IP telephony solution. Gatekeepers can be used for scalability of dial plans and for bandwidth management when using the H.323 protocol. Figure shows a company with a headquarters in Chicago, two offices in the San Francisco area, and one smaller office in Dallas. In Chicago, three headquarters locations connect via a metropolitan area network (MAN). The main west coast office in San Francisco connects with the San Jose office via a MAN. Dallas has a single site. The three main locations of the company (Chicago, San Francisco Bay Area, and Dallas) interconnect via an IP WAN. The Chicago, San Francisco, and Dallas locations each have a Cisco Unified CallManager cluster serving local IP phones and IP phones located in the MAN-connected sites. At the airport campus in Chicago, IP phones are not used because the company is using a managed telephony service that is offered by the owner of the building. However, a voice gateway connects to the managed PBX, allowing VoIP calls to and from the airport office telephones through the gateway. The Chicago headquarters sites use the IP WAN router as a gatekeeper that provides CAC and bandwidth management for H.323 calls. In addition, each main site has a voice gateway that connects to the PSTN, allowing off-net calls. These gateway routers are equipped with digital signal processors (DSPs) that provide conferencing and transcoding resources. Within each area, the G.711 codec is used, while calls between the three areas use the G.729 codec. All calls within the enterprise should use the IP WAN. If the IP WAN fails, or if calls are denied by CAC, the calls are rerouted through the PSTN.
Content 2.5 Implementing VoIP in an Enterprise Network 2.5.2 Deploying CAC IP telephony solutions offer CAC to limit the number of concurrent voice calls permitted in order to prevent oversubscription of WAN resources. Figure shows how CAC is used in a network. Without CAC, if too many calls are active and too much voice traffic is sent at once, delays and packet drops occur. Even giving RTP packets absolute priority over all other traffic does not eliminate problems when the physical bandwidth is not sufficient to carry all voice packets. QoS mechanisms do not associate individual RTP packets with individual calls; therefore, all RTP packets are treated equally. All RTP packets will experience delays, and any RTP packets can be dropped. The effect of this behavior is that all voice calls experience voice quality degradation when oversubscription occurs. A common misconception is that only calls that are beyond the bandwidth limit suffer from quality degradation. CAC is the only method that prevents general voice quality degradation that is caused by too many concurrent active calls. Example: CAC Deployment
Figure shows a scenario with two sites, with three telephones at each site connected to a VoIP gateway through a PBX. The two gateways are connected via an IP network. The network is designed for a maximum of two concurrent calls. If CAC is not used, whenever there are three active calls, all three calls experience severe voice quality degradation. When CAC is deployed, the gateways are configured to allow no more than two calls at the same time. When a third call is attempted, the call is blocked. With the CAC configuration, no voice quality problems should be experienced.
Content 2.5 Implementing VoIP in an Enterprise Network 2.5.3 Voice Gateway Functions on a Cisco Router Cisco routers, especially the Cisco Integrated Services Router (ISR), such as the Cisco 3800 Series ISR, are voice capable. These routers can be equipped with traditional telephony interfaces to act as gateways for analog and digital devices including telephones, faxes, PBXs, and the PSTN, allowing those devices to interact with VoIP networks. Figure lists voice gateway functions in Cisco routers. The routers support numerous analog interfaces, digital interfaces, and signaling protocols: Gateways with analog interfaces convert analog signals into digital format before encapsulating voice into IP packets. The gateways can compress digitized voice before the encapsulation happens. This compression reduces the bandwidth that each call needs. Cisco IOS routers support H.323, session initiation protocol (SIP), and Media Gateway Control Protocol (MGCP) for VoIP signaling. In addition, gateways can be equipped with DSPs, which provide conferencing and transcoding resources. In IP telephony environments, gateways support fallback scenarios for IP phones that have lost IP connectivity to their call agent (that is, Cisco Unified CallManager). This feature, called Cisco Survivable Remote Site Telephony (SRST), enables the gateway to take the role of the call agent during WAN failure. Local calls can then proceed even if IP connectivity to Cisco Unified CallManager is broken. In addition, Cisco SRST can route calls out to the PSTN and, thus, use the PSTN as the backup route for calls toward any site that is not reachable via IP. Further, Cisco IOS routers can permanently act as a call agent for IP phones. The feature that provides this functionality is Cisco Unified CallManager Express. With Cisco Unified CallManager Express, the router provides Cisco Unified CallManager functionality. If the router is also a voice gateway, the router combines IP telephony and VoIP gateway functionality in a single box. Cisco IOS gateways also support other features, such as call preservation (Real-Time Transport Protocol [RTP] stream) in case of a lost signaling channel, dual tone multifrequency (DTMF) relay capabilities, supplementary services support (for user functions, such as hold, transfer, and conferencing), and fax and modem support.
Content 2.5 Implementing VoIP in an Enterprise Network 2.5.4 Cisco Unified CallManager Functions Cisco Unified CallManager is the IP-based PBX in an IP telephony solution. Cisco Unified CallManager acts as a call agent for IP phones and MGCP gateways and can also interact with H.323 or SIP devices using the devices’ protocols. For redundancy and load sharing, multiple Cisco Unified CallManager servers operate in a cluster as shown in Figure . From an administration perspective, the whole cluster is a single logical instance. There are six main functions performed by Cisco Unified CallManager: