transfer and termination of calls
  • 16. SIP ARCHITECTURE Intelligent SIP User Agents (UAC/UAS) Registrar Redirect Location Proxy Server REGISTER Here I am INVITE I want to talk to another UA Proxied INVITE Ill handle it for you Where is this name/phone#? 3xx Redirection They moved, try this address SIP Gateway SIP-GW IP Network PSTN SIP Servers sip:14083831088@vovida.org sip:hostname@192.168.10.1
  • 17. SIP Client SIP Redirect Server SIP Client (User Agent Server) Location Server 2 1 RTP Media 4 5 3 11 7 6 2. bob 3. play.com 4. Bob moved. Temporarily contact [email_address] 5. ACK 6. INVITE bob@play.com 7. Ringing ok 1. INVITE bob@ieee.org SIP OPERATION IN REDIRECT MODE 8. ACK (ieee.org) (sjsu.edu) (play.com) 8
  • 18. SIP Client SIP Redirect & Location Servers SIP Client (User Agent Server) SIP Proxy 2. INVITE bob@ieee.org 3 1 RTP Media SIP Proxy 2 5 6 4 9 12 11 7 10 3. bob 4. play.com 5. Bob moved. Temporarily contact bob@play.com 6. ACK 7. INVITE bob@play.com 12. ACK 9. Ringing ok 1. INVITE bob@ieee.org 8. INVITE bob@play.com 10. Ringing ok SIP OPERATION IN PROXY MODE 8 11. ACK (Ieee.org) (sjsu.edu) (play.com)
  • 19. WHY WAS SIP DESIGNED?
    • Flexibility Does not dictate specifics for architecture, messaging etc. Can even use H.323 URLs to route call.
    • Scalability and Simplicity Based on internet model and not single LAN segments. Less storage required.
    • Ease of creation of new services like buddy lists, instant messaging etc.- Integrating multimedia communications with ease (web-based, email routing mechanisms etc.)
    • Mobility (Location/Redirect Servers)
    • Call redirection/forking/multiparty calls ..
  • 20. COMPARISON OF H.323 and SIP Uses MCU for users > 3. Fn. overlaps with RTCP Multicasting. No restrictions on no. of users Conferencing With Gatekeeper Registration with Registrar Security Complex and Monolithic Simple and Modular Complexity/Struct H.245 SDP Session Description Peer-to-peer Peer-to-peer Relationship Intelligent H.323 Terminals Intelligent User Agents Client UDP and TCP for signaling, RTP for Media UDP and TCP for signaling, RTP for Media Signaling and Media H.323 ID Alias Address mapping mechanism in Gatekeeper SIP URL ID Redirect or location servers Endpoint Addressing and Call Routing Binary Based Text based Control channel T