UC Deployment Models: Centralized Call Processing

The centralized call processing model offers toll bypass efficiencies for inter-site calls. Inter-site calls are routed over the IP WAN, instead of using PSTN resources. Inter-site call routing over the data network is commonly referred to as toll bypass because the toll charges of the PSTN are avoided when calls are routed over the PSTN. The centralized call processing model also allows a company to more efficiently utilize PSTN circuits from utilization and fault tolerance perspectives. Routing phone calls over an IP WAN allows PSTN resources to be centralized at the headquarters sites so the company does not have to manage and maintain gateway resources at each site. Companies choosing to centrally locate all PSTN resources should consider purchasing at least one analog FXO PSTN gateway resource at each site for 911 emergency call routing (911 in the United States). One or two analog voice ports at each site would allow call routing to the local public safety answering point (PSAP). The Call Manager 4.2/6.0 device mobility feature should be configured for environments with nomadic users. Cisco Emergency Responder offers the best solution for emergency call routing with nomadic users, but requires additional resources. Centralized call processing scenarios with distributed PSTN resources allows the company to support additional PSTN calls for each site and save additional toll charges for long distance and international calls. We will examine a simple scenario where a company has sites in New York, New Jersey, and Reading, England. Each U.S. site has one PRI to the PSTN and the England site has an E1 to the PSTN. Calls from New York or New Jersey to PSTN destinations in England could be routed over the IP WAN from New York or New Jersey to England, where calls can be routed to the PSTN as a local call. This scenario would avoid the international toll charges normally incurred in routing calls between the U.S. and international sites. The functionality described above is commonly referred to as tail-end hop off (TEHO) or least cost routing (LCR). Centralized call processing with distributed PSTN connectivity also allows resiliency and utilization advantages. If all of the outgoing B channels of the PRI in New York are utilized, CUCM can route additional calls across the IP WAN to New Jersey where the additional calls are routed to the PSTN. If the PRI in New York went down, all outgoing PSTN calls could be routed through the New Jersey PRI utilizing the IP WAN. If the IP WAN goes down in a centralized call processing model, the Cisco IP phones at the branch locations will start power cycling. The Cisco IP phones rely on CUCM to remain registered. Cisco IP phones send keepalives to CUCM every 30 seconds by default. If 3 keepalives are not responded to, the Cisco IP phones attempt to register to the other CUCM servers in the Call Manager group that is part of the phone configuration file. The Cisco IP phone will attempt to register with the SRST (survivable remote site telephony) router or switch after all CUCM servers are attempted. SRST is Cisco IOS code in the Cisco router that allows the Cisco router to provide call routing functionality. The Cisco IP phones register with the SRST gateway. The number of Cisco IP phones that can register with the SRST gateway is limited by the router hardware at the site. The 1751 router can only accommodate 24 Cisco IP phones, but the 3845 router can accommodate up to 240 phone registrations. The SRST administration guide includes details of how many phones and directory number are allowed for each router. The administration guide includes the details of SRST configuration. SRST configuration details are also covered in the Global Knowledge Cisco GWGK class. The next blog will continue our conversation of centralized call processing. REFERENCES: CUCM 7.x Solution Reference Network Design (SRND) Guide: http://www.cisco.com/en/US/netsol/ns818/networking_solutions_program_home.html www.cisco.com/go/srnd SRST 7.0 Administration Guide http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/admin/srst/configuration/guide/srstsa.html