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Controlling Session Initiation Protocol (SIP)-Established Content Delivery Channels Using the Real-Time Streaming Protocol (RTSP)
draft-lindquist-sip-rtsp-02

Document Type Expired Internet-Draft (individual)
Expired & archived
Authors Jouni Maenpaa , Priya Rajagopal , Jan Lindquist , Xavier Marjou
Last updated 2012-07-12 (Latest revision 2010-06-09)
RFC stream (None)
Intended RFC status (None)
Formats
Stream Stream state (No stream defined)
Consensus boilerplate Unknown
RFC Editor Note (None)
IESG IESG state Expired
Telechat date (None)
Responsible AD (None)
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This Internet-Draft is no longer active. A copy of the expired Internet-Draft is available in these formats:

Abstract

The Session Initiation Protocol (SIP) is widely used for establishing multimedia sessions, whereas the Real Time Streaming Protocol (RTSP) is used in streaming media systems. RTSP has a dual role: it establishes a media session for the delivery of streaming media as well as controls the streaming session once it has been set up. Since SIP is also used for session establishment, there exists an overlap between the functionality provided by SIP and RTSP. In this document, we describe a model adopted by ETSI TISPAN, 3GPP, and Open IPTV Forum (OIPF) in which SIP and the SDP offer/answer model are used to set up a streaming session with an RTSP control channel and one or more media delivery streams. Such a model is beneficial since it allows the reuse of current architecture and functionality (e.g., authentication, charging, and QoS) established around SIP for RTSP- based streaming. The model benefits applications such as Internet Protocol Television (IPTV). In addition to the model, the document specifies two new variants of RTSP.

Authors

Jouni Maenpaa
Priya Rajagopal
Jan Lindquist
Xavier Marjou

(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)