Chapter 55 Configuring a VoIP Network

Understanding How VLANs Work

•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 includes 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 the 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 the 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 the voice packets are sensitive to delays, TCP is not suitable for voice transmission because the timeouts and retries increase the delay between the 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 routes it to the PSTN.

Understanding How VLANs Work

This section describes the native VLANs and the auxiliary VLANs. This section uses the following terminology:

•Auxiliary VLAN—Separate VLAN for IP phones

•Native VLAN—Traditional VLAN for data

•Auxiliary VLAN ID—VLAN ID of an auxiliary VLAN

•Native VLAN ID—VLAN ID of a native VLAN

Catalyst 6500 Series Switch Software Configuration Guide—Release 8.7

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OL-8978-02

 

 

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Cisco Systems OL-8978-02 manual How a Call Is Made, 55-8
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OL-8978-02 specifications

Cisco Systems OL-8978-02 is a comprehensive training curriculum designed to enhance the skills and knowledge of network professionals seeking to advance their careers in network management and security. This curriculum offers a significant focus on Cisco networking technologies and concepts, ensuring that learners are well-equipped to handle the intricacies of modern network environments.

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