Cisco WS-X6K-SUP1A-PFC= Software Guide - Page 823

Chapter 44 Configuring a VoIP Network Understanding How a VoIP Network Works Converged Voice Gateway The Cisco Voice Gateway 200 (VG200) allows you to connect standard POTS phones (connected directly to the gateway or anywhere on the PSTN) with Cisco IP or any H.323-compliant telephony devices. When used with Cisco CallManager, the VG200 functions as a Media Gateway Control Protocol (MGCP) gateway. The Cisco VG200 provides a 10/100BASE-T Ethernet port for connection to the data network. The following telephony connections are also available: • One to four Foreign Exchange Office (FXO) ports for connecting to a central office or PBX • One to four FXS ports for connecting to POTS telephony devices • One or two T1 digital ports for connecting to the following: - PSTN using FXO emulation - T1 channel bank using FXS emulation - PBX through a trunk (tie) line using ear and mouth (E&M) emulation These ports can be used to integrate a VoIP network with POTS devices, PBXs, or the PSTN. To configure the Cisco VG200, refer to the documentation that shipped with the gateway. How a Call Is Made An IP phone connects to a LAN either through a hub port or a switch port. The IP phone boots up and uses DHCP to get its IP address and the IP address of its TFTP file server. The IP phone uses its IP address to talk to the TFTP server and gets its configuration file. The configuration file includs the IP address of the phone's Cisco CallManager(s). The phone then talks with Cisco CallManager and registers itself. Each time a phone boots up, it might get a different IP address. Cisco CallManager knows how to associate a consistent user phone number to a particular phone by using the MAC address of the phone. Cisco CallManager always maintains a table mapping the "phone MAC address" and "phone number." Each time a phone registers, the table is updated with the new IP address. During registration, Cisco CallManager downloads the key pad template and the feature capability for the phone. It tells the phone which run-time image it should use. The phone then goes to the TFTP server to get its run-time image. Each phone has a dedicated TCP connection to Cisco CallManager called the "control channel." All control information, such as key pressing, goes from the phone to Cisco CallManager through this channel. Instructions to generate ring tone, busy tone, and so on comes from Cisco CallManager to the phone through this channel. Cisco CallManager stores the IP-address-to-phone-number mapping (and vice versa) in its tables. When a user wants to call another user, the user keys in the called party's phone number. Cisco CallManager translates the phone number to an IP address and generates an IP packet version of ring tone to the called IP phone through the TCP connection. When the called IP phone receives the packet, it generates a ring tone. When the user picks up the phone, Cisco CallManager instructs the called IP phone to start talking with the calling party and removes itself from the loop. From this point on, the call goes between the two IP phones through the Real-Time Transport Protocol (RTP) which runs over the User Datagram Protocol (UDP). Because voice packets are sensitive to delays, TCP is not suitable for voice transmission as timeouts and retries increase the delay between packets. When any change occurs during the call due to a feature being pressed on one of the phones, or one of the users hanging up or pressing the flash button, the information goes to Cisco CallManager through the control channel. If a call is made to a number outside of the IP PBX network, Cisco CallManager routes the call to an analog or digital trunk gateway which in turn routes it to the PSTN. 78-13315-02 Catalyst 6000 Family Software Configuration Guide-Releases 6.3 and 6.4 44-7