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Signaling for Internet Telephony

Henning G. Schulzrinne; Jonathan D. Rosenberg

Title:
Signaling for Internet Telephony
Author(s):
Schulzrinne, Henning G.
Rosenberg, Jonathan D.
Date:
Type:
Technical reports
Department:
Computer Science
Permanent URL:
Series:
Columbia University Computer Science Technical Reports
Part Number:
CUCS-005-98
Publisher:
Department of Computer Science, Columbia University
Publisher Location:
New York
Abstract:
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.
Subject(s):
Computer science
Item views:
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