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VOIP基础知识培训资料
 

VOIP基础知识培训资料

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VOIP基础知识培训资料 VOIP基础知识培训资料 Presentation Transcript

  • Voice over IP (VoIP) Brian Gracely Technical Marketing Engineer
  • Agenda
    • Why VoIP?
    • Comparing & Understanding the VoIP Protocols - H.323 - Skinny - MGCP - SIP
    • SIP Tutorial
    • Sample VoIP Applications
    • Cisco VoIP products
  • Why VoIP? The Interesting Stuff
    • Telecommunications Act of 1996 - Deregulation of the Bell networks - Open the competitive markets for Service Providers
    • Converged Networks - Voice, Video & Data over an IP network - Reduced the costs of managing parallel networks - Allows voice to be an IP “application”
    • Centralized or distributed architectures - Add features where they are needed
  • Why VoIP? The Challenging Stuff
    • Do we need to replicate all the existing PSTN / PBX features?
    • What’s the right architecture? - Centralized - Distributed - Mix of both
    • How do we? - Provide better than PSTN QoS - Provide Admission Control - Secure the signaling & media - Meet all the regulatory requirements
  • 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
  • AVVID Architecture - Open Packet Telephony The World Is Now Global— All Apps Must Travel Time and Distance Applications Call Processing Infrastructure Clients IP SoftPhone
    • PSTN gateways
    • Analog phone support
    • DSP farms
    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
  • Agenda
    • Why VoIP?
    • Comparing & Understanding the VoIP Protocols - H.323 - Skinny - MGCP - SIP
    • SIP Tutorial
    • Sample VoIP Applications
    • Cisco VoIP products
  • VoIP Signaling Protocols
    • 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
    • Skinny - Centralized Call-Control architecture. - CallManager controls all features. - over 700,000 IP Phones deployed
    • MGCP - IETF RFC2705 - Centralized Call-Control Architecture - Call-Agents (MGC) & Gateways (MG)
    • SIP - IETF RFC2543 - Distributed Call-Control - Used for more than VoIP…SIMPLE: Instant Messaging / Presence
  • 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
  • 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
  • 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
  • Agenda
    • Why VoIP? How does it work & why is it interesting?
    • Comparing & Understanding the VoIP Protocols - H.323 - Skinny - MGCP - SIP
    • SIP Tutorial
    • Sample VoIP Applications
    • Cisco VoIP products
  • Why are we talking about SIP?
    • Cisco has never met a protocol it didn’t like…. - Customers haven’t chosen 1 protocol to define VoIP
    • SIP is a very Internet friendly protocol, and Cisco likes Internet friendly stuff…. - SIP reuses a lot of Internet protocols & formatting
    • Customers still weary about proprietary protocols…. - Skinny works well, but it is proprietary
    • It’s about the Applications!! - The next “Killer App” is the integration of voice, data, video, IM & Presence… SIP can do this.
    • Microsoft!! 250 millions desktops might speak SIP soon…. - SIP client will be added to WindowsXP in October
  • The history of SIP
    • S ession I nitiation P rotocol (SIP) is defined via RFC2543 on March 17, 1999.
    • Additional “feature” drafts have been written to address issues which concern SS7/ISUP handling, QoS, Alerting, DHCP, 3PCC, Firewalls & NAT, etc…
    • IETF SIP-WG created in September, 1999
    • RFC2543bis (additions) created in April 2000.
    • Vendor interoperability testing done at the semi-annual SIP Bakeoff (8th in August in UK)
  • The various flavors of SIP
    • RFC2543 - “vanilla” SIP - the most commonly deployed & developed by commercial vendors
    • SIP-T - inter Call Agent (MGC) protocol for carrying SS7 / ISUP messaging - basically maps ISUP messaging to a MIME attachment
    • SIP extension from PacketCable - additions to Security, QoS & Privacy areas
  • 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
  • SIP Basics - Architectural Elements
    • Clients: SIP Phones, Softphones, Gateways, Media Gateway Controllers, PDAs, Robots - User Agent Client (UAC) / User Agent Server (UAS) - Originate & Terminate SIP requests
    • Typically an endpoint will have both UAC & UAS, UAC for originating requests, and UAS for terminating requests
    • Servers: - Proxy Server - Redirect Server - Registrar Server
  • 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
  • SIP Methods
    • Consists of Requests and Responses
    • 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
    • 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
    • Messages contain SIP Headers and Body. Body might be SDP or an attachment or some other application
  • SIP Addressing
    • Modeled after mailto URLs. May be a combination of FQDNs or E.164 numbers or both.
    • Support for Fully-Qualified Domain Names (FQDNs) using sip: URLs - sip: “John Doe” <jdoe@cisco.com>
    • Support for E.164 addresses - sip:14085551234@gateway.com; user=phone
    • Support for mixed addresses - sip:14085551234@10.1.1.1; user=phone sip:jdoe@10.1.1.1
    • Support for E.164 addresses using tel: URLs - tel:14085551234
  • 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
  • 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.
  • 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
  • 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
  • Agenda
    • Why VoIP? How does it work & why is it interesting?
    • Comparing & Understanding the VoIP Protocols - H.323 - Skinny - MGCP - SIP
    • SIP Tutorial
    • Sample VoIP Applications
    • Cisco VoIP products
    • IP IVR
      • Voice Portal
    • Auto Attendant
    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
    • IP Telephony Appliance
    • - Corporate directory integration via LDAP
    • - Web site integration via XML
    • - Personalized menu’s via softkeys
    • Extensible interface with IP services offers clear differentiation to PBX connected devices
    IP Phone Display Applications *
  • Convergence:Presence Services Managing your communications through web browsers, Instant Messaging and mobile devices
  • 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
  • Web Attendant
    • Ubiquitous access via a browser
    • Extension look-up via LDAP
    • Easy of use with drag and drop interface
    • Benefits:
      • Eliminates specialized receptionist phones
      • Access via URL
    • Included with Call Manager 3.0(tbd)
  • Voice Portal Solution
    • Extracts XML information from web page into IP IVR
    • Benefit
      • Only one place to configure and maintain data
      • Consistency
      • Lower admin costs
    IP Intranet Press #1 to Hear Stock Quote IP IVR Stock Quote *
  • VoiceXML PSTN Cisco Voice Gateway RTSP Server VoiceXML in IOS: HTTP Server Architectural Model: VXML Interpreter Context Document Server Implementation Platform VXML Interpreter
  • Agenda
    • Why VoIP? How does it work & why is it interesting?
    • Comparing & Understanding the VoIP Protocols - H.323 - Skinny - MGCP - SIP
    • SIP Tutorial
    • VoIP Applications
    • Cisco VoIP products
  • Cisco VoIP Products
    • Call-Processing - Cisco CallManager - Multimedia Conference Mgr - H.323 Gatekeeper / Proxy - Cisco SIP Proxy Server (CSPS) - BTS10200 Softswitch - VSC3000 Softswitch
    • 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
    • IP Phones - 7910, 7940, 7960, 7935, Softphone
    • Applications - Unity UM, Personal Assistant, Conference Connection, IP IVR, IP Contact Center, Web Attendant, XML / BTXML on IP Phones - 80+ EcoSystem partners
    • Cisco Infrastructure - IOS QoS features, Line-Powered Catalyst Switches, Catalyst QoS features - Application Layer Gateway (ALG) in IOS-NAT / Firewall, PIX
  • Questions?
  • Voice over IP (VoIP) Brian Gracely - [email_address]
  • Presentation_ID © 2001, Cisco Systems, Inc. All rights reserved.