Cisco Cisco Agent Desktop 8.0 クイック設定ガイド

Capturing an IP Phone Call

November 2013

Capturing an IP Phone Call

In order to capture an IP phone call, we need access to the data travelling over the
network and the format of the data. We will discuss the access issue later. In this
section, we discuss the format of the audio portion of the phone call.

Cisco Unified Communications Manager (Unified CM) and Unified Contact Center
(Unified CC) software, in concert with Cisco switches, gateways, and routers, package
audio phone calls as streams.

There are two streams of audio data for each call. To help distinguish the two streams
of audio data, the end points are referred to here as follows:

■

Agent phone. The agent is the person in the contact center whose phone is an
IP end point on the contact center’s network.

■

Caller phone. The caller refers to another person on the call (using an
IP phone or appropriate device).

Figure 1

shows one stream of audio being sent from the agent phone to the caller

phone and the other stream being sent from the caller phone to the agent phone. If
software can capture these two streams of packets from the network, the data can be
processed and stored in a format that can be listened to at a later time.

The protocol used to encapsulate the audio data is known as the Real-Time Transport
Protocol (RTP). As shown in

Figure 2

, an audio stream is made up of many individual

data packets sent over the network.

Figure 1.

IP phone call audio streams

Figure 2.

RTP audio stream