1. Introduction to SIP “Trunking” in the Enterprise
Henry Pinera
Systems Engineer
February 2009

2. Why SIP? A great pedigree
SMTP
HTTP
SIP

3. Limitation & Challenges of Private IP (H.323/SIP) Trunking
Why SIP Trunking?
Limitation & Challenges of Private IP (H.323/SIP) Trunking
Private IP (H.323/SIP) trunks are limited to VoIP communications between internal systems/sites
Separate TDM interfaces are required for external communication (partners/suppliers/customers)
Extra cost, extra hardware, extra complexity
Customers/ Partners/ Suppliers
PSTN
Local &
Long Distance
TDM interfaces
External Communications
TDM interfaces
IP WAN
External Communications
IP Data &Telephony
LAN
LAN
Internal Communications
IP Phone
IP Phone
IP Phone
IP Phone

4. SIP Trunking A Single Pipe to the Cloud
Single IP link for voice/data
Optimize use of WAN access by consolidating voice and data services
Reduce PSTN interfaces for long-distance and local access
Assign local telephone numbers to any ‘virtual location,’ independent of physical location
Prepares for future SIP solutions
Customers/ Partners/ Suppliers
PSTN
Local &
Long Distance
SIP
SIP
WAN
H.323
LAN
LAN
IP Phone
IP Phone
IP Phone
IP Phone

5. Flavors of SIP Trunking
Service Provider
Enterprise / Contact Center (one or multiple locations)
Enterprise / Contact Center
Application Servers
Public SIP Trunking
Public SIP Trunks are used to provide access to the PSTN through SIP-enabled Service Providers such as AT&T and Verizon Business.
Communication Manager 4.0 and later support direct SIP trunking to SIP Service Providers without the SIP Enablement Services (SES).
If the SIP Service Provider does not support TCP or TLS, Communication Manager will need to interface to a Session Border Controller such as the Covergence SBC to perform protocol conversion from TCP or TLS to UDP. Communication Manager supports SIP over TCP and TLS whereas most SIP Service Providers only support UDP.
For U.S. Service Providers, DevConnect testing for Public SIP Trunking is highly recommended, but not required.
For non-U.S. Service Providers, DevConnect testing for Public SIP Trunking is required.
Private SIP Trunking
Private SIP Trunks are used to network Communication Manager to other SIP devices within the Enterprise such as other Communication Manger instances or even third-party PBXs.
Communication Manager Private SIP Trunking to 3rd party (non-Avaya) SIP devices, PBXs, gateways, applications, etc. require CM product management approval. This is true regardless of whether there is an Avaya SES or a SIP Proxy in an element such as a Session Border Controller in the configuration. Product management will take the following into consideration in making its approval decision:
Customer relationship
Business opportunity/revenue potential
3rd party device/manufacturer
Prior experience with similar configurations
Successful testing via Proof of Concept lab, System Interoperability Test Lab, or DevConnect lab
SIP trunking can be over the LAN, WAN
or dedicated circuits.

6. Building Blocks of SIP Trunking
Server / Gtwy
Service Provider
SBC
Service Provider
Service
Session Border Controller (SBC)
Customer Premise
Edge security device: SBC, Firewall, etc
Communication Platform: Server(s), Gateway(s), SIP Proxies, etc
Application Servers: messaging, video & audio conferencing, collaboration tools, IM, Presence, and more
Endpoints: hardphones, softphones, mobile devices, application integration
… or direct connection

7. Components of SIP Trunking Solution
Customers
Partners
Suppliers
Customers
Partners
Suppliers
ITSP
IP Office
PSTN
G860
DS3/ISDN
Communication Manager
CM Branch Edition
SBC
Single IP link for voice/Multimedia/Data
Optimize use of WAN access by consolidating voice and data services
Eliminate PSTN interfaces for long-distance and local access (carrier provides the gateways)
Assign local telephone numbers to any ‘virtual location,’ independent of physical location
Save on toll charges
Prepares for future SIP solutions
Modular Messaging
Meeting Exchange
SES
SIP
SIP
SIP
Avaya one-X©
Communicator
Voice Portal

8. SIP Trunking - Overview
SIP phone (future)
SIP
Provider(s)
IP DECT
IP phone (46xx/56xx/T3 IP)
T1/E1/
ISDN
Provider(s)
Digital phone (24xx/54xx/T3)
IP Office acts as a SIP gateway
Analog phone
Analog
Provider(s)
No need for a SIP phone to make/receive a SIP call
SIP Trunks are licensed in increments of 1, 5, 10 and 20 (up to 128 channels)
Benefits – Lower operational costs
Lower cost/flat rate calls with Internet Telephony SPs
Offers back-up to Analog/Digital Trunks + Number Portability
Future-proof your investment

9. SIP Trunking – IP Office Interoperability
Internet Engineering Task Force (IETF) RFC Support
All the major RFC’s are supported to ensure SIP interoperability with most Service Providers
Some of the vendors that we have performed Interoperability with;
North America:
AGN (US)
AT&T/SBC (US)
Bandtel (US)
Cbeyond (US)
Curry IP Solutions (US)
XO (US)
FWD / Free-World Dialup (US)
Global Crossing (US)
Les.net (Canada/USA)

10. Will CM Support SIP Trunks from XYZ SP?
DevConnect compliance testing
For U.S. SIP SPs, no longer a requirement but still HIGHLY RECOMMENDED
No change to non-U.S. Service Providers; DevConnect compliance testing required
Contact:
Professional Services engagement for enterprise customers
In-region R&D
Contact: In-region Product Manager

11. SIP SP – DevConnect Members
United States
AT&T
BandTel
Bandwidth.com
Clear Channel Satellite
ConneXon Telecom (911 Enable)
Cox Communications
Global Crossing
Nectar (AGN Network)
PAETEC (McLeodUSA)
Telcordia
Telepacific
Verizon Business
XO Communications
EMEA
BT [Germany and Spain]
Club Communications [UK]
COLT [UK and Germany]
KPN [Netherlands]
APAC
Singtel [Singapore]
Telstra [Australia]
TFN [Taiwan]

12. Communication Manager systems
Architecture
High Density Gateways
(Avaya G860)
Service Provider
3rd party
systems
Voice
Portal
SIP
Proxy
Outsourcer
Business Applications
Branch
Agents
Remote Agents
Communication Manager systems
Conferencing
Unified Messaging

13. Peering Enterprise Contact Centers

14. © 2007 Avaya Inc. All rights reserved.
Avaya – Proprietary & Confidential. Under NDA 14

15. The Enterprise InfrastructureB
Large and/or Backup Sites
G650, ESS
IP Phone
Avaya Communication Manager
G650 Gateways A
LAN
IP WAN C
Medium Sites
G450’s
S8300 LSP
The primary server contains all the licensing
Single system to manage and maintain
View system wide trunk information
Manage call routing based on location (best service routing).
All translation are automatically synchronized with alternate survivable servers in the system.
ANY Gateway E
Digital/Analog D
Micro Sites
G250
S8300 LSP or basic survivability
Small Sites G350
S8300 LSP or basic survivability

16. Building a SIP Trunk (Drag & Drop)
Service Provider
Application Notes for Configuring Avaya Communication Manager SIP Trunking to a Typical SIP Service Provider – Issue 1.1
Session Border Controller
Avaya SIP Enablement Server (SIP Proxy)
Avaya Communication Manager (IP-PBX)

17. CM to ITSP (Outbound call)
If the Far-end Domain is filled out then the “Request URI” and the “To” headers will reflect the information.
The Public-unknown numbering plan inserts the From extension.
The Near-end Clan for the signaling group has an ip-network-region authoritative domain that will appear in the “From” header.
Understand that the URI is the routable portion of the request where the To is the Logical recipient of the request and the From is the Initiator of the request.