Brian Gracely Technical Marketing Engineer 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 Open Service Application Layer (JAIN, AIN, TAPI, JTAPI, XML etc.) TDM/ Circuit Switch Open/Standard Interface Switching Network Line Concentration Call Control Connection Control Features Open Call Control Layer (SIP, H.323, MGCP, etc.) Digital Trunk Subsystem Common Channel Signaling Complex Administration Maintenance Billing Open/Standard Interface Standards-Based Packet Infrastructure Layer (IP, ATM)

AVVID Architecture - Open Packet Telephony Collaboration IP IVR, IP AA Apps Engine Intelligent Contact Manager ICM Applications Cisco Unity Voice Mail, UMS Video Voice Portal Call Processing Call Processing GK Directory PSTN PSTN gateways Analog phone support DSP farms IP Network Infrastructure Clients IP SoftPhone The World Is Now Global— All Apps Must Travel Time and Distance

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 IP Network Gatekeeper A Gatekeeper B LRQ LCF ACF ACF RRQ/RCF RRQ/RCF ARQ H.225 (Q.931) Setup ARQ V V H.225 (Q.931) Alert and Connect H.245 RTP Gateway A Gateway B Phone A Phone B

Basic Skinny Call IP WAN PSTN Cisco CallManager H.323/MGCP Gateway Voice Mail Server Call Setup Cisco CallManager IP WAN E.164 Lookup Ring Back Ring RTP Stream Off Hook H.323/MGCP Gateway PSTN

MGCP Architectures & Mixed Protocols SCP PSTN Gateway SIP or H.323 Network BTS / VSC SS7 SIP H.323 P S T N IMT V GK PSTN PRI V V Access Gateway MGCP RTP SIP / H.323

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 Session Initiation Protocol (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 NTELL GENT SERV CES Application Services 3pcc CPL eMail LDAP CPL Oracle XML SIP Proxy, Registrar & Redirect Servers SIP SIP SIP PSTN SIP User Agents (UA) CAS or PRI RTP (Media) Legacy PBX

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) LocationDatabase Registrar Redirect “Where is this name/phone#?” 3xx Redirection “They moved, try this address” REGISTER “Here I am” SIP Proxy Proxied INVITE “I’ll handle it for you” INVITE “I want to talk to another UA SIP User Agents 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 MEDIA 200 OK ACK MEDIA BYE 200 OK

Basic SIP Functionality - Call Forking “Contact 1234@10.1.1.1, 1234@10.1.1.2 and 1234@10.1.1.3” Location Database INVITE sip:1234@10.1.1.3 “Where is sip:1-800-GO-CISCO@cisco.com?” INVITE sip:1234@10.1.1.2 Proxy / Redirect Server INVITE sip:1234@10.1.1.1 INVITE sip:1-800-GO-CISCO@cisco.com 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. LOCAL PSTN

Basic SIP Functionality - Call Redirection Location Database “Where is sip:3921234@cisco.com?” 392-1234 “You need to contact 4721111” Proxy / Redirect Server INVITE sip:3921234@cisco.com 3xx Moved Contact: sip:4721111@10.1.1.3 INVITE sip:4721111@10.1.1.3 LOCAL PSTN The user at 392-1234 informed the network that he could be reached on his cell-phone at 472-1111 National PSTN

3rd-Party Call-Control (3pcc) & Back-to-Back UserAgent (B2BUA) 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 SIP Controller - 3pcc Application INVITE sip:1234 w/o SDP INVITE sip:9194721111 w/ SDP of SIP Phone 18x / 200 OK w/ SDP 18x / 200 OK w/ SDP ACK w/ SDP of SIP Gateway x1234 LOCAL PSTN

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

Application Engine Architecture External Services Packaged Solutions Application Toolkit IP IVR Voice Portal Auto Attendant VXML services Unity Telephony Queuing ICM Directory Access DB Access Notification Services Queuing (ACD) Personalized Apps Customer Apps LDAP Notification Server Web Access E-Mail Paging Enterprise Database Web Pages

IP Phone Display Applications 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 *

Convergence:Presence Services Managing your communications through web browsers, Instant Messaging and mobile devices

Informal Agent Queuing (IAQ) Central Site Distribution Groups with Queuing for Resources 2 Types of Queues Requestor Servicer IAQ Server IP SoftPhone PSTN Branch Agents IP Phones Remote Agents

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)

Press #1 to Hear Stock Quote Voice Portal Solution IP IVR Stock Quote IP Intranet Extracts XML information from web page into IP IVR Benefit Only one place to configure and maintain data Consistency Lower admin costs Press #1 to Hear Stock Quote *

VXML Interpreter Context Implementation Platform VoiceXML Architectural Model: VXML Interpreter Context Document Server VXML Interpreter Implementation Platform VoiceXML in IOS: HTTP Server PSTN RTSP Server Cisco Voice Gateway

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? Q&A

Brian Gracely - bgracely@cisco.com Voice over IP (VoIP) Brian Gracely - bgracely@cisco.com

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