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Using SIP for Flexible Integration in Call Centers  

The Session Initiation Protocol (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.

 

 

February 26, 2009

Reposted by Uni-Ta | Supplier of IP Phone, IP PBX, Cards for Asterisk, VoIP Adapter.

 

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