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  1. 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. 2. OUTLINE <ul><li>INTRODUCTION </li></ul><ul><li>ADVANTAGES OF VoIP </li></ul><ul><li>POPULAR VoIP PROTOCOLS </li></ul><ul><ul><li>H.323 </li></ul></ul><ul><ul><li>SIP </li></ul></ul><ul><ul><li>MGCP </li></ul></ul><ul><li>SUPPORTING PROTOCOLS </li></ul><ul><li>TECHNICAL ISSUES </li></ul><ul><li>HARWARE REQUIREMENTS </li></ul><ul><li>SOFTWARE REQUIREMENTS </li></ul><ul><li>PRODUCTS </li></ul><ul><li>SERVICES </li></ul><ul><li>FUTURE DEVELOPMENTS </li></ul><ul><li>CONCLUSION </li></ul>
  3. 3. INTRODUCTION <ul><li>VoIP - The ability to carry toll quality voice using compression techniques and packet switching over the IP packet network. </li></ul>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. 4. INTRODUCTION (cont’d…) <ul><li>Real time voice traffic can be carried over IP networks in three different ways </li></ul><ul><ul><li>PC to PC </li></ul></ul><ul><ul><li>PC to Phone </li></ul></ul><ul><ul><li>Phone to Phone </li></ul></ul>
  5. 5. INTRODUCTION (cont’d…) <ul><li>Protocols commonly implemented by Voice over IP </li></ul><ul><ul><li>H.323 </li></ul></ul><ul><ul><li>SIP ( Session Initiation Protocol) </li></ul></ul><ul><ul><li>MGCP (Media Gateway Control Protocol) </li></ul></ul><ul><ul><li>RSVP (Resource Reservation Protocol) </li></ul></ul>
  6. 6. ADVANTAGES OF VoIP <ul><li>INTEGRATION OF VOICE AND DATA: Web servers capable of interacting with voice, data and images. </li></ul><ul><li>SIMPLIFICATION: Allows more standardization and less equipment management. </li></ul><ul><li>NETWORK EFFICIENCY: Provides bandwidth consolidation. </li></ul><ul><li>COST REDUCTION: Slashes high charges for long distance calls. </li></ul><ul><li>ADVANCED APPLICATIONS: To be derived from multimedia and multi-service applications. </li></ul>
  7. 7. H.323 <ul><li>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. </li></ul>
  8. 8. COMPONENTS OF H.323 <ul><ul><li>TERMINALS : Can be either a personal computer or a stand-alone device </li></ul></ul><ul><ul><li>GATEWAYS : A H.323 gateway provides connectivity between an H.323 network and a non-H.323 network. </li></ul></ul><ul><ul><li>GATEKEEPERS : Provide call control services such as address translation, bandwidth management, admission control and zone management. </li></ul></ul><ul><ul><li>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. </li></ul></ul>
  9. 9. Layout of H.323-enabled inter-network
  10. 10. H.323 PROTOCOL ARCHITECTURE <ul><li>An integrated set of software programs that follows the ITU (Int’l Telecomm Union) H.323 recommendation and all associated recommendations. </li></ul><ul><li>CALL CONTROL LAYER </li></ul><ul><ul><li>Signaling for call setup and capability exchange </li></ul></ul><ul><ul><li>Signaling of commands, indications and messages to open </li></ul></ul><ul><ul><li>Describes the content of logical channels. </li></ul></ul><ul><ul><li>Formats the data streams into messages for output </li></ul></ul><ul><ul><li>Performs logical framing, sequence numbering, and error detection and correction for each media type. </li></ul></ul>
  11. 11. H.323 PROTOCOL ARCHITECTURE (cont’d) <ul><li>CALL SIGNALING </li></ul><ul><ul><li>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. </li></ul></ul><ul><ul><li>Q.931 resides within H.223 and it is a link layer protocol for establishing connections and framing data. </li></ul></ul>
  12. 12. H.323 PROTOCOL ARCHITECTURE (cont’d) <ul><li>REGISTRATION, ADMISSION, AND STATUS </li></ul><ul><ul><li>The H.225 also includes registration, admission, and status (RAS) control. </li></ul></ul><ul><ul><li>RAS is the protocol between endpoints and gatekeepers that makes connections available between them. </li></ul></ul><ul><li>CONTROL SIGNALING </li></ul><ul><ul><li>The H.245 standard provides the call control mechanism that allows H.323-compatible terminals to connect to each other. </li></ul></ul><ul><ul><li>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. </li></ul></ul>
  13. 13. H.323 Protocol Stack
  14. 14. SIP <ul><li>The Session Initiation Protocol (SIP) is an application layer signaling protocol that defines initiation, modification and termination of interactive multimedia communication sessions between users. </li></ul><ul><li>It was developed by the IETF and is explained in RFC 2453. It was approved in early 1999. </li></ul>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. 15. HOW DOES SIP MAKE A CALL? <ul><li>User Registering and Location - determination of the end system to be used for communication </li></ul><ul><li>User Availability - determination of the willingness of the called party to engage in communications </li></ul><ul><li>User Capabilities - determination of the media and media parameters to be used </li></ul><ul><li>Call Setup - ringing and establishing call parameters at both called and calling party </li></ul><ul><li>Call Modification – change of media, call forward etc </li></ul><ul><li>Call Handling - the transfer and termination of calls </li></ul>
  16. 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 “ I’ll 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:hostname@
  17. 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. 4. Bob moved. Temporarily contact [email_address] 5. ACK 6. INVITE 7. Ringing ok 1. INVITE SIP OPERATION IN REDIRECT MODE 8. ACK ( ( ( 8
  18. 18. SIP Client SIP Redirect & Location Servers SIP Client (User Agent Server) SIP Proxy 2. INVITE 3 1 RTP Media SIP Proxy 2 5 6 4 9 12 11 7 10 3. bob 4. 5. Bob moved. Temporarily contact 6. ACK 7. INVITE 12. ACK 9. Ringing ok 1. INVITE 8. INVITE 10. Ringing ok SIP OPERATION IN PROXY MODE 8 11. ACK ( ( (
  19. 19. WHY WAS SIP DESIGNED? <ul><li>Flexibility – Does not dictate specifics for architecture, messaging etc. Can even use H.323 URLs to route call. </li></ul><ul><li>Scalability and Simplicity – Based on internet model and not single LAN segments. Less storage required. </li></ul><ul><li>Ease of creation of new services like buddy lists, instant messaging etc.- Integrating multimedia communications with ease (web-based, email routing mechanisms etc.) </li></ul><ul><li>Mobility (Location/Redirect Servers) </li></ul><ul><li>Call redirection/forking/multiparty calls .. </li></ul>
  20. 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 Telephony model Internet/WWW model Origin ITU IETF Standards Body H.323 SIP VoIP Protocol
  21. 21. MGCP <ul><li>Media Gateway Control Protocol is a master-slave protocol that defines communication between telephony Gateways and external call control elements called Media Gateway Controllers or Call Agents. </li></ul><ul><li>It was developed by the IETF and explained in RFC 2705. It assumes limited intelligence at endpoints and concentrates it in the core of the network. </li></ul><ul><li>Call Agent (master) </li></ul><ul><ul><li>provides call signaling, control and processing intelligence to the Gateway </li></ul></ul><ul><ul><li>sends and receives commands to/from Gateway </li></ul></ul><ul><li>Gateway (slave) </li></ul><ul><ul><li>provides translations between packet and circuit switched networks </li></ul></ul><ul><ul><li>sends notification to the call agent about endpoint events. </li></ul></ul>
  23. 23. SUPPORTING PROTOCOLS RSVP RTCP RTP SAP/SDP H.323 SIP UDP TCP Underlying Physical, Data Link and Network Layers
  24. 24. SUPPORTING PROTOCOLS (cont’d) <ul><li>RTP/RTCP (Real-Time Transport & Control Protocols) is used for transporting real time data </li></ul><ul><li>RSVP (Resource Reservation Protocol) for reserving resources </li></ul><ul><li>RTSP (Real-Time Streaming Protocol) for controlling delivery of real-time media streams </li></ul><ul><li>SDP (Session Description Protocol) for advertising multimedia sessions </li></ul><ul><li>SAP (Session Announcement Protocol) for describing multimedia session </li></ul>
  25. 25. TECHNICAL ISSUES <ul><li>Quality of service </li></ul><ul><ul><li>Delay, jitter, congestion, echo, packet loss, mis-ordered packet arrival </li></ul></ul><ul><li>Measure of QoS </li></ul><ul><ul><li>The mean opinion score is widely used </li></ul></ul><ul><ul><li>Algorithms: PSQM, PAMS and PESQ </li></ul></ul><ul><li>Bandwidth consumption </li></ul><ul><ul><li>A quality call requires at least 64 kbps. It is impossible to dedicate so much for voice on data network </li></ul></ul><ul><ul><li>Speech compression techniques are used. For example, silence compression which brings down the bandwidth to 5-6 kbps </li></ul></ul>
  26. 26. TECHNICAL ISSUES (cont’d) <ul><li>Transparency to the user </li></ul><ul><ul><li>ease of configuration </li></ul></ul><ul><ul><li>mapping between IP addresses and phone numbers </li></ul></ul><ul><li>Security </li></ul><ul><ul><li>provides for secure environment using TCP/IP </li></ul></ul><ul><ul><li>access control can be implemented using authentication </li></ul></ul><ul><ul><li>calls can be made private using encryption </li></ul></ul><ul><li>Security features use four primary components </li></ul><ul><ul><li>packet filtering router </li></ul></ul><ul><ul><li>connection gateway </li></ul></ul><ul><ul><li>address translating firewall </li></ul></ul><ul><ul><li>application proxy </li></ul></ul>
  27. 27. HARDWARE REQUIREMENTS <ul><li>Minimum Requirements </li></ul><ul><ul><li>PC 386 or higher </li></ul></ul><ul><ul><li>Sound card </li></ul></ul><ul><ul><li>Full duplex capability </li></ul></ul><ul><ul><li>Network card or connection to internet or other kind of interface to allow communication between 2 PCs </li></ul></ul><ul><li>Companies offering hardware </li></ul><ul><ul><li>Quicknet, Lucent, 3COM, Cisco, Nortel, Alcatel </li></ul></ul><ul><li>Hardware accelerating cards </li></ul><ul><ul><li>Quicknet PhoneJack </li></ul></ul><ul><ul><li>Quicknet LineJack </li></ul></ul><ul><ul><li>VoiceTronix V4PCI </li></ul></ul><ul><ul><li>VoiceTronix VPB4 </li></ul></ul><ul><ul><li>VoiceTronix VPB8L </li></ul></ul>
  28. 28. SOFTWARE REQUIREMENTS <ul><li>Operating Systems </li></ul><ul><ul><li>Windows 95, 98, 2000, ME and XP </li></ul></ul><ul><ul><li>Linux </li></ul></ul><ul><li>Gateway </li></ul><ul><ul><li>Internet Switch Board </li></ul></ul><ul><ul><li>PSTNGW (Packet Switching Transfer Network Gateway) </li></ul></ul><ul><li>Gatekeeper </li></ul>
  29. 29. PRODUCTS <ul><li>Gateways: MICOM V/IP Gateway, Nortel Networks CVX SS7 Gateway, Lucent Technologies Pathstar Access Server, Cisco Systems DE-30+ Gateway, 3Com Gateway, VocalTec Series 2000 Gateway, Nuera Solutions Access plus F200 IP </li></ul><ul><li>Gatekeepers: Eriksson H.323 gatekeeper, VocalTec Gatekeeper, Nortel Netwroks’ IPConnect, Elemedia H.323 gatekeeper GK2000S </li></ul>
  30. 30. SERVICES <ul><li>IP telephones: Cisco's IP phones, Selsius IP phones, Nokia Systems’ IPCourier </li></ul><ul><li>PC based software phones: VocalTec IPhone, Netscape’s CoolTalk, Microsoft NetMeeting, WhitePine’s CU-SeeME Pro </li></ul>
  31. 31. FUTURE DEVELOPMENTS <ul><li>Directory services over telephones </li></ul><ul><li>Inter office trunking over the corporate intranet </li></ul><ul><li>Remote access to voice, data and fax services of office from home </li></ul><ul><li>Fax over IP </li></ul><ul><li>Conference bridging </li></ul><ul><li>Voice/data synchronization </li></ul><ul><li>Text to speech conversion </li></ul>
  33. 33. CONCLUSION <ul><li>VoIP sends voice over data networks instead of data over voice network </li></ul><ul><li>Internet along with TCP/IP are driving forces for VoIP technology </li></ul><ul><li>Ideal for computer based communications </li></ul><ul><li>Market for VoIP is established and is rapidly growing </li></ul><ul><li>VoIP cuts communication costs and improves efficiency </li></ul><ul><li>Needs QoS for acceptable quality </li></ul>
  34. 34. REFERENCES <ul><li> </li></ul><ul><li> </li></ul><ul><li> </li></ul><ul><li> </li></ul><ul><li>SIP Understanding the Session Initiation Protocol by Alan B. Johnston </li></ul>
  35. 35. <ul><li>Q & A </li></ul>
  36. 36. Thank You!