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Brian Gracely
Technical Marketing Engineer
Internet2_VoIP
Agenda
Why VoIP? Comparing & Understanding the VoIP Protocols - H.323 - Skinny - MGCP - SIP SIP Tutorial Sample VoIP Applications Cisco VoIP products
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Converged Networks
- Voice, Video & Data over an IP network - Reduced the costs of managing parallel networks - Allows voice to be an IP application
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GK
Directory
Infrastructure
IP Network
Clients
IP SoftPhone
The World Is Now Global All Apps Must Travel Time and Distance
Internet2_VoIP
2001, Cisco Systems, Inc. All rights reserved.
Agenda
Why VoIP? Comparing & Understanding the VoIP Protocols - H.323 - Skinny - MGCP - SIP SIP Tutorial Sample VoIP Applications Cisco VoIP products
Internet2_VoIP
Gatekeeper B
RRQ/RCF
ARQ
IP Network
H.225 (Q.931) Setup
RRQ/RCF
ARQ
V
Gateway A Phone A
RTP
V
Gateway B Phone B
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Cisco CallManager
E.164 Lookup
Ring Back
IP WAN
RTP Stream
Ring
Off Hook
H.323/MGCP Gateway
PSTN
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PSTN Gateway
BTS / VSC SS7
P S T N
SIP H.323
IMT
V
GK
PSTN
PRI
V V
Access Gateway
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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
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SIP is a very Internet friendly protocol, and Cisco likes Internet friendly stuff.
- SIP reuses a lot of Internet protocols & formatting
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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
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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
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LDAP
Oracle
XML
CPL CPL
SIP Proxy, Registrar & Redirect Servers SIP SIP SIP SIP User Agents (UA) PSTN
CAS or PRI
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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
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Registrar
Redirect
REGISTER Here I am
SIP Proxy
Proxied INVITE Ill handle it for you
SIP-GW
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Internet2_VoIP
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
Internet2_VoIP
2001, Cisco Systems, Inc. All rights reserved.
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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
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INVITE w/ SDP for Media Negotiation 100 Trying 180/183 Ringing w/ SDP for Media Negotiation MEDIA 200 OK ACK MEDIA BYE 200 OK
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Location Database
Where is sip:1-800-GO-CISCO@cisco.com?
INVITE sip:1234@10.1.1.1
INVITE sip:1-800-GO-CISCO@cisco.com
LOCAL PSTN
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.
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Where is sip:3921234@cisco.com?
Location Database
You need to contact 4721111
392-1234
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
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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
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Unity
Queuing
ICM
Directory Access
DB Access
LDAP
E-Mail
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Paging
Web Pages
Enterprise Database
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IP Telephony Appliance - Corporate directory integration via LDAP - Web site integration via XML - Personalized menus via softkeys Extensible interface with IP services offers clear differentiation to PBX connected devices
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Convergence:Presence Services
Managing your communications through web browsers, Instant Messaging and mobile devices
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IP
Distribution Groups with Queuing for Resources 2 Types of Queues Requestor Servicer
SoftPhone
PSTN
IP Phones
Remote Agents
Branch Agents
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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)
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VoiceXML
Architectural Model:
VXML Interpreter Context
VXML Interpreter Implementation Platform
Document Server
VoiceXML in IOS:
PSTN
HTTP Server
RTSP Server
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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
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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 - Systems, Application Layer Gateway (ALG) in IOS-NAT / Firewall, PIX 2001, Cisco Inc. All rights reserved.
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Questions?
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Presentation_ID
37