Network Working Group                                        C. Jennings
Internet-Draft                                                     Cisco
Intended status:  Standards Track                           J. Rosenberg
Expires:  May 2, 2012                              
                                                               J. Uberti
                                                                R. Jesup
                                                        October 30, 2011

                  RTCWeb Offer/Answer Protocol (ROAP)


   This document describes an protocol used to negotiate media between
   browsers or other compatible devices.  This protocol provides the
   state machinery needed to implement the offer/answer model (RFC
   3264), and defines the semantics and necessary attributes of messages
   that must be exchanged.  The protocol uses an abstract transport in
   that it does not actually define how these messages are exchanged.
   Rather, such exchanges are handled through web-based transports like
   HTTP or WebSockets.  The protocol focuses solely on media negotiation
   and does not handle call control, call processing, or other

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
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   Drafts is at

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on May 2, 2012.

Copyright Notice

   Copyright (c) 2011 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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   Provisions Relating to IETF Documents
   ( in effect on the date of
   publication of this document.  Please review these documents
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   it for publication as an RFC or to translate it into languages other
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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
   2.  Requirements and Design Goals  . . . . . . . . . . . . . . . .  5
   3.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  6
   4.  Protocol Overview  . . . . . . . . . . . . . . . . . . . . . .  6
   5.  Semantics & Syntax . . . . . . . . . . . . . . . . . . . . . .  8
     5.1.  Reliability Model  . . . . . . . . . . . . . . . . . . . .  8
     5.2.  Common Fields  . . . . . . . . . . . . . . . . . . . . . .  9
       5.2.1.  Session IDs  . . . . . . . . . . . . . . . . . . . . .  9
       5.2.2.  Seq  . . . . . . . . . . . . . . . . . . . . . . . . . 10
       5.2.3.  Session Tokens . . . . . . . . . . . . . . . . . . . . 10
       5.2.4.  Response Tokens  . . . . . . . . . . . . . . . . . . . 10
     5.3.  Media Setup  . . . . . . . . . . . . . . . . . . . . . . . 11
       5.3.1.  OFFER Message  . . . . . . . . . . . . . . . . . . . . 12  Offerer Behavior . . . . . . . . . . . . . . . . . 12  Answerer Behavior  . . . . . . . . . . . . . . . . 12
       5.3.2.  ANSWER . . . . . . . . . . . . . . . . . . . . . . . . 13  moreComing Flag  . . . . . . . . . . . . . . . . . 13
       5.3.3.  OK . . . . . . . . . . . . . . . . . . . . . . . . . . 14
       5.3.4.  ERROR  . . . . . . . . . . . . . . . . . . . . . . . . 14
     5.4.  Changing Media Parameters  . . . . . . . . . . . . . . . . 14
       5.4.1.  Conflicting OFFERS (glare) . . . . . . . . . . . . . . 15
       5.4.2.  Premature OFFER  . . . . . . . . . . . . . . . . . . . 17
     5.5.  Notification of Media Termination  . . . . . . . . . . . . 18
     5.6.  Errors . . . . . . . . . . . . . . . . . . . . . . . . . . 18
       5.6.1.  NOMATCH  . . . . . . . . . . . . . . . . . . . . . . . 18
       5.6.2.  TIMEOUT  . . . . . . . . . . . . . . . . . . . . . . . 19
       5.6.3.  REFUSED  . . . . . . . . . . . . . . . . . . . . . . . 19
       5.6.4.  CONFLICT . . . . . . . . . . . . . . . . . . . . . . . 19
       5.6.5.  DOUBLECONFLICT . . . . . . . . . . . . . . . . . . . . 19
       5.6.6.  FAILED . . . . . . . . . . . . . . . . . . . . . . . . 19
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 19
   7.  Companion APIs . . . . . . . . . . . . . . . . . . . . . . . . 19
     7.1.  Capabilities . . . . . . . . . . . . . . . . . . . . . . . 20
     7.2.  Hints  . . . . . . . . . . . . . . . . . . . . . . . . . . 20
     7.3.  Stats  . . . . . . . . . . . . . . . . . . . . . . . . . . 20
   8.  Relationship with SIP & Jingle . . . . . . . . . . . . . . . . 21
   9.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 21
   10. Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 21
   11. Open Issues  . . . . . . . . . . . . . . . . . . . . . . . . . 22
   12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 22
     12.1. Normative References . . . . . . . . . . . . . . . . . . . 22
     12.2. Informative References . . . . . . . . . . . . . . . . . . 22
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 23

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1.  Introduction

   This specification defines a protocol that allows an RTCWeb browser
   to exchange information to control the set up of media to another
   browser or device.  The scope of this protocol is limited to
   functionality required for the setup and negotiation of media and the
   associated transports, referred to as media control.  The protocol
   defines the minimum set of messages and state machinery necessary to
   implement the offer/answer model as defined in [RFC3264].  The offer
   answer model specifies rules for the bilateral exchange of Session
   Description Protocol (SDP) messages [RFC4566] for creation of media

   The protocol specified here defines the state machines, semantic
   behaviors, and messages that are exchanged between instances of the
   state machines.  However, it does not specify the actual on the wire
   transport of these messages.  Rather, it assumes that the
   implementation of this protocol would occur within the browser
   itself, and then browser APIs would allow the application's
   JavaScript to request creation of messages and insert messages into
   the state machine.  The actual transfer of these messages would be
   the responsibility of the web application, and would utilize
   protocols such as HTTP and WebSockets.  To facilitate implementation
   within a browser, messages are encoded in JSON [RFC4627].  This
   protocol, with appropriate selected transports, could also be
   implemented by a signalling gateway that converts ROAP to SIP or

   This protocol is designed to be closely aligned with the
   PeerConnection API defined in the RTCWeb API[webrtc-api]
   specification.  It is important to note that while ROAP does not
   require what has been referred to as a low level API for media
   manipulation, ROAP does not prevent having a such an API as well and
   both styles of API could coexist and be used where appropriate.

   The protocol defined here does not provide any call control.
   Concepts like ringing of phones, user search, call forwarding,
   redirection, transfer, hold, and so on, are all the domain of call
   processing and are out of scope for this specification.  It is
   assumed that the application running within the browser provides any
   call control based on the needs of the application, the scope of
   which is not a matter for standardization.

   Despite that fact that it has an abstract transport, ROAP is still a
   protocol.  This means it has state machines, and it has rules
   governing the behavior of those state machines which guarantee that
   system operates properly based on any set of inputs.  It is assumed
   that this state machinery is implemented in the browser and thus

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   immutable by the application, which can then guarantee proper
   behavior regardless of the operation of the resident JavaScript.

   The protocol is designed to operate between two entities (browsers
   for example), which exchange messages "directly" - meaning that a
   message output by one entity is meant to be directly processed by the
   other entity without further modification.  In practice, this means
   that a web server can treat ROAP messages as opaque and just shuffle
   them between browser instances.  This allows for simple
   implementations.  However, more powerful applications can be built in
   which the web server or JavaScript can modify the messages in order
   to provide more complex features.  As long as those modifications
   produce messages compliant to this specification, SDP Offer/Answer
   [RFC3264], SDP [RFC4566], ICE [RFC5245] and any other dependencies,
   interoperability is still possible.

   This protocol is designed for two major use cases:

   o  Browser to browser
   o  Browser to SIP device via a SIP gateway

   In the browser to SIP use case, the gateway obviously needs to be
   somewhat more sophisticated.  However, because this design is a small
   subset of the design space covered by SIP [RFC3261], it is intended
   to be simple to translate to and from/SIP via a signalling gateway.
   Moreover, many of the elements in messages have clear mappings to
   elements in SIP messages, thus allowing simple, stateless

2.  Requirements and Design Goals

   There has been extensive debate about the best architecture for
   RTCWeb signaling.  To a great extent this decision is dictated by the
   requirements that the signaling mechanism is intended to fit.  The
   protocol in this document was designed to minimize the amount of
   implementation effort required outside the browser and RTC-Web
   signaling gateways.  This implies the following requirements:

   It should be possible to develop a simple browser to browser voice
   and video service in a small amount of code.  In particular, it MUST
   be possible to implement a functional service such that:

   o  It's possible to build a web service that maintains only
      transaction state, not call state;
   o  In the browser to browser case, the web server can simply pass
      protocol messages between the browser agents without examining or
      modifying them;

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   o  The service operates without needing to examine the details of the
      browser capabilities (e.g., new codecs should be automatically
      accommodated without modifying either the service or the
      associated JS.

   It should be possible to implement a simple RTC-Web gateway that:

   o  Connects to legacy SIP devices ranging from multiscreen video
      phones to PSTN gateways;
   o  Has a deterministic mapping between RTC-Web messages and SIP
   o  Permits the mechanical translation of messages without knowledge
      of the details of all the browser capabilities;
   o  is only required to maintains transaction state, not call state
      (note is fine if an implementation want to maintain call state);
   o  Does not need to send or receive the media (unless also acting as
      a relay or a translator for codecs which are not jointly

   Finally it seems clear that SDP is too complicated to reinvent, so
   despite its manifest deficiencies we opt to take it as-is rather than
   trying to reinvent it.

3.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHOULD", "SHOULD NOT",
   "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be
   interpreted as described in [RFC2119].

   This draft uses the API and terminology described in [webrtc-api].

4.  Protocol Overview

   We start with a simple example.  Consider the case where browser A
   wishes to setup up a media session with browser B. At the high level,
   A needs to communicate the following information:

   o  This is a new media session and not an update to a different
   o  Here is A's SDP offer, including media parameters and ICE

   The OFFER message is used to carry this information.  For example, A
   might send B:

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     "seq": 1,
   o=- 2890844526 2890842807 IN IP4\n
   s= \n
   c=IN IP4\n
   t=2873397496 2873404696\n
   m=audio 49170 RTP/AVP 0"

   The messageType field indicates that this is an OFFER and the
   offererSessionId indicates the media session that this OFFER is
   associated with.  B can tell that this is for a new media session
   because it contains a offererSessionId that he has not seen before.
   The sdp field contains the offer itself, which is just an ordinary
   SDP offer rendered as a string.

   If B elects to start a media session, B responds with an ANSWER
   message containing SDP, as shown below.

     "seq": 1,
   o=- 2890844526 2890842807 IN IP4\n
   s= \n
   c=IN IP4\n
   t=2873397496 2873404696\n
   m=audio 49175 RTP/AVP 0"

   The contents of this message are more or less the same as those in
   the OFFER, except that B also includes a answererSessionId to
   uniquely identify the session from B's perspective.  The combination
   of offererSessionId and answererSessionId uniquely identifies this

   Finally, in order to confirm that A has seen B's ANSWER, A responds
   with an OK message.

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     "seq": 1

   Note that all of these messages contain a seq field which contains a
   transaction sequence number.  The seq field makes it possible to
   correlate messages which belong to the same transaction, as well as
   to detect duplicates, which is described later in section
   Section 5.1.

   The messageType value of "OFFER" will always contain an SDP offer,
   and an object with a messageType value of "ANSWER" will always
   contain an SDP answer.  The complete list of message types is defined
   in Section 5.  Only a small number of messages are permitted and much
   of the message set is devoted to error handling.

   In building web systems it is often useful for a request to contain
   some state that is passed back in future messages.  This system
   includes two types of state:  session state and request state.  If a
   browser receives a message that contains state in a setSessionState
   attribute, any future messages it sends that have the same
   offererSessionId MUST include this state in a sessionState attribute.
   Similarly if a request contains an setResponseState attribute, that
   state MUST be included in any response to that request in a
   responseState attribute.

   Once a session has been set up, additional rounds of offer/answer can
   be sent using the OFFER/ANSWER/OK sequence.  Note that the seq
   attribute makes it easy to differentiate these additional rounds from
   the initial exchange and from each other.

   At the point that one side which to end the session, it simply sends
   a SHUTDOWN message which is responded to with an OK response.  A
   SHUTDOWN can be sent regardless of it any response has been received
   to the initial OFFER.  The key purpose of the SHUTDOWN messages is to
   allow the other side to know they can clean up any state associated
   with the session.

5.  Semantics & Syntax

5.1.  Reliability Model

   ROAP messages are typically carried over a reliable transport (likely
   HTTP via XMLHttpRequest or WebSockets), so the chance of message loss

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   is low (though non-zero), provided that the signaling service is up.
   However, the common web reliability and scaleability model is based
   on the principle that transactions are idempotent and that requests
   can just be discarded and will be retried.  A retry of a transaction
   might happened if a given host was down and the DNS round robin
   approach wanted to move to the next server, or if a server was
   overloaded, or if there was a hiccup in the network.  Web
   applications that want to work well need to deal with theses issues
   to get the advantages of the general web design pattern for
   scaleability and reliability.  Because only the application knows
   what its internal reliability characteristics are, the JS application
   (and whatever associated servers it uses) are ultimately responsible
   for ensuring end-to-end delivery; the browser simply assumes that
   messages which are provided to the JS will be delivered eventually.

   However, in order to maintain OFFER/ANSWER transaction state, the SDP
   state machine does need to understand when the far end has received
   an ANSWER if it caused an error or not.  To support this model, OFFER
   and ANSWER messages are acknowledged end to end with an ANSWER or OK
   however any retransmission need to be handled by the JS or whatever
   is providing the transport of the ROAP messages.  The combination of
   the sessionID and seq allow the browser to detect and discard
   duplicate requests and to detect glare.

   NOTE:  The split of the reliability model between the JS and browser
      is something where implementations are playing around with and
      trying to get some experience with what works best.  This is an
      area that is highly likely to change as understanding of the
      implications evolves.

5.2.  Common Fields

5.2.1.  Session IDs

   Each call is identified by a pair of session identifiers:

   offererSessionId  The offerer's half of the session ID (supplied in
      the OFFER)

   answererSessionId  The answerer's half of the session ID (supplied in
      the response to an OFFER)

   The session ID values MUST be generated so that they are globally
   unique.  Thus, the combination of both sessionIds is itself globally
   unique.  Session IDs never change for the duration of an media

   All messages MUST contain the "offererSessionId", and all messages

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   other than OFFER or an error in response to an OFFER MUST contain
   both "offererSessionId" and "answererSessionId".

5.2.2.  Seq

   This is a sequence counter for the key requests that helps correlate
   responses to the correct request.

   This is a 32-bit unsigned integer.  On each new OFFER (from either
   browser) it is incremented by one.  The Seq of an OK or ANSWER is set
   to the same Seq that was used in the OFFER which caused it.  When a
   PeerConnection objects originates a new session by sending an OFFER
   type message, it starts the Seq at 1.

   Note:  If browser A starts an OFFER/ANSWER/OK transaction with a seq
      of 1 to browser B, then later B initiates a second
      OFFER/ANSWER?/OK transaction, it will have a seq of 2.

5.2.3.  Session Tokens

   While session IDs serve to uniquely identify a session, it may be
   useful to allow one or another sides to offload state onto the other
   side (for instance to enable a stateless gateway).  The
   "setSessionToken" and "sessionToken" fields are used for this
   purpose.  When an implementation receives a message with a
   "setSessionToken" field, it MUST associate the field value with the
   session.  For all future messages in the session MUST send the
   associated value in the "sessionToken" field (unless the session
   token is reset by another "setSessionToken" value).  If no session
   token has yet been received, the "sessionToken" field MUST be

5.2.4.  Response Tokens

   In addition to tokens which persist for the life of a session, it is
   also possible to have tokens which are only valid for the lifetime of
   a given request/response pair.  The "setResponseToken" and
   "responseToken" fields are used for this purpose.

   When an implementation responds to a message from the other side
   (e.g., supplies an answer to an offer, or replies to an answer with
   an OK), it MUST copy into the "responseToken" field any value found
   in a "setResponseToken" field in the message being responded to.  If
   no "setResponseToken" field is present, then the "responseToken"
   field MUST be omitted.

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5.3.  Media Setup

   In order to initiate sending media between the browsers, the offerer
   sends an OFFER message.  In order to accept the media, the answerer
   responds with an ANSWER message.  A sample message flow for this is
   shown below:

   participant OffererUA
   participant OffererJS
   participant AnswererJS
   participant AnswererUA
   OffererJS->OffererUA: peer=new PeerConnection();

   OffererJS->OffererUA: peer->addStream();
   OffererUA->OffererJS: sendSignalingChannel();
   OffererJS->AnswererJS: {"type":"OFFER", "sdp":"..."}
   AnswererJS->AnswererUA: peer=new PeerConnection();
   AnswererJS->AnswererUA: peer->processSignalingMessage();
   AnswererUA->AnswererJS: onconnecting();

   AnswererUA->OffererUA: ICE starts checking

   note right of AnswererUA: User decides it is OK to send video
   AnswererJS->AnswererUA: peer->addStream();
   AnswererUA->OffererUA: Media

   AnswererUA->AnswererJS: sendSignalingChannel();
   AnswererJS->OffererJS: {"type":"ANSWER","sdp":"..."}
   OffererJS->OffererUA: peer->processSignalingMessage();
   OffererUA->OffererJS: onaddstream();
   OffererUA->AnswererUA: Media

   AnswererUA->OffererUA: ICE Completes
   AnswererUA->AnswererJS: onopen();
   OffererUA->OffererJS: onopen();

   OffererUA->OffererJS: sendSignalingChannel();
   OffererJS->AnswererJS: {"type":"OK" }
   AnswererJS->AnswererUA: peer->processSignalingMessage();
   AnswererUA->AnswererJS: onaddstream();

   The above figure shows a simple message flow for negotiating media:

   o  The offerer sends an OFFER to initiate the call;
   o  At this point, ICE negotiation starts;
   o  Once the browser authorizes sending media to the far side, the
      answerer sends an ANSWER containing the media parameters; and

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   o  Once ICE is completed and an OK to the ANSWER is received, both
      sides know that media can flow.

   The contents of each of these messages is detailed below.

5.3.1.  OFFER Message

   The first OFFER message with a given offererSessionId is used to
   indicate the desire to start a media session.  Offerer Behavior

   In order to start a new media session, a offerer constructs a new
   OFFER message with a fresh offererSessionId.  The answererSessionId
   field MUST be empty.  Like all SDP offers, the message MUST contain
   an "sdp" field with the offerer's offer.  It MUST also contain the
   tieBreaker field, containing a 32 bit random integer used for glare
   resolution as described in Section 5.4.1.  Answerer Behavior

   A answerer can receive an OFFER in three cases:

   o  A new session (this is detected by seeing a new offererSessionId
   o  A retransmit of a new OFFER (known offererSessionId, empty
      answererSessionId); or
   o  A request to change media parameters (known offererSessionId,
      known answererSessionId, new seq value).

   The first two situations are described in this section.  The third
   case is described in Section 5.4.  Any other condition represents an
   alien packet and SHOULD be rejected with Error:NOMATCH

   If no media session exists with the given "offererSessionId" value,
   then this is a new media session.  The answerer has three primary

   o  Reject the request, either silently with no response or with an
      Error:REFUSED message;
   o  Reply to the OFFER message with a final ANSWER message; or
      Section 5.3.2
   o  Send back a non final ANSWER message and then later respond with
      an final ANSWER.

   In either of the latter two cases, the answerer performs the
   following steps:

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   1.  Generate a "answererSessionId" value;
   2.  Create some local call state (i.e., a PeerConnection object) and
       bind it to the "offererSessionId"/"answererSessionId" pair.  All
       future messages on this session MUST then be delivered to that
       PeerConnection object;
   3.  Start ICE handshaking with the offerer; and finally,
   4.  Respond with a message containing an SDP answer in the "sdp"
       field.  This will contain the answerer's (potentially with
       moreComing=true) media information and the ICE parameters.

   If an OFFER is received that has already been received and responded
   to and the media session still exists, then the answerer MUST respond
   with the same message as before.  If the session has been terminated
   in the meantime, then an Error:NOMATCH message SHOULD be sent.

5.3.2.  ANSWER

   The ANSWER message is used by the receiver of an OFFER message to
   indicate that the offer has been accepted.  The ANSWER message MUST
   contain the answererSessionId for this media session and an sdp
   parameter containing ICE candidates and the final media parameters
   for the session (although of course these can be adjusted by a new
   OFFER/ANSWER exchange.  See Section 5.4).  In addition, ANSWERs MAY
   contain the moreComing flag, as described below.  moreComing Flag

   This is a boolean flag that can only appear in an ANSWER and, if set
   to true, indicates that this answer is not the final answer that will
   be sent for the associated OFFER.  If this flag is not present, it is
   assumed to be false.

   One motivating use case for moreComing is where an Agent wishes to
   respond immediately to an OFFER in order to start ICE checking before
   the user has provided authorization to send media.  The Agent cannot
   send an ANSWER containing media information but can send ICE
   candidate.  In this case, the Agent could send an ANSWER that had
   moreComing=true but that allowed ICE to start.  Then later, when the
   user had authorized the media, the Agent could send an ANSWER with
   the moreComing flag=false that indicated this was the final media

   To see why simply having multiple independent offers (as opposed to
   multiple answers for a single offer), consider the case where browser
   A requests video with B. When the A side that sent the initial OFFER
   gets an ANSWER that rejects the video, it may very well present a UI
   indication that there is no media.  Five seconds later when browser B
   sends an OFFER requesting video, browser A may present a UI element

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   that asks is OK to do the video that was just rejected.  This results
   in a bad user experience and in the extreme can result in both sides
   always rejecting the other side's OFFER of video, then waiting for
   the user to authorize video that results in a new OFFER that is
   always rejected.

   It easier to be able to indicate that OFFER resulted in one valid
   ANSWER, but that the OFFER needs to be held open as other valid
   ANSWERS which would replace the current one.  This stops the other
   side from generating new a new OFFER while this is taking place.
   This is also needed to support a SIP gateway doing early media.

5.3.3.  OK

   The OK message is used by the receiver of an ANSWER message to
   indicate that it has received the ANSWER message.  It has no contents
   itself and is merely used to stop the retransmissions of the ANSWER.

5.3.4.  ERROR

   The ERROR message is used to indicate that there has been an error.
   The contents and semantics of this message are defined in
   Section 5.6.

5.4.  Changing Media Parameters

   Once a call has been set up, it is common to want to adjust the media
   parameters, e.g., to add video to an audio-only call.  This is also
   done with the OFFER/ANSWER/OK sequence of messages, though the
   details are slightly different.

   Either side may initiate a new OFFER/ANSWER exchange by sending an
   OFFER message.  However, implementations MUST NOT attempt this for
   sessions which are still in active negotiation.  Specifically, the
   offerer MUST NOT send a new OFFER until it has received the ANSWER,
   and the answerer MUST NOT send a new OFFER until it has received the
   OK indicating receipt of the ANSWER.

   A new OFFER MUST contain a complete set of media parameters
   describing the proposed new media configuration as well as a full set
   of ICE parameters.  The recipient of a new OFFER on a valid
   connection MUST respond with an appropriate ANSWER message.  However
   that message MAY refuse to accept the proposed new configuration.  If
   the session has been terminated in the meantime, then an Error:
   NOMATCH message SHOULD be sent.

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5.4.1.  Conflicting OFFERS (glare)

   Because a change of media parameters may be initiated by either side,
   there is a potential for the change requests to occur simultaneously
   (i.e., "glare").  This document defines a glare handling procedure
   that results in immediate resolution of the glare condition allowing
   one OFFER message to continue to be processed while the other is
   terminated.  It is defined in such a way that it can interwork with
   SIP's glare handling mechanism.  However SIP's timer based mechanism
   aren't suitable for the ROAP as strict requirements on ROAP message
   transport between end-points are not possible and thus easily could
   result in an repeated glare situation.

   To achieve immediate resolution each OFFER message includes a 32
   unsigned integer value, the tie breaker, that is randomly generated
   for each new OFFER message an end-point issues.  Whenever a end-point
   receives an OFFER message that has the same sequence number as an
   outstanding OFFER the end-point itself sent, a glare condition has
   arisen.  In a glare condition the end-point compares the received
   OFFER's tiebreaker value with the tiebreaker value of the tiebreaker
   in the OFFER outstanding.  The OFFER with the greatest numerical
   value wins and that OFFER is allowed to continue being processed.  IF
   the received OFFER lost the tie breaking an Error:CONFLICT message is
   sent.  If it is the outstanding OFFER that lost, the end-point can
   expect an Error:CONFLICT message to be eventually received.  However,
   that OFFER can immediately be considered as terminated.

   Some special considerations has been made in this glare handling for
   interworking well with SIP glare handling as currently specified.
   Thus it has the notion of a gateway that converts the ROAP message
   into SIP message.  This process is discussed in more detail below
   after the basic rules are defined normatively.

   A regular end-point SHALL generate a random 32-bit unsigned numerical
   value for each OFFER message.  In the case the random value becomes 0
   or 4,294,967,295 a new random value SHALL be generated until it is
   neither values.  The values 0 and 4,294,967,295 MAY be assigned to
   ROAP messages generated by gateways to ensure efficient glare
   handling towards other systems.

   An ROAP message end-point that has an outstanding OFFER, i.e. an
   OFFER where it has not yet received an ANSWER SHALL upon receiving an
   OFFER perform the following processing:

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   1  Check if the incomming OFFER has a answererSessionId, if not it is
      an initial offer.  If the outstanding OFFER also is an intial
      OFFER there is an Error.  If the outstanding OFFER is not an
      initial OFFER and the outstanding OFFER do have answererSessionId
      equal to the offererSessionId in the received message then the
      sequence numbers are checked.  In case the incomming OFFER's
      sequence number is equal to the sequence number of the outstanding
      OFFER there is glare.  If the sequence number is not the same and
      the sequence number of the incomming is larger than the
      outstanding OFFER's sequence number, then this message is out of
      order with an ANSWER to the out-standing message.  If the sequence
      number of the incomming is lower than the outstanding, then this
      is a old request.

   2  In case of glare, compare the tie-breaker values for each OFFER.
      The tie-breaker value that is greater than the other wins.  The
      OFFER with the winning value is processed as if there was no
      glare.  The OFFER with the losing value is terminated, see 3A or
      3B. In case the tie-breaker values are equal the double-glare case
      in 3C is invoked.

   3A The OFFER being terminated is the received one:  The end-point
      SHALL send a Error:CONFLICT response message.

   3B The OFFER being terminated is this end-points outstanding OFFER:
      The end-point knows the OFFER will be terminated and can expect an
      Error:CONFLICT response.  The end-point can assume this
      termination and MAY issue a new OFFER as soon as possible after
      having concluded the transactions for the winning OFFER.

   3C The two tie-breaker values where equal, in this case both OFFERs
      are terminated and a Error:DOUBLCONFLICT message is sent.  Both of
      the Offerer SHOULD re-attempt their offers by generating new OFFER
      messages, these messages SHALL have new tie-breaker values and
      incremented sequence number.  Also gateways SHOULD generate random
      values, as one reason for this double conflict is that two
      gateways have become interconnected and both selects either 0 or

   The following figure assumes the previous message flow has happened
   and media is flowing.

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  participant OffererUA
  participant OffererJS
  participant AnswererJS
  participant AnswererUA

  note left of OffererJS: "Hi, Let's do video"
  note right of AnswererJS: "Sounds great"
  OffererJS->OffererUA: peer->addStream( new MediaStream() );
  OffererUA->OffererJS: sendSignalingChannel();
  AnswererJS->AnswererUA: peer->addStream( new MediaStream() );
  AnswererUA->AnswererJS: sendSignalingChannel();
  OffererJS->AnswererJS: {"type":"OFFER", tiebreaker="123", "sdp":"..."}
  AnswererJS->OffererJS: {"type":"OFFER", tiebreaker="456", "sdp":"..."}
  AnswererJS->AnswererUA: peer->processSignalingMessage();
  OffererJS->OffererUA: peer->processSignalingMessage();

  OffererUA->OffererJS: sendSignalingChannel();
  AnswererUA->AnswererJS: sendSignalingChannel();
  OffererJS->AnswererJS: {"type":"ERROR",error="conflict","sdp":"..."}
  AnswererJS->OffererJS: {"type":"ANSWER", "sdp":"..."}
  AnswererJS->AnswererUA: peer->processSignalingMessage();
  OffererJS->OffererUA:  peer->processSignalingMessage();

  OffererUA->OffererJS: sendSignalingChannel();
  OffererJS->AnswererJS: {"type":"OK"}
  AnswererJS->AnswererUA: peer->processSignalingMessage();
  AnswererUA->AnswererJS: onaddstream();

  AnswererUA->AnswererJS: sendSignalingChannel();
  AnswererJS->OffererJS: {"type":"OFFER", tiebreaker="789", "sdp":"..."}
  OffererJS->OffererUA: peer->processSignalingMessage();
  OffererUA->OffererJS: sendSignalingChannel();
  OffererJS->AnswererJS: {"type":"ANSWER", "sdp":"..."}
  AnswererJS->AnswererUA: peer->processSignalingMessage();
  AnswererUA->OffererUA: Both way Video
  AnswererUA->AnswererJS: sendSignalingChannel();
  AnswererJS->OffererJS: {"type":"OK"}
  OffererJS->OffererUA: peer->processSignalingMessage();
  OffererUA->OffererJS: onaddstream();

5.4.2.  Premature OFFER

   It is an error, though technically possible, for an agent to generate
   a second OFFER while it already has an unanswered OFFER pending.  An
   agent which receives such an offer MUST respond with an Error:FAILED

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   message containing a "RetryAfter" attribute generated as a random
   value from 0 to 10 seconds.

5.5.  Notification of Media Termination

   The SHUTDOWN message is used to indicate the termination of an
   existing session.  Either side may initiate a SHUTDOWN at any time
   during the session, including while the initial OFFER is outstanding
   (i.e., before an ANSWER has been sent/received.)

   participant OffererUA
   participant OffererJS
   participant AnswererJS
   participant AnswererUA

   OffererJS->OffererUA: peer->close();
   OffererUA->OffererJS: sendSignalingChannel();
   OffererJS->AnswererJS: { "type":"SHUTDOWN" }
   AnswererJS->AnswererUA: peer->processSignalingMessage();
   AnswererUA->AnswererJS: onclose();

   AnswererUA->AnswererJS: sendSignalingChannel();
   AnswererJS->OffererJS: {"type":"OK"}
   OffererJS->OffererUA: peer->processSignalingMessage();
   OffererUA->OffererJS: onclose();

   Upon receipt of a SHUTDOWN which corresponds to an existing session,
   an agent MUST immediately terminate the session and send an OK
   message.  Subsequent messages directed to this session MUST result in
   an Error:NOMATCH message.  Similarly, on receipt of the OK, the agent
   which sent the SHUTDOWN MUST terminate the session and SHOULD respond
   to future messages with Error:NOMATCH.

5.6.  Errors

   Errors are indicated by the messageType "ERROR".  All errors MUST
   contain an "errorType" field indicating the type of error which
   occurred and echo the "seq" value (if any) and the session id values
   of the message which generated the error.  The following sections
   describe each error type.

5.6.1.  NOMATCH

   An implementation which receives a message with either an unknown
   offererSessionId (for an OFFER) or an unknown offererSessionId/
   answererSessionId pair SHOULD respond with a NOMATCH error.

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5.6.2.  TIMEOUT

   The TIMEOUT error is used to indicate that the corresponding message
   required some processing which timed out.  For instance, an agent
   which is a SIP gateway translates ROAP signaling messages into SIP
   messages.  If those SIP messages time out, the gateway would generate
   a TIMEOUT error.

5.6.3.  REFUSED

   An agent which has received an initial OFFER MAY indicate its refusal
   of the media session by sending a REFUSED error.  Note that this
   error is not required; an agent MAY simply drop the OFFER with no
   acknowledgement at all.  However, agents which do not wish to accept
   subsequent OFFERS SHOULD [OPEN ISSUE:  MUST?] send a REFUSED in order
   to avoid timeouts and confusion on the offerer side.

5.6.4.  CONFLICT

   The CONFLICT error is used to indicate that an agent has received an
   OFFER while it has its own OFFER outstanding.  The offerer's behavior
   in response to this error is defined in Section 5.4.1.


   The DOUBLECONFLICT error is used to indicate the tiebreaker values in
   CONFLICT were the same.  See Section 5.4.1.

5.6.6.  FAILED

   FAILED is a catch-all error indicating that something went wrong
   while processing a message.  A FAILED error MAY contain a
   "retryAfter" field, which indicates the time (in seconds) after which
   the message MAY be retried (though retries are OPTIONAL).

6.  Security Considerations


7.  Companion APIs

   Note:  This section may need to move to the requirements
      draft[I-D.ietf-rtcweb-use-cases-and-requirements] but for now it
      is convenient to put it here just to help see how all the pieces
      fit together.

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   The offer / answer concepts in this draft are not enough to meet all
   the use cases of RTCWeb.  They need to be combined with some
   additional functionality that the browser exposes to the JavaScript
   applications.  This additional functionality loosely falls into three
   categories:  capabilities, hints, and stats.  The capabilities allow
   the JS application to find out what video codecs and capabilities a
   given browser supports before initiating a media session.  The hints
   provide a way for the JS application to provide useful information to
   the browser about how the media will be used so that the browser can
   negotiate appropriate codecs and modes.  Stats provides statistics
   about what the current media sessions.  The capabilities, hints, and
   stats do not need to be communicated between the two browsers, so
   they are not specified in this draft.  However, this drafts assumes
   the existence of API so that these three can be used to build
   complete systems.  Some of the assumptions about these APIs are
   described in the following sections.

7.1.  Capabilities

   The APIs need to provide a way to find out the capabilities as
   defined in section 9 of RFC 3264.  This allows the JS to find out the
   codecs that the browser supports.

7.2.  Hints

   When creating a new PeerConenction in a browser, the application
   needs to be able to provide optional hints to the browser about
   preferences for the media to be negotiated.  These include:

   1.  Whether the session has audio, video, or both;
   2.  Whether the audio is spoken voice or music;
   3.  Preferred video resolution and frame rate (perhaps these just
       come from the MediaTrack objects);
   4.  Whether the video should prefer temporal or spatial fidelity;
   5.  <add more here>

   The JS applications should also be able to update and change these
   hints mid-session.  Some types of hint changes may simply impact the
   parameter on various codecs and require no signalling to the other
   end of the media stream.  Other types of hint changes may cause a new
   offer answer exchange.

7.3.  Stats

   Several parts of the media session create statistics that are
   important to some applications.  APIs should provide the JS
   applications with information on the following statistics:

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   1.  Total IP data rate for the session;
   2.  ICE statistics including current candidates, active pairs, RTT;
   3.  RTP statistics including codecs selected, parameters, and bit
   4.  RTCP statistics including packet loss rate; and
   5.  SRTP statistics.

8.  Relationship with SIP & Jingle

   The SIP [RFC3261] specifies an application protocol that provides a
   complete solution for setting up and managing communications on the
   Internet.  It combines both "call processing" functions - identity
   and name spaces, call routing, user search, call features,
   authentication, and so on - as well as media processing through its
   transport of SDP and support for the offer/answer model.

   In a web context, application processing can be done through
   proprietary logic implemented in Javascript/HTML, along with
   proprietary logic implemented in the web server, and proprietary
   messaging transported through HTTP and WebSockets.  One of the
   advantages of the web is to allow a rich set of applications to be
   built without changing the browser.  Although application processing
   and be done in JavaScript and the web servers, we do require raw
   media control in the browser.  ROAP basically extracts the offer/
   answer media control processing used in SIP, and puts it into an
   protocol that can operate independently of SIP itself.

   The information contained in ROAP messages corresponds closely to the
   offer/answer information carried by complete solutions such as SIP
   and Jingle, so it is straightforward to build gateways to and from
   ROAP.  These gateways need only translate the signaling, while
   allowing end-to-end media without the need for media relays (except,
   of course, for NAT traversal.)  In the case of SIP, which uses SDP
   directly, such gateways would translate between SIP and ROAP, while
   transporting SDP end-to-end.  In the case of Jingle [XEP-0166], it
   would also be necessary to translate between SDP and the Jingle
   offer/answer format; [XEP-0167] describes such a mapping.

9.  IANA Considerations

   This document requires no actions from IANA.

10.  Acknowledgments

   The text for the glare resoltuion section was provided by Magnus

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   Westerlund.  Many thanks for comment, ideas, and text from Eric
   Rescorla, Harald Alvestrand, Magnus Westerlund, Ted Hardie, and
   Stefan Hakansson.

11.  Open Issues

   How to negotiate support for enhancements to this JSON message.
   (consider supported / required )

   Common way to indicate destination in offer going to a signalling

   Need to generate proper ASCII art version of message flows.

12.  References

12.1.  Normative References

   [RFC4627]  Crockford, D., "The application/json Media Type for
              JavaScript Object Notation (JSON)", RFC 4627, July 2006.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              June 2002.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

12.2.  Informative References

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

              Ludwig, S., Beda, J., Saint-Andre, P., McQueen, R., Egan,
              S., and J. Hildebrand, "Jingle", XSF XEP 0166,
              December 2009.

              Ludwig, S., Saint-Andre, P., Egan, S., McQueen, R., and D.
              Cionoiu, "Jingle RTP Sessions", XSF XEP 0167,

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              December 2008.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              April 2010.

              Bergkvist, Burnett, Jennings, Narayanan, "WebRTC 1.0:
              Real-time Communication Between Browsers", October 2011.

              Available at

              Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use-cases and Requirements",
              draft-ietf-rtcweb-use-cases-and-requirements-06 (work in
              progress), October 2011.

Authors' Addresses

   Cullen Jennings
   170 West Tasman Drive
   San Jose, CA  95134

   Phone:  +1 408 421-9990

   Jonathan Rosenberg


   Justin Uberti
   Google, Inc.

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   Randell Jesup


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