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IVR - Using SIP for Flexible Integration in Call Centers
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February 26, 2009

Using SIP for Flexible Integration in Call Centers



By Berthold Hofmann, Chief Technology Officer


The Session Initiation Protocol (News - Alert) (SIP) establishes and terminates multimedia communication sessions such as voice and video over IP. It is rapidly becoming the standard telephony protocol in contact centers because it facilitates the integration of third-party applications, enables encryption and avoids the need to invest in expensive proprietary hardware. SIP also lets call centers protect their previous investments by helping to incorporate existing infrastructure with new solutions.

 
When two SIP-based VoIP systems are integrated, they usually must exchange customer-specific or application-specific “private” data in a seamless flow through the standard network infrastructure. Previously, this private transmission required opening specific IP ports throughout the whole series of network switches between the connected applications.
 
Other Methods for Transporting Private Data
Among the many possibilities for transporting private data inside the SIP signaling stream, two have been identified as the best for recording purposes because they enrich the recording with valuable data. The first method uses SIP extension headers (RFC 3327), and the second uses a modified payload within the SIP message.
 
SIP messages allow the addition of several extension headers capable of transporting private information. But these extension headers are incompatible with some VoIP systems so this approach is limited in its applications.
 
The second method, the SIP-INFO message described in RFC 2976, is the most efficient and flexible way for real-time transport of private data in the SIP signaling stream. This method also lets the integration partners freely define and change the payload type carried in the body of the message.
 
One payload type, XML, provides an especially powerful and flexible format. Its standardized architecture delivers clearly structured, customer-specific information to the connected system and may be adapted for nearly any customer requirement.
 
Integration partners have already successfully connected a communications recording and quality monitoring systems to a SIP-based telephony infrastructure by relying on the SIP-INFO message for integration. The confidential message processing is strictly contained within the user application residing on top of the unmodified SIP stack. A flexible analysis module to interpret the private message is implemented in the application, and users can easily modify it with a graphic-based development tool.
 
These privacy-enhanced communications systems now offer virtually unlimited functionality including real-time call tagging, recording with start/stop commands, online monitoring and enhanced search-and-replay.
 
Future Trends
In the future, new elements such as instant messaging, web chat, video conferencing, presence information and other real-time information services will be added as well, without radically changing the communication interfaces and data transportation paths.
 
A call center with a SIP and XML-based software infrastructure for its recording and quality monitoring system will be well positioned for future applications of unified communications (UC) and communication-enabled business processes (CEBP).
 
 
ASC (News - Alert) is a leading global provider of innovative solutions to record, analyze and evaluate multimedia-based communications. ASC’s solutions reveal vital information, enabling companies and organizations to considerably improve their value creation: contact centers enhance customer service, efficiently deploy staff and increase productivity. Financial institutions fulfill documentation requirements, achieve a higher level of legal security and reduce costs. First responders and public safety organizations enhance reactivity in emergency situations; therefore, ASC provides an important contribution to public safety.
 
For more information, visit www.asctelecom.com, call 201-252-3001, email info@asctelecom.com or write ASC, 1 International Boulevard, Suite #623, Mahwah, NJ 07495.
 



Edited by Greg Galitzine
Read More SIP Server Stories »
 

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