.NODE

Influencing Path Selection

Thus far, this book has discussed various ways to configure a gateway to ensure that your calls go through. This chapter introduces ways to control the path that calls take and to prevent calls from being placed without adequate resources to support them.

It is usually desirable to have more than one path for a call to take. Multiple paths provide redundancy in the event of a link failure or insufficient resources and sometimes are used to reduce the transport cost of a call. These paths might consist completely of Voice over IP (VoIP), or they might be a mixture of VoIP and plain old telephone service (POTS).

When you are using VoIP, keep in mind that voice is an application on the network, and voice messages are carried in IP packets. You can implement techniques to choose which VoIP dial peer is used for a call, but the IP routing structure of your network determines the path to that target IP address. This chapter talks about ways to control call routing using commands and techniques that apply to the voice application, but IP routing ultimately controls VoIP routing. VoIP traffic can be affected by router configuration such as access lists that might block a voice subnet, or policy-based routing that might send voice traffic out a different interface than the one desired. Various commands enable you to control the source IP address of voice packets; you must consider this when you are configuring policies along the entire voice path so that you avoid inadvertently blocking that traffic.

When you are planning and troubleshooting VoIP path selection, take these two levels of call routing into account. In planning, be sure that your voice calls will take the path at each hop that you want them to. When troubleshooting, test the IP path chosen through the network.

Many different ways exist to control call routing and make call admission decisions. With some methods, you make the decisions on the gateway. With others, you use a separate device. In H.323 networks, gatekeepers can control call routing; in Session Initiation Protocol (SIP) networks, proxy servers can perform that function. In Cisco Unified Communications Systems, CallManager makes call routing and admission decisions. In this chapter, you will learn about the following:

  • Use of hunt groups and trunk groups
  • Use of tail-end hop-off
  • CAC techniques based on local gateway settings, on measurements of network performance, and on router resource availability
  • Use of the IP Service Level Agreement (SLA) tool
  • Use of Resource Reservation Protocol (RSVP) and the RSVP Agent
  • Considerations when routing between POTS calls

Hunt Groups

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

show all menu





Cisco Voice Gateways and Gatekeepers
Cisco Voice Gateways and Gatekeepers
ISBN: 158705258X
EAN: 2147483647
Year: 2004
Pages: 218
Similar book on Amazon

Flylib.com © 2008-2017.
If you may any questions please contact us: flylib@qtcs.net