The Session Initiation Protocol (SIP) has been widely adopted for VoIP and Instant Messaging systems. However, the deployment has been primarily on a per-application basis. In this paper, we describe the design and implementation of a generic SIP client service and API that is available to all applications and provides unified mechanisms for functions like conference joins, pub-sub, call transfers etc. The service allows a user to plug in a softphone of his choice, and to use any one of multiple softphones, IP or PSTN phones on a per-session basis. We demonstrate the utility of this API by enhancing the user experience for existing applications as well as describing new converged applications. This client service is supported on the network side by prototyping a number of building blocks in addition to integrating a commercially available packet-audio mixer.
By: Aameek Singh, Priya Mahadevan, Arup Acharya, Zon-Yin Shae
Published in: RC23148 in 2004
LIMITED DISTRIBUTION NOTICE:
This Research Report is available. This report has been submitted for publication outside of IBM and will probably be copyrighted if accepted for publication. It has been issued as a Research Report for early dissemination of its contents. In view of the transfer of copyright to the outside publisher, its distribution outside of IBM prior to publication should be limited to peer communications and specific requests. After outside publication, requests should be filled only by reprints or legally obtained copies of the article (e.g., payment of royalties). I have read and understand this notice and am a member of the scientific community outside or inside of IBM seeking a single copy only.
Questions about this service can be mailed to reports@us.ibm.com .