- Geographic and Regulatory Considerations
- IP Connectivity Options
- Dial Plans and Call Routing
- Supplementary Services
- Network Demarcation
- Security Considerations
- Session Management, Call Traffic Capacity, Bandwidth Control, and QoS
- Trunk Provisioning
- Scalability and High Availability
- SIP Trunk Monitoring
- Further Reading
Dial Plans and Call Routing
Adding a SIP trunk service to your network most likely means there are service changes (accessible numbers and their associated cost), and you should optimize call routing in your network for the most cost-efficient calling patterns. This optimization, in turn, can affect current call admission control (CAC) and bandwidth-allocation policies implemented in your network.
Some specific items that might affect your dial-plan and call-routing configuration include
- If you currently have a separate dedicated Time Division Multiplexing (TDM) Public Switched Telephone Network (PSTN) voice gateway per Cisco Unified Communications Manager (CUCM) cluster or IP Private Branch Exchange (PBX), then you have a single enterprisewide SIP trunk shared between them.
- If the SIP trunk offers only long-distance (or certain types of inter-regional) calls, then your TDM PSTN gateways offers both local and long-distance calls.
- Whether the SIP trunk is going to be used by all the users in your network (all sites) or only by users colocated at the site where the SIP trunk terminates.
- Whether routing of emergency or fax, modem, Point of Service (POS), or Telecommunications Device for the Deaf (TDD) calls need to be rethought because they might not initially make use of the SIP trunk service.
Certain service providers require that a "+" be added to the front of a phone number sent on a SIP trunk. Specifically, the From field in a SIP message must be valid, as in From: +14085551212. When interconnecting through CUCM, this configuration can be accomplished by using translation rules on a Cisco Unified Border Element (CUBE) between CUCM and the SIP trunk service provider.
Certain SIP trunk providers require users to complete a registration before they can use the service. This security practice is a good one for service providers to ensure that calls originate from only well-known endpoints. CUCM does not natively support registration on SIP trunks, but this support can also be accomplished by using a CUBE. The CUBE registers to the service provider with the phone numbers of the enterprise on behalf of CUCM.
Two additional considerations regarding call routing include
- Direct Inward Dial (DID) number reachability
- Emergency call routing
Porting Phone Numbers to SIP Trunks
When an enterprise starts using a SIP trunk for incoming calls, the phone number must be ported to this service. When external end users call the number, rather than ringing at the traditional TDM gateway owned by the enterprise, it rings in the service provider's core network, and the call is routed to the enterprise with the SIP trunk.
Because of the complexity of porting phone numbers, most SIP deployments find it easier to start services with outbound calls or with inbound contact center toll-free service calls (non-DID). It is important for the enterprise to understand the timelines and transition plans offered by the service provider for porting DID numbers. Enterprises' business users cannot afford to be unreachable on their primary PSTN phone numbers while this porting activity occurs.
Emergency calling is an important consideration to account for when integrating SIP trunk access into the enterprise. Traditionally emergency calling is based on the emergency responder knowing the physical location of the TDM connection from which the call is coming. With a SIP trunk, that relationship between the physical location and the calling number no longer exists.
Options for handling emergency calling include
- Continuing to route emergency calls through your TDM PSTN gateways
- Having a small number of TDM trunks dedicated to this function at the physical location of the service provider
- Adopting a SIP-based emergency calling solution
All SIP trunk providers should provide clear explanations of their solution for providing emergency calling when an IP connection is evaluated. Some aspects to emergency calling have not been solved technologically or with the currently offered services. In the United States, the Federal Communications Commission (FCC) continues to work with the industry to define E911 operation, and a geolocation SIP header is in an Internet Engineering Task Force (IETF) draft status.
Investigate these issues in all countries and areas of the world where your network is considering a SIP trunk for PSTN access because the capabilities and regulations vary significantly.