SDP for the WebRTC
draft-nandakumar-rtcweb-sdp-00

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Last updated 2012-10-15
Replaced by draft-ietf-rtcweb-sdp
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Network Working Group                                      S. Nandakumar
Internet-Draft                                               C. Jennings
Intended status:  Informational                                    Cisco
Expires:  April 18, 2013                                October 15, 2012

                           SDP for the WebRTC
                     draft-nandakumar-rtcweb-sdp-00

Abstract

   The Web Real-Time Communication (WebRTC) [WEBRTC] working group is
   charged to provide protocol support for direct interactive rich
   communication using audio, video and data between two peers' web
   browsers.  With in the WebRTC framework, Session Description protocol
   (SDP) [RFC4566] is used for negotiating session capabilities between
   the peers.  Such a negotiataion happens based on the SDP Offer/Answer
   exchange mechanism described in the RFC 3264 [RFC3264].

   This document serves a introductory purpose in describing the role of
   SDP for the most common WebRTC use-cases.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
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   This Internet-Draft will expire on April 18, 2013.

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   Copyright (c) 2012 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
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   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents

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   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  SDP and the WebRTC . . . . . . . . . . . . . . . . . . . . . .  3
   4.  Offer/Answer and the WebRTC  . . . . . . . . . . . . . . . . .  4
   5.  WebRTC Session Description Examples  . . . . . . . . . . . . .  5
     5.1.  Secure Two-Way Audio,Video and Data with RTCP Feedback . .  5
     5.2.  Secure Two-way Audio,Video,Data and remove data stream . . 11
     5.3.  Secure Two-Way Audio,Video w/Bundle  . . . . . . . . . . . 15
     5.4.  Successful One Way Session with 2 Video Streams  . . . . . 19
     5.5.  Add New Media (video)  . . . . . . . . . . . . . . . . . . 23
   6.  WebRTC <-> Legacy Interop Examples . . . . . . . . . . . . . . 28
     6.1.  Secure Two-Way Audio,Video w/Feedback - WebRTC <->
           Legacy Interop . . . . . . . . . . . . . . . . . . . . . . 28
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 33
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 33
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 33
     8.2.  Informative References . . . . . . . . . . . . . . . . . . 33
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 34

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1.  Introduction

   Javascript Session Exchange Protocol(JSEP) [JSEP] specifies a generic
   protocol needed to generate [RFC3264] offers and answers negotiated
   between the WebRTC peers for setting up, updating and tearing down a
   multimedia session.  For this purpose, SDP is used to construct
   [RFC3264] offers/answers for describing (media and non-media) streams
   as appropriate for recipients of a session description to participate
   in the session.

   The remainder of this document is organized as follows:  Section 3
   and 4 provide an overview of SDP and the Offer/Answer exchange
   mechanism.  Section 5 and 6 provide sample SDP usages for the most
   common WebRTC use-cases.
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