Introduction to SIP “Trunking” in the Enterprise Henry Pinera Systems Engineer hpinera@avaya.com February 2009

Why SIP? A great pedigree SMTP HTTP SIP

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

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

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 (fchang@avaya.com) 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.

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

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

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

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)

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: http://www.avaya.com/devconnect Professional Services engagement for enterprise customers In-region R&D Contact: In-region Product Manager http://www.avaya.com/master-usa/en-us/resource/assets/applicationnotes/genspsiptrk-v1-1.pdf

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]

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

Peering Enterprise Contact Centers

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The Enterprise Infrastructure B 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

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 http://www.avaya.com/master-usa/en-us/resource/assets/applicationnotes/genspsiptrk-v1-1.pdf Session Border Controller Avaya SIP Enablement Server (SIP Proxy) Avaya Communication Manager (IP-PBX)

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.