Introduction to IP Telephony | Cisco VOIP

IP telephony describes products and solutions to transport voice traffic over IP. You can use IP to create a converged network to transport voice, video, and data communications. There are numerous benefits to this type of infrastructure, including simplified administration, cost savings on telecommunications fees, and unified messaging services.

Voice traffic and data traffic require two completely different solutions. Data traffic is relatively resilient and tolerant of slow WAN links, lost packets, and unsequenced packets. Voice traffic, on the other hand, is not. Voice requires packets to be received in the same order in which they were sent; if a packet is lost, it should remain lost, as retransmitting the packet would only confuse the person on the receiving end of the call.

There are several components that must be added to your infrastructure. These components include, but are not limited to, specialized router interfaces, specialized LAN switch modules and interfaces, IP telephone handsets, Cisco CallManager servers, and Cisco Unity Mail, as well as other unified messaging solutions.

Conferencing and Transcoding, and Other Services

Conferencing allows multiple participants to communicate in a single call. Cisco technologies support two types of conferencing: ad-hoc and meet-me. In ad-hoc conferencing, the originating caller controls the conference, and determines who will be on the call. The participants may even continue the call after the originating caller hangs up.

A meet-me conference allows participants to participate in a conference by calling an assigned number out of a pool of numbers. More participants can continue to join the conference call until the maximum number allowed is reached. DSP resources support both types of conferencing, and the Cisco CallManager uses DSP resources to provide conferencing services, as shown in Figure 1.

Figure 1: Conferencing

In this scenario, an IP phone caller joins another IP phone and an outside or PSTN initiated caller in a three-way conference call. This is an example of an ad-hoc conference. The Catalyst DSP resources are one way a Cisco CallManager is able to provide a conference bridge. A four-way G.711 conference call would utilize four DSP resources, one for each participant to stream into a single call.

Software conferencing is based on G.711, whereas hardware-based solutions support G.711, G.729a, and G.723. The newer Cisco IP phones 7900 Series supports G.711 and G.729a, but the older style supports G.711 and G.723.

Transcoding is the process of converting IP packets of voice streams between a low bit-rate (LBR) CODEC to and from a G.711 CODEC. An LBR is a CODEC such as G.729a or G.723. An example of a need for transcoding is when a user across the IP WAN wants to access a voice mail system which only supports G.711 and CallManager is configured to initiate remote IP calls using a G.729a CODEC, as shown in Figure 9.18. In this scenario, transcoding must be performed to convert the G.729a voice stream to G.711 in order to communicate with the voice mail system.

Figure 2: Transcoding

Figure 2 serves to illustrate that there are several components necessary to make your voice network a dream come true. Being familiar with these components is one of your first steps to achieving your voice goals. There are also several common telephone functions that any system should be able to provide, regardless of whether they are traditional PSTN or VoIP.

Call Transfer

Cisco IP phones support call transfer. By signaling back to the Call Manager (CM), a call can be transferred to the final destination.

Call Forward

Cisco IP telephony supports three types of call forwarding:

• Call Forward All Forwards all calls.

• Call Forward Busy Forwards calls only when the line is in use.

• Call Forward No Answer Forwards calls when the phone is not answered within a certain configurable number of seconds.

Call Park and Call Pickup

The Call Park feature allows a person to receive a call at another telephone for privacy. A Park soft key allows the receiver to place the caller on hold and dial a designated extension number. At another phone, the extension number can be dialed to pick up the call.

The Call Pickup feature is used to answer an incoming call that is ringing at an unattended telephone. Buttons or soft keys may be configured to activate this function.

Music on Hold

Music on Hold plays music when a caller is on hold, and the music stream may be a .WAV file or a fixed external device controlled by the CM.

Interactive Voice Response

IVRs are useful for routing calls to the appropriate person or department and are less expensive than having an individual do it. In most typical situations, there is an IVR on any public incoming line. Support on Cisco products for IVR is achieved with Tool Command Language (TCL) scripts and voice files, which are referenced in the configuration. When a call comes into the router and matches a set of criteria, the script is queried. The script runs and, depending on the digits it captures, plays an audio file. This audio file is stored in the router's flash and loaded into memory. The audio files use the standard .AU format. The scripts also have the capability to reroute calls.