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Automated Call Center Implementation Using SIP Call Signaling

IP.com Disclosure Number: IPCOM000004602D
Original Publication Date: 2001-Feb-28
Included in the Prior Art Database: 2001-Feb-28
Document File: 4 page(s) / 33K

Publishing Venue

Motorola

Related People

James E. Womack: AUTHOR [+2]

Related Documents

RFC 2543: OTHER [+5]

Abstract

Automated Call Center Implementation Using SIP Call Signaling

This text was extracted from a Microsoft Word 97 document.
This is the abbreviated version, containing approximately 57% of the total text.

Automated Call Center Implementation Using SIP Call Signaling

By

James E. Womack and Andrew Hull

Introduction

Current automated call systems require that one navigate through automated call centers by pushing the telephone buttons after choices have been given to a caller by a voice on the other end. This is time consuming process. If a menu can be downloaded and the user has a means to render it, the user can quickly navigate through the menu and get to the operator that is needed. We propose a way this can be done with no voice connection (other than the destination operator), and hence no unnecessary bandwidth wasted. Once the navigation is complete, if the operator is not available, the system will later initiate a call (when the operator is ready) to the original caller through either a public switched telephone network (PSTN) or the public data network (PDN).

Technique

This technique is a method for allowing a user to dial a phone number that spawns a data session for navigating a menu to the proper operator. It sets up an automated call center data/voice conversation between two communications devices (not necessarily wireless). The calling device is assumed Session Initiation Protocol (SIP) capable [1]. The caller dials the given automated call center number that is converted into an IP address by a DNS server. This allows the connection to a server that contains call center menus. This requires the calling party have the ability to render data and the called (possibly 3rd) party able to create the data. Good examples of rendering techniques are the wireless markup language (WML) [2] or HTTP [3]. Once the menu is navigated the call center menuing system can complete the call with the appropriate operator, or the operator can place a PSTN call. An example follows.

1. The process begins with the user entering a normal E.164 telephone number [4] to, say, a credit card company in an attempt to determine a discrepancy in his/her credit card bill. SIP initiates the call signaling with an INVITE message that goes to a SIP proxy. The session description protocol (SDP) [5] information in the SIP INVITE will describe the telephone and data capabilities of the caller.

2. The SIP proxy knows to send the number to a DNS server to see if there is a corresponding IP address and domain. This is a well-known operation between SIP and a DNS server.

3. The result is returned. If there is no IP address, the SIP proxy sends the INVITE to a circuit-switched gateway that will connect the call over the PSTN.

4. If there is an IP address, the INVITE message goes to the domain of the 3rd party that is handling the menuing system. Note that this is an extension to the case where the credit-card company is dialed directly. In this case it would be the responsibility of the credit-card company to ensure that the authoritative DNS server has the proper IP address for the corresponding E.164 number.

5. The INVITE continues on to the 3rd party server. Note that the 3rd party serve...