VOIP基础知识培训资料

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  • Q&A
  • VOIP基础知识培训资料

    1. 1. Voice over IP (VoIP) Brian Gracely Technical Marketing Engineer
    2. 2. Agenda <ul><li>Why VoIP? </li></ul><ul><li>Comparing & Understanding the VoIP Protocols - H.323 - Skinny - MGCP - SIP </li></ul><ul><li>SIP Tutorial </li></ul><ul><li>Sample VoIP Applications </li></ul><ul><li>Cisco VoIP products </li></ul>
    3. 3. Why VoIP? The Interesting Stuff <ul><li>Telecommunications Act of 1996 - Deregulation of the Bell networks - Open the competitive markets for Service Providers </li></ul><ul><li>Converged Networks - Voice, Video & Data over an IP network - Reduced the costs of managing parallel networks - Allows voice to be an IP “application” </li></ul><ul><li>Centralized or distributed architectures - Add features where they are needed </li></ul>
    4. 4. Why VoIP? The Challenging Stuff <ul><li>Do we need to replicate all the existing PSTN / PBX features? </li></ul><ul><li>What’s the right architecture? - Centralized - Distributed - Mix of both </li></ul><ul><li>How do we? - Provide better than PSTN QoS - Provide Admission Control - Secure the signaling & media - Meet all the regulatory requirements </li></ul>
    5. 5. Open Packet Telephony TDM/ Circuit Switch Digital Trunk Subsystem Line Concentration Administration Maintenance Billing Call Control Connection Control Features Common Channel Signaling Complex Standards-Based Packet Infrastructure Layer (IP, ATM) Open Call Control Layer (SIP, H.323, MGCP, etc.) Open Service Application Layer (JAIN, AIN, TAPI, JTAPI, XML etc.) Open/Standard Interface Open/Standard Interface Switching Network
    6. 6. AVVID Architecture - Open Packet Telephony The World Is Now Global— All Apps Must Travel Time and Distance Applications Call Processing Infrastructure Clients IP SoftPhone <ul><li>PSTN gateways </li></ul><ul><li>Analog phone support </li></ul><ul><li>DSP farms </li></ul>IP Network PSTN Directory Call Processing Cisco Unity Voice Mail, UMS Intelligent Contact Manager IP IVR, IP AA Apps Engine Voice Portal ICM Collaboration Video GK
    7. 7. Agenda <ul><li>Why VoIP? </li></ul><ul><li>Comparing & Understanding the VoIP Protocols - H.323 - Skinny - MGCP - SIP </li></ul><ul><li>SIP Tutorial </li></ul><ul><li>Sample VoIP Applications </li></ul><ul><li>Cisco VoIP products </li></ul>
    8. 8. VoIP Signaling Protocols <ul><li>H.323 - ITU standard, ISDN-based, distributed topology - 90%+ of all Service Provider VoIP networks - The current interconnect for CallManager to Service Providers - Useful for video applications </li></ul><ul><li>Skinny - Centralized Call-Control architecture. - CallManager controls all features. - over 700,000 IP Phones deployed </li></ul><ul><li>MGCP - IETF RFC2705 - Centralized Call-Control Architecture - Call-Agents (MGC) & Gateways (MG) </li></ul><ul><li>SIP - IETF RFC2543 - Distributed Call-Control - Used for more than VoIP…SIMPLE: Instant Messaging / Presence </li></ul>
    9. 9. Basic H.323 Call Gatekeeper A Gatekeeper B RRQ/RCF ARQ RRQ/RCF LRQ IP Network Phone A Gateway A Gateway B H.225 (Q.931) Setup H.225 (Q.931) Alert and Connect H.245 RTP ACF LCF ARQ ACF Phone B V V
    10. 10. Basic Skinny Call PSTN Cisco CallManager IP WAN Voice Mail Server Call Setup E.164 Lookup Ring Off Hook RTP Stream Ring Back H.323/MGCP Gateway
    11. 11. MGCP Architectures & Mixed Protocols PSTN BTS / VSC SS7 PSTN Gateway SIP or H.323 Network Access Gateway SCP MGCP SIP H.323 IMT PRI RTP SIP / H.323 GK P S T N V V V
    12. 12. Agenda <ul><li>Why VoIP? How does it work & why is it interesting? </li></ul><ul><li>Comparing & Understanding the VoIP Protocols - H.323 - Skinny - MGCP - SIP </li></ul><ul><li>SIP Tutorial </li></ul><ul><li>Sample VoIP Applications </li></ul><ul><li>Cisco VoIP products </li></ul>
    13. 13. Why are we talking about SIP? <ul><li>Cisco has never met a protocol it didn’t like…. - Customers haven’t chosen 1 protocol to define VoIP </li></ul><ul><li>SIP is a very Internet friendly protocol, and Cisco likes Internet friendly stuff…. - SIP reuses a lot of Internet protocols & formatting </li></ul><ul><li>Customers still weary about proprietary protocols…. - Skinny works well, but it is proprietary </li></ul><ul><li>It’s about the Applications!! - The next “Killer App” is the integration of voice, data, video, IM & Presence… SIP can do this. </li></ul><ul><li>Microsoft!! 250 millions desktops might speak SIP soon…. - SIP client will be added to WindowsXP in October </li></ul>
    14. 14. The history of SIP <ul><li>S ession I nitiation P rotocol (SIP) is defined via RFC2543 on March 17, 1999. </li></ul><ul><li>Additional “feature” drafts have been written to address issues which concern SS7/ISUP handling, QoS, Alerting, DHCP, 3PCC, Firewalls & NAT, etc… </li></ul><ul><li>IETF SIP-WG created in September, 1999 </li></ul><ul><li>RFC2543bis (additions) created in April 2000. </li></ul><ul><li>Vendor interoperability testing done at the semi-annual SIP Bakeoff (8th in August in UK) </li></ul>
    15. 15. The various flavors of SIP <ul><li>RFC2543 - “vanilla” SIP - the most commonly deployed & developed by commercial vendors </li></ul><ul><li>SIP-T - inter Call Agent (MGC) protocol for carrying SS7 / ISUP messaging - basically maps ISUP messaging to a MIME attachment </li></ul><ul><li>SIP extension from PacketCable - additions to Security, QoS & Privacy areas </li></ul>
    16. 16. SIP Basics - Architecture Legacy PBX SIP User Agents (UA) Application Services eMail LDAP Oracle XML SIP SIP RTP (Media) SIP CPL CPL 3pcc PSTN CAS or PRI I NTELL I GENT SERV I CES SIP Proxy, Registrar & Redirect Servers
    17. 17. SIP Basics - Architectural Elements <ul><li>Clients: SIP Phones, Softphones, Gateways, Media Gateway Controllers, PDAs, Robots - User Agent Client (UAC) / User Agent Server (UAS) - Originate & Terminate SIP requests </li></ul><ul><li>Typically an endpoint will have both UAC & UAS, UAC for originating requests, and UAS for terminating requests </li></ul><ul><li>Servers: - Proxy Server - Redirect Server - Registrar Server </li></ul>
    18. 18. SIP Servers/Services (cont) SIP User Agents Registrar Redirect LocationDatabase SIP Proxy SIP Servers/ Services 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 User Agents SIP-GW
    19. 19. SIP Methods <ul><li>Consists of Requests and Responses </li></ul><ul><li>Requests (unless mentioned, each has a response) • REGISTER: UA registers with Registrar Server • INVITE: request from a UAC to initiate a session • ACK: confirms receipt of a final response to INVITE • BYE: sent by either side to end a call • CANCEL: sent to end a call not yet connected • OPTIONS: sent to query capabilities outside of SDP </li></ul><ul><li>Newly Adopted Methods: • SUBSCRIBE & NOTIFY: used to identify device status / presence. The foundation of SIP IM / Presence (IMPP). • INFO: a means of carrying “data” in a message body • REFER: the mechanism to initiate a Transfer • MESSAGE: the means of carrying “data” for SIP IMPP </li></ul><ul><li>Messages contain SIP Headers and Body. Body might be SDP or an attachment or some other application </li></ul>
    20. 20. SIP Addressing <ul><li>Modeled after mailto URLs. May be a combination of FQDNs or E.164 numbers or both. </li></ul><ul><li>Support for Fully-Qualified Domain Names (FQDNs) using sip: URLs - sip: “John Doe” <jdoe@cisco.com> </li></ul><ul><li>Support for E.164 addresses - sip:14085551234@gateway.com; user=phone </li></ul><ul><li>Support for mixed addresses - sip:14085551234@10.1.1.1; user=phone sip:jdoe@10.1.1.1 </li></ul><ul><li>Support for E.164 addresses using tel: URLs - tel:14085551234 </li></ul>
    21. 21. Basic SIP Call-Flow SIP UA1 SIP UA2 INVITE w/ SDP for Media Negotiation 100 Trying 180/183 Ringing w/ SDP for Media Negotiation 200 OK 200 OK BYE MEDIA MEDIA ACK
    22. 22. Basic SIP Functionality - Call Forking LOCAL PSTN Proxy / Redirect Server Location Database INVITE sip:1-800-GO-CISCO@cisco.com “ Where is sip:1-800-GO-CISCO@cisco.com?” “ Contact 1234@10.1.1.1, 1234@10.1.1.2 and 1234@10.1.1.3” INVITE sip:1234@10.1.1.1 INVITE sip:1234@10.1.1.2 INVITE sip:1234@10.1.1.3 Forked Calls can be in parallel or sequential. The first phone to answer will get the call, the others will get a CANCEL from the Proxy Server.
    23. 23. Basic SIP Functionality - Call Redirection LOCAL PSTN Proxy / Redirect Server Location Database 392-1234 INVITE sip:3921234@cisco.com “ Where is sip:3921234@cisco.com?” “ You need to contact 4721111” 3xx Moved Contact: sip:4721111@10.1.1.3 INVITE sip:4721111@10.1.1.3 National PSTN The user at 392-1234 informed the network that he could be reached on his cell-phone at 472-1111
    24. 24. 3rd-Party Call-Control (3pcc) & Back-to-Back UserAgent (B2BUA) LOCAL PSTN SIP Controller - 3pcc Application INVITE sip:1234 w/o SDP x1234 18x / 200 OK w/ SDP INVITE sip:9194721111 w/ SDP of SIP Phone 18x / 200 OK w/ SDP ACK w/ SDP of SIP Gateway A user could manage their communications via a webpage. The webpage would invoke the SIP 3PCC application to create SIP sessions to all parties involved. HTTP post
    25. 25. Agenda <ul><li>Why VoIP? How does it work & why is it interesting? </li></ul><ul><li>Comparing & Understanding the VoIP Protocols - H.323 - Skinny - MGCP - SIP </li></ul><ul><li>SIP Tutorial </li></ul><ul><li>Sample VoIP Applications </li></ul><ul><li>Cisco VoIP products </li></ul>
    26. 26. <ul><li>IP IVR </li></ul><ul><ul><li>Voice Portal </li></ul></ul><ul><li>Auto Attendant </li></ul>Application Engine Architecture Web Pages Enterprise Database Application Toolkit External Services Packaged Solutions Telephony Directory Access Web Access DB Access LDAP Notification Server Queuing Paging E-Mail VXML services ICM Notification Services Queuing (ACD) Personalized Apps Customer Apps Unity
    27. 27. <ul><li>IP Telephony Appliance </li></ul><ul><li>- Corporate directory integration via LDAP </li></ul><ul><li>- Web site integration via XML </li></ul><ul><li>- Personalized menu’s via softkeys </li></ul><ul><li>Extensible interface with IP services offers clear differentiation to PBX connected devices </li></ul>IP Phone Display Applications *
    28. 28. Convergence:Presence Services Managing your communications through web browsers, Instant Messaging and mobile devices
    29. 29. Informal Agent Queuing (IAQ) Remote Agents SoftPhone IP Phones PSTN IP Central Site IAQ Server Branch Agents Distribution Groups with Queuing for Resources 2 Types of Queues Requestor Servicer
    30. 30. Web Attendant <ul><li>Ubiquitous access via a browser </li></ul><ul><li>Extension look-up via LDAP </li></ul><ul><li>Easy of use with drag and drop interface </li></ul><ul><li>Benefits: </li></ul><ul><ul><li>Eliminates specialized receptionist phones </li></ul></ul><ul><ul><li>Access via URL </li></ul></ul><ul><li>Included with Call Manager 3.0(tbd) </li></ul>
    31. 31. Voice Portal Solution <ul><li>Extracts XML information from web page into IP IVR </li></ul><ul><li>Benefit </li></ul><ul><ul><li>Only one place to configure and maintain data </li></ul></ul><ul><ul><li>Consistency </li></ul></ul><ul><ul><li>Lower admin costs </li></ul></ul>IP Intranet Press #1 to Hear Stock Quote IP IVR Stock Quote *
    32. 32. VoiceXML PSTN Cisco Voice Gateway RTSP Server VoiceXML in IOS: HTTP Server Architectural Model: VXML Interpreter Context Document Server Implementation Platform VXML Interpreter
    33. 33. Agenda <ul><li>Why VoIP? How does it work & why is it interesting? </li></ul><ul><li>Comparing & Understanding the VoIP Protocols - H.323 - Skinny - MGCP - SIP </li></ul><ul><li>SIP Tutorial </li></ul><ul><li>VoIP Applications </li></ul><ul><li>Cisco VoIP products </li></ul>
    34. 34. Cisco VoIP Products <ul><li>Call-Processing - Cisco CallManager - Multimedia Conference Mgr - H.323 Gatekeeper / Proxy - Cisco SIP Proxy Server (CSPS) - BTS10200 Softswitch - VSC3000 Softswitch </li></ul><ul><li>VoIP Gateways - Low End: ATA 186, 827v4, CVA122, uBR924, 1750, VG200 - Mid Range: 3810, 2421, 2600, 3600, Cat4000, AS5300, 7200, 7500 - High End: AS5350, AS5400, Cat6000, AS5850, MGX8850 </li></ul><ul><li>IP Phones - 7910, 7940, 7960, 7935, Softphone </li></ul><ul><li>Applications - Unity UM, Personal Assistant, Conference Connection, IP IVR, IP Contact Center, Web Attendant, XML / BTXML on IP Phones - 80+ EcoSystem partners </li></ul><ul><li>Cisco Infrastructure - IOS QoS features, Line-Powered Catalyst Switches, Catalyst QoS features - Application Layer Gateway (ALG) in IOS-NAT / Firewall, PIX </li></ul>
    35. 35. Questions?
    36. 36. Voice over IP (VoIP) Brian Gracely - [email_address]
    37. 37. Presentation_ID © 2001, Cisco Systems, Inc. All rights reserved.

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