One of the primary benefits of a centralized call processing model is the ability to provide remote sites with all the advanced calling features that were previously available only on large campuses. Survivable Remote Site Telephony (SRST) allows a Cisco voice gateway to handle call processing if the connection between the remote site and the central site is severed.
H.323 and Session Initiation Protocol (SIP) gateways are intrinsically resilient to network connectivity failures because the dial plan and call control are in the gateway, and the gateway is always in control of call-routing decisions. Multiple Gateway Control Protocol (MGCP) gateways are under the control of the call agent, and only the call agent can make dial-plan and call-routing decisions. Therefore, when the call agent and the MGCP gateway have a connectivity failure, you must use a fallback method to return call control to the gateway for the period of the failure. The MGCP Gateway Fallback feature allows the gateway to assume call control for MGCP-controlled voice ports. The gateway uses this feature in conjunction with SRST to allow the router to take care of IP phone and public switched telephone network (PSTN) gateway call routing during network failures.
This chapter helps you to do the following:
Part I: Voice Gateways and Gatekeepers
Gateways and Gatekeepers
Part II: Gateways
Media Gateway Control Protocol
H.323
Session Initiation Protocol
Circuit Options
Connecting to the PSTN
Connecting to PBXs
Connecting to an IP WAN
Dial Plans
Digit Manipulation
Influencing Path Selection
Configuring Class of Restrictions
SRST and MGCP Gateway Fallback
DSP Resources
Using Tcl Scripts and VoiceXML
Part III: Gatekeepers
Deploying Gatekeepers
Gatekeeper Configuration
Part IV: IP-to-IP Gateways
Cisco Multiservice IP-to-IP Gateway
Appendix A. Answers to Chapter-Ending Review Questions
Index