1. Voice over IP (VoIP)

2. The World Is Now Global— All Apps Must Travel Time and Distance
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

3. VoIP Signaling Protocols
H ITU standard, ISDN-based, distributed topology %+ 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 RFC Centralized Call-Control Architecture - Call-Agents (MGC) & Gateways (MG)
SIP IETF RFC Distributed Call-Control Used for more than VoIP…SIMPLE: Instant Messaging / Presence

4. 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

5. 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

6. 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

7. SIP Basics - Architecture
NTELL
GENT
SERV
CES
Application
Services
3pcc
CPL
LDAP
CPL
Oracle
XML
SIP Proxy, Registrar
& Redirect Servers
SIP
SIP
SIP
PSTN
SIP User
Agents (UA)
CAS or PRI
RTP
(Media)
Legacy PBX

8. 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

9. 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

10. 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

11. 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”
Support for E.164 addresses - user=phone
Support for mixed addresses - user=phone
Support for E.164 addresses using tel: URLs - tel:

12. 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

13. Basic SIP Functionality - Call Forking
“Contact
and
Location Database
INVITE
“Where is
INVITE
Proxy / Redirect
Server
INVITE
INVITE
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

14. Basic SIP Functionality - Call Redirection
Location Database
“Where is
“You need to contact ”
Proxy / Redirect
Server
INVITE
3xx Moved
Contact:
INVITE
LOCAL PSTN
The user at informed the network
that he could be reached on his cell-phone at
National PSTN

15. 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: w/ SDP of SIP Phone
18x / 200 OK w/ SDP
18x / 200 OK w/ SDP
ACK w/ SDP of SIP Gateway
x1234
LOCAL PSTN

16. Agenda Why VoIP? How does it work & why is it interesting?
Comparing & Understanding the VoIP Protocols - H Skinny MGCP SIP
SIP Tutorial
Sample VoIP Applications
Cisco VoIP products

17. 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
Paging
Enterprise Database
Web Pages

18. 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 *

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

20. 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

21. 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 *

22. 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

23. 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, VG Mid Range: 3810, 2421, 2600, 3600, Cat4000, AS5300, 7200, High End: AS5350, AS5400, Cat6000, AS5850, MGX8850
IP Phones , 7940, 7960, 7935, Softphone
Applications Unity UM, Personal Assistant, Conference Connection, IP IVR, IP Contact Center, Web Attendant, XML / BTXML on IP Phones EcoSystem partners
Cisco Infrastructure IOS QoS features, Line-Powered Catalyst Switches, Catalyst QoS features Application Layer Gateway (ALG) in IOS-NAT / Firewall, PIX