Voice over IP - Considerations for a Next Generation Architecture
EUROMICRO '05 Proceedings of the 31st EUROMICRO Conference on Software Engineering and Advanced Applications
Tracking anonymous peer-to-peer VoIP calls on the internet
Proceedings of the 12th ACM conference on Computer and communications security
Voice pharming attack and the trust of VoIP
Proceedings of the 4th international conference on Security and privacy in communication netowrks
Identity Federation for VoIP systems
Journal of Computer Security - Digital Identity Management (DIM 2007)
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Internet telephony must offer the standard telephony services. However, the transition to Internet-based telephony services also provides an opportunity to create new services more rapidly and with lower complexity than in the existing public switched telephone network (PSTN). The Session Initiation Protocol (SIP) is a signaling protocol that creates, modifies and terminates associations between Internet end systems, including conferences and point-to-point calls. SIP supports unicast, mesh and multicast conferences, as well as combinations of these modes. SIP implements services such as call forwarding and transfer, placing calls on hold, camp-on and call queueing by a small set of call handling primitives. SIP implementations can re-use parts of other Internet service protocols such as HTTP and the Real-Time Stream Protocol (RTSP). In this paper, we describe SIP, and show how its basic primitives can be used to construct a wide range of telephony services.