Network Working Group                                         R. Raymond
Internet-Draft                                               E. Lagerway
Intended status: Informational                                 Hookflash
Expires: January 07, 2014                                I. Baz Castillo
                                                              R. Shpount
                                                           July 06, 2013

                 WebRTC JavaScript Object API Rationale


   This document describes the reasons why a JavaScript Object Model
   approach is a far better solution than using SDP [RFC4566] as a
   surface API for interfacing with WebRTC.  The document outlines the
   issues and pitfalls as well as use cases that are difficult (or
   impossible) with SDP with offer / answer [RFC3264], and explains the
   benefits and goals of an alternative JavaScript object model

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
   working documents as Internet-Drafts.  The list of current Internet-
   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 January 07, 2014.

Copyright Notice

   Copyright (c) 2013 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
   Provisions Relating to IETF Documents
   ( in effect on the date of

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   publication of this document.  Please review these documents
   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
     1.1.  Terminology . . . . . . . . . . . . . . . . . . . . . . .   4
   2.  Issues with a Universal Session Description Format (and Offer
       / Answer) . . . . . . . . . . . . . . . . . . . . . . . . . .   4
     2.1.  Goal of Minimized Requirements  . . . . . . . . . . . . .   6
     2.2.  Offer / Answer State Machine  . . . . . . . . . . . . . .   7
       2.2.1.  Offer / Answer Violations . . . . . . . . . . . . . .   8
     2.3.  Browser to Browser Format Compatibility Issue . . . . . .   8
     2.4.  Browser to JavaScript Compatibility Issues  . . . . . . .   9
     2.5.  SDP as a surface API for JavaScript developers  . . . . .   9
     2.6.  Is SDP allowed to be mangled? . . . . . . . . . . . . . .  10
     2.7.  SDP errata and bugs compatibility issues  . . . . . . . .  11
       2.7.1.  SDP Bugs Become Enshrined . . . . . . . . . . . . . .  11
     2.8.  SIP/SDP compatibility worsened  . . . . . . . . . . . . .  12
     2.9.  Increased surface API . . . . . . . . . . . . . . . . . .  12
     2.10. Impossible API to implement to achieve browser
           compatibility . . . . . . . . . . . . . . . . . . . . . .  13
       2.10.1.  Example Oddities That Need Definition  . . . . . . .  13
     2.11. Plan A, Plan B vs NoPlan  . . . . . . . . . . . . . . . .  14
     2.12. SIP Forking Issue . . . . . . . . . . . . . . . . . . . .  15
   3.  Alternatives to Fixing these Issues Now . . . . . . . . . . .  15
     3.1.  Waiting for WebRTC 2.0  . . . . . . . . . . . . . . . . .  15
       3.1.1.  Cost now to fix versus fixing later . . . . . . . . .  16
       3.1.2.  If starting over, would even SIP people want SDP as a
               surface API?  . . . . . . . . . . . . . . . . . . . .  16
       3.1.3.  Incremental Approach may make Compatibility Worse . .  16
     3.2.  Session Description Format Construction API . . . . . . .  17
   4.  Example Difficult Usage Cases with Current Model  . . . . . .  19
     4.1.  On / off hold example usage case  . . . . . . . . . . . .  19
     4.2.  One-Sided Constraints Negotiation use Case Scenario . . .  20
     4.3.  Meet-me Negotiation Use Case Scenario . . . . . . . . . .  22
     4.4.  Browser to Browser Compatibility Extension Compatibility
           Issue Scenario  . . . . . . . . . . . . . . . . . . . . .  22
     4.5.  Building Interoperability between WebRTC and a SIP
           Service Scenario  . . . . . . . . . . . . . . . . . . . .  23
     4.6.  Bit-rate Change Scenario  . . . . . . . . . . . . . . . .  24
     4.7.  Video Codec Option Change Scenario  . . . . . . . . . . .  25
     4.8.  Video Upgrade Scenario  . . . . . . . . . . . . . . . . .  25
   5.  Proposal: WebRTC JavaScript Object Model  . . . . . . . . . .  26

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     5.1.  Overview  . . . . . . . . . . . . . . . . . . . . . . . .  26
     5.2.  Benefits  . . . . . . . . . . . . . . . . . . . . . . . .  26
       5.2.1.  Greater compatibility . . . . . . . . . . . . . . . .  26
       5.2.2.  Easier to extend  . . . . . . . . . . . . . . . . . .  26
       5.2.3.  Faster Reaction Time To Issues  . . . . . . . . . . .  27
       5.2.4.  Decreased surface API . . . . . . . . . . . . . . . .  27
       5.2.5.  Greater compatibility for SIP . . . . . . . . . . . .  27
       5.2.6.  Alternative formats . . . . . . . . . . . . . . . . .  28
     5.3.  Design Goals and Considerations . . . . . . . . . . . . .  28
       5.3.1.  Objects Model Kept Simple . . . . . . . . . . . . . .  28
       5.3.2.  Simple to Gather Negotiation Information  . . . . . .  28
       5.3.3.  Offer / Answer  . . . . . . . . . . . . . . . . . . .  28
       5.3.4.  Extensions  . . . . . . . . . . . . . . . . . . . . .  28
       5.3.5.  Well Defined Behaviors  . . . . . . . . . . . . . . .  29
       5.3.6.  Data Channel  . . . . . . . . . . . . . . . . . . . .  29
       5.3.7.  Satisfy the expectations of the RTCWEB charter  . . .  29
       5.3.8.  SIP/SDP and current WebRTC API shim compatibility
               statement . . . . . . . . . . . . . . . . . . . . . .  29
       5.3.9.  Greater Separation of RTCWEB Working Group and Other
               Working Groups  . . . . . . . . . . . . . . . . . . .  30
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  30
   7.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  30
     7.1.  Normative References  . . . . . . . . . . . . . . . . . .  30
     7.2.  Informative References  . . . . . . . . . . . . . . . . .  31
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  32

1.  Introduction

   While the IETF RTCWEB WG is not specifically tasked with providing an
   API by the W3C, the group has effectively defined a surface API with
   the mandate to use SDP [RFC4566] with offer / answer [RFC3264].

   SDP is a condensed text based format that typically describes all of
   the real-time media streams, networking properties, codecs, media
   state and media attributes.  SDP is completely extensible and can be
   used to describe absolutely anything so long as it is formatted
   correctly within its minimally defined limitations.

   The points for mandating SDP with an offer / answer API typically
   boils down to:

   1.  It's really easy to establish communication, especially with SIP

   2.  The decision was already made.

   3.  SDP yields greater compatibility (especially with SIP networks).

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   4.  We must have some kind of universal exchange format.

   5.  There is no alternative to this approach except destroying
       everything created and starting from scratch.

   This document will explain why these reasons are insufficient to
   continue with an SDP with offer / answer mandate approach given
   strong logical arguments and reasons with real world scenarios where
   this approach fails and due in no small part to its lasting
   consequences (including negative consequences for SIP).

   The document highlights the benefits and goals for a different
   "JavaScript Object Model" approach, which satisfies the RTCWEB WG
   charter's requirements, yields greater compatibility and offers a
   road-map where future potential extensions can be readily added
   without breaking existing implementations.

   A "JavaScript shim" is described including details on how it can
   offer a wrapped API around a core WebRTC JavaScript Object Model.
   This Shim will provide the same level of "ease of use" as experienced
   with the current SDP WebRTC API.  However, this JavaScript shim is
   not mandatory to use for those who do not require an "SDP with offer
   / answer" model.

1.1.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [RFC2119].

2.  Issues with a Universal Session Description Format (and Offer /

   The issue with SDP is not the expressiveness of the format but its
   usage as an arbitrary universal format and an API surface instead of
   providing JavaScript developers an object model they can readily
   understand.  JavaScript could be used to control the plumbing of
   media objects using familiar JavaScript expressive concepts enshrined
   with methods, properties and events.  Today, in many real-world use
   cases, controlling WebRTC requires modifying SDP directly.

   Requiring JavaScript developers to serialize their API control
   requests into a text format (via modifications of SDP existing blobs)
   is only one aspect of the many issues the SDP approach creates for
   developers.  Needlessly, an offer / answer state machine is imposed
   on JavaScript developers as well.

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   While the currently mandated SDP based API allows developers to
   quickly implement basic calling demos and interoperability with some
   SIP networks, it has many issues that will be explored and explained
   in this document and include (but not limited to):

   1.   Defining a standard universal all-encompassing session
        description format for use with WebRTC that describes all
        connections, media, constraints, streams and tracks for all
        scenarios is especially challenging.

   2.   Rather than focusing and defining the properties needed for
        communication, the focus is put on the best way to express the
        format where every nuance and behavior will need to be detailed
        for any browser vendor to capably implement the SDP based WebRTC

   3.   The bar for browsers (or other applications with WebRTC engines)
        to produce a WebRTC engine is raised substantially by forcing
        the browser to implement an entire SDP offer / answer engine
        too, with little to no added benefit.

   4.   A universal format built into the browser's API is entirely
        unneeded and goes well beyond the RTCWEB chartered mandate for
        the RTCWEB Working Group.

   5.   A flexible and expendable universal exchange format leads to
        greater interpretations and mistakes in various implementations,
        which in turn leads to increased incompatibilities.

   6.   Given the format is entirely flexible and open to
        interpretation, resulting implementations will more likely be
        prone to errors relative to the other truly needed aspects of
        RTC (which have better defined boundaries, behaviours, and

   7.   Mistakes in the format won't be fixed until a new browser binary
        update is released and deployed amongst users.

   8.   Mistakes in implementation of the session description format can
        become enshrined and difficult to deprecate (for the sake of

   9.   Compatibility issues caused by the format will not be limited to
        browsers-only as many hybrid browser-engine based applications
        now exist too.

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   10.  Using alternative signaling formats will require complete
        understanding of the universal format to be able to translate it
        into other alternative signaling formats.

   11.  JavaScript (or proxies) will need to parse and rewrite the
        output session description format with 100% precision and
        without loss.  They will also require pre-knowledge of what each
        browser produces and expects, despite the likelihood of a
        multitude of outputted flavors, on various platforms, and from
        version to version and despite the inability to easily predict
        or detect the variants.

   12.  JavaScript developers trying to control WebRTC features will
        need to manipulate any defined universal format rather than
        interacting with JavaScript objects.

   13.  Offer / answer is mandated and the state machine is required but
        the exact rules and violations of the rules ill defined when
        used within WebRTC.

   14.  The rules of how a universal format can be modified before being
        delivered to remote parties need to be meticulously defined or
        compatibility issues will arise (including the allowed rules of
        post browser format regeneration as to what can be modified and
        fed back into the browser).

   15.  Due to the issues defined above, SIP compatibility will worsen,
        not strengthen.

   An alternative to all of the issues caused by a universal format and
   state machine are described later in the document.  This alternative
   allows JavaScript to control the behavior of the media engine's
   plumbing while providing extensible and modifiable shims written
   entirely in JavaScript that produce consistent signaling and exchange
   formats for the specific network where those formats operate.

2.1.  Goal of Minimized Requirements

   While the primary goal of WebRTC is to enable browser to browser
   communication, the definition of a "browser" is ever expanding.
   Beyond just traditional hand-held applications, hybrid applications
   that are part HTML-5 and part native code exist.  Servers will become
   as much as part of the WebRTC infrastructure as browsers.  Minimizing
   the requirements to the basic wire compatibility necessary to achieve
   RTC is essential for maximum compatibility, flexibility and varying
   usage scenarios.

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   The mandate for the RTCWEB charter is to simply define requirements,
   provide basic "on-the-wire" compatibility, and define security
   requirements (such as enforcing ICE connection agreements).  The
   RTCWEB charter goals have been exceeded by going well beyond that
   scope by mandating an API that works fine for simple SIP
   interoperability demos but does not provide easy compatibility to the
   basic constructs needed as outlined from the charter for use with
   other on-the-wire signaling protocols (other than SIP).  If SIP is
   the only end goal of the WG, then that goal must be specifically
   stated rather than effectively mandated by making alternative
   signaling approaches unreasonably difficult to achieve.

2.2.  Offer / Answer State Machine

   The current SDP approach requires an offer / answer state machine.
   Mandating an offer / answer state machine implies that:

   1.  SDP be generated by browser A and sent to browser B

   2.  Browser B must respond with the offer with an answer

   3.  If either party issues a new offer but the offer is rejected, the
       state must revert to the previous agreed SDP (or fail to none)

   4.  If one side receives an offer while the other side has an
       outstanding offer, a conflict occurs and both sides must reject
       and revert and perform SDP conflict resolution to issue an offer

   5.  The only changes to the media that are allowed happens if both
       sides agree

   6.  Any change required to the SDP requires a network round trip
       where both sides mutually agree (at least as traditionally
       defined in offer / answer but the rules are in flux)

   This offer / answer model is defined as required with the current
   implementation.  Not only do the browser vendors have to enforce the
   rules, all JavaScript authors must also adhere to these rules of
   signaling.  While WebRTC does not dictate the signaling mechanism
   between browsers, effectively it is imposing this signaling state
   machine on all implementations (which is not a mandate of the RTCWEB
   Working Group).

   There are other models for signaling other than offer / answer.  For
   example, one-sided constraints based negotiation is an alternative
   model.  This type of negotiation requires each side to determine what
   it wants to receive independent of the other.  This signaling is akin

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   to saying "if you plan to send anything, make sure it conforms to the
   following".  Changes to the media may occur without agreement from
   the remote party where each side decides what is acceptable to
   receive without agreement from the other.  The remote side can decide
   if it wants to send within those constraints or not.  There is no
   round trip offer / answer required in this model to affect change.

   Offer / answer introduces the unnecessary asynchronism to the API and
   JavaScript implementations.  For example, changing the list of codecs
   expecting to receive or the current sending codec can be done
   immediately without the need for asynchronous calls.

   Offer / answer is not required to achieve RTC wire compatibility but
   it is currently mandated when alternatives could exist.

2.2.1.  Offer / Answer Violations

   The offer / answer SDP state machine is already violated in WebRTC.
   Trickle ICE precludes offer / answer round trips and other proposed
   standards like NoPlan [I-D.ivov-rtcweb-noplan] suggest relaxing the
   offer / answer model even more.  The rules of what offer / answer at
   this point is undefined and in clear violation of the strict previous
   rules without clear direction on what exactly constitutes offer /
   answer anymore and where it should and should not be used.

   A new state for offer / answer called PRANSWER is now defined, which
   did not exist as part of the standard offer / answer state machine.
   Offer rollback is not adequately defined either should an offer /
   answer conflict occur.

   Currently, switching codecs requires an SDP offer / answer should
   perform a round trip even though it is not technically needed for an
   RTC engine to change codecs.  Should this be another exception to the
   offer / answer state machine?

2.3.  Browser to Browser Format Compatibility Issue

   SDP is a flexible format, and it allows many alternative methods to
   express the same intentions.  The smallest change may alter the SDP's

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   This creates a parsing and SDP generation compatibility issues.  If
   SDP is packaged by JavaScript and delivered to the remote browser
   then each browser must support every single possible variant of SDP
   for every browser version and platform in existence.  They must do
   this without failure.  To maximize compatibility, a browser should
   generate the SDP format in the variant expected by the remote party
   (despite not having sufficient knowledge about the remote party to
   provide the correct SDP).

2.4.  Browser to JavaScript Compatibility Issues

   Since WebRTC is not supposed to mandate the format on the wire for
   signaling, one supported use case for WebRTC must be allowing the
   browser generated SDP to be converted into alternative on-the-wire
   formats.  This SDP conversion may be performed by JavaScript in the
   browser, or later by an intermediate gateway.  In either case, the
   converter must be entirely aware of all variants to the SDP possible
   from every browser platform and version, despite browser version
   detection being heavily frowned upon by industry best practices.
   Likewise, the JavaScript or gateway must know how to generate the
   correct SDP for all browsers and versions before passing the
   serialize SDP blob into the browser.  Generating compatible SDP may
   be impossible unless the exact formats and restrictions are
   unquestionably clear by all implementers of the specification (which
   is anything but clearly described in the current WebRTC SDP based API
   that developers are mandated to use).

2.5.  SDP as a surface API for JavaScript developers

   The current SDP based API is limited to placing a call and answering
   a call and adding media.  To perform common edge cases or to utilize
   RTC features beyond the basic API typically requires SDP mangling.

   Many of the operations from JavaScript to control or fetch properties
   from RTC will be through serialization to / from the SDP instead of a
   developer using familiar JavaScript language constructs (e.g. object
   methods, structures, properties and events).  The JavaScript
   developer must learn an entirely new protocol called "SDP" and be
   able to parse and generate not only basic SDP but any SDP extensions
   without introducing a single compatibility issue.

   Examples; A JavaScript developer wants to hold / un-hold media
   streams.  The developer must use a widely adopted but hidden feature
   to parse the SDP from the browser, change it to add the appropriate
   "hold" state, send that hold state to the remote side, wait for the
   "answer" to accept the hold, parse the result on the return to see if
   the hold was accepted and feed the result to the browser.

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   Worse, a flood of extensions to SDP for WebRTC are being written to
   "enhance" and "extend" the functionality of the browser with new
   features.  Many basic things are ill defined in the current SDP based
   API, for example, changing non-negotiated codec parameters, such as
   codec bandwidth.

   There is no facility for JavaScript to detect what SDP the browser is
   currently using or capable of delivering.  The developer has no idea
   of the extensions available, or what SDP will be produced, or what
   SDP is compatible.  The developer's JavaScript code must be able to
   handle everything generated by the browser for any use case beyond
   basic call, answer and hang-up.  This is a heavy burden to place on a
   JavaScript developer who is not familiar with the details of RTC
   concepts as expressed in SDP, and is a challenge even for those who
   are familiar.

   Effective APIs are meant to be contracts between a producer and
   consumer, whereas this SDP methodology offers little in the form of
   any such contract.

   If SDP is to become standardized for use with WebRTC then JavaScript
   developers must learn SDP to use RTC's available features and build
   new features.  Alternatively, accessors will need to be provided to
   manipulate the SDP on behalf of the JavaScript (and if so, then why
   not move to an object model straight away and do away with SDP?).

2.6.  Is SDP allowed to be mangled?

   The choice must be made if SDP may be modified or not.  If
   modifications are the only way to achieve RTC features available then
   what is allowed to be modified must be clearly defined in exact
   detail and the expected behavior of each feature (and modification of
   each feature), as expressed in SDP, must be defined.  Anything short
   of exact specifications will cause incompatibility.  Again, the
   implication is that Web / JavaScript developers must learn SDP to
   utilize the available RTC features and they must learn the rules of
   modification equally well, which virtually do not exist at all today.

   If the choice is to not allow complete SDP modification at all, then
   the protocol becomes extremely tied to SDP based protocols like SIP.
   Yet, there is no mandate for SIP to be the standardized protocol in
   WebRTC.  In fact, the mandate to require SIP was explicitly denied,
   which presents the argument that SDP manipulation must be allowed.

   The SDP mangling issue isn't just an issue when the format is sent
   on-the-wire.  If Browser A sends Browser B an SDP, the current
   philosophy is that the SDP is allowed to be modified.  However, there
   is the possibility of modifying the SDP generated by Browser A and

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   giving that modified SDP back to Browser A to change it's behavior
   (i.e. a serialized text based API call) before the offer is given to
   Browser B (and likewise with Browser B when it responds with its SDP

   How much of the SDP is allowed to be modified before giving the SDP
   back to the local browser?  SDP is a free-form format so anything can
   theoretically get changed, but should it be allowed?  If not, what
   can and cannot be modified?  CODECS?  SSRC?  SDES?  Fingerprints?
   Transports?  M-lines?  And so on...

   This issue becomes further compounded when extensions are factored in
   as well.

2.7.  SDP errata and bugs compatibility issues

   With the SDP baked into the browser binary, the only way SDP
   compatibility issues can be fixed is by releasing a new browser
   update, and the JavaScript developers must support or work around
   flaws until the browser vendors deliver the fix and the user base
   upgrades their browsers.

   While it could be argued that any bug must be worked around, SDP is a
   unique problem.  SDP is a free-form format.  Being compatible isn't
   as easy as implementing a limited wire protocol for media transport
   or a API contract with well defined features and attributes.  The
   likelihood of free-form SDP containing errors is far greater than a
   typical well defined API due to SDPs many flavors, interpretations
   and lack of strong definition.

2.7.1.  SDP Bugs Become Enshrined

   To illustrate a scenario:

   1.  Browser Vendor A has a bug

   2.  Browser Vendor B can't work with A because of the bug so it
       implements a "work around"

   3.  Browser Vendor A fixes the bug but implements a work around to be
       compatible with Browser Vendor B's "work around"

   This situation demonstrates is how browser bugs can become enshrined
   as there's no way to update the SDP produced by the browser binary
   once it's released until the next update release cycle occurs.  This
   would not be true if JavaScript was used via a shim to produce SDP as
   JavaScript can be dynamically updated as needed at any time and a
   service provider can choose to update their JavaScript implementation

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   to exacting expectations for their network regardless of the browser

   The lower level RTC wire protocols that need to be mandated by the
   RTCWEB Working Group have limited scopes and well defined behaviors.
   Any mistakes are obvious, likely to present very rapidly, and easy to
   spot which party is doing something wrong and much easier to fix
   earlier as a result.  This is not true with a free form highly
   descriptive language for sessions.  The combinations are limitless
   and every scenario is difficult to test, especially in concert with
   every other browser vendor with every version released.  The session
   description will be the likely place of failure across the browsers
   when the session description is generated inside the browser's

2.8.  SIP/SDP compatibility worsened

   One of the main arguments for using SDP with offer / answer was
   supposed to be ease of compatibility with existing signaling
   networks, like SIP.  Instead, variations in the browser's SDP will
   likely worsen SIP compatibility instead of enhance it.

   A SIP provider must now be compatible with every browser's SDP on
   every platform and version and the browser's SDP must be compatible
   with every SDP from a SIP network.  Alternatively, JavaScript or SBCs
   (Session Border Controller) must be used to re-write any incompatible
   SDP to be compatible.  However, this moves the problem from the
   browser to JavaScript, or requires SBCs to "fix" the problem.

   Had SDP been entirely generated by JavaScript rather than come from
   the browser engine, the JavaScript could create only SDPs compatible
   with a particular SIP provider under control of their own JavaScript
   and the SIP provider could chose which JavaScript SDP parsing /
   generation code to run, for maximum compatibility.

2.9.  Increased surface API

   By mandating SDP, the requirement for compatibility with WebRTC is
   increased substantially with little benefit.  Instead of just
   supporting basic media RTP [RFC3550], STUN/ICE/TURN [RFC5389]/
   [RFC5245]/[RFC5766], DTLS [RFC6347] and CODECS an additional bar must
   be passed, i.e. a browser or other WebRTC compliant API must support
   SDP with a full offer / answer state machine (or a state machine with
   additional rules to make it flexible for various scenarios).

   With an alternative approach, the entire requirement for SDP could be
   removed without any loss of compatibility or increase in complexity
   while achieving greater compatibility via the JavaScript shim.

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2.10.  Impossible API to implement to achieve browser compatibility

   The current mandated SDP based API cannot be implemented as a
   standard by independent browser vendors in its current form.  A list
   of subsequent behaviors regarding the usage, parsing, handling,
   extensions, behaviors, constraints and other such reference documents
   must be meticulously defined for SDP with the modified offer / answer
   state machine or no browser can ever claim to be "compliant".  The
   current definition process is far from complete.

   The current WebRTC SDP based API is far from achieving that goal due
   to the inclusion of free-form SDP with offer / answer and it is
   grounds for removing it as it goes beyond the RTCWEB's charter and
   limited scope.

   Any incremental approach that does not remove the offer / answer
   model requirement yields a road block to achieving alternative WebRTC
   signaling protocols other than SIP.

   An alternative WebRTC JavaScript object model approach that does not
   require an all-encompassing session description and related state
   machine is being proposed as an alternative solution so the RTCWEB
   charter can complete its defined goals in a timely fashion.

2.10.1.  Example Oddities That Need Definition

   There are many oddities in the SDP RFC [RFC4566] and the various
   related extensions.

   For example; will RTP CODEC maps be required or not?  They are not
   required for basic CODECs according to the SDP RFC.  However, with
   all the flavors of CODECs being offered, defining a mapping between
   payloads is critical to compatibility and not just a good idea.

   Another example; should "t=0 0" be respected?  Is that allowed to be
   changed?  Do the browser vendors need to enforce the attribute, or
   should the JavaScript layer enforce it?  Should the streams wait to
   start until the NTP time stamp and close when the NTP time completes?

   These are just small samples of questions that must all be completely
   addressed in detail.  This could also cause a cascade of updated
   reference drafts and confusion as to which version is to be adhered
   by browsers as well as what each browser specifically supports.
   Nominally referencing the SDP RFC will not be sufficient, and deltas
   from the established standards when violated will need to be defined
   when the rules change.

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2.11.  Plan A, Plan B vs NoPlan

   At the time of authoring this document, three plans on how to handle
   large number of media streams in SDP have emerged currently under
   consideration from the IETF, referred to as PlanA
   [I-D.roach-rtcweb-plan-a], PlanB [I-D.uberti-rtcweb-plan] and NoPlan

   PlanA and PlanB acknowledge that using SDP as it is historically
   defined in SIP is inefficient and problematic for large number of
   media streams, especially factoring in that each media line must have
   its own unique ports.

   NoPlan allows for media to be described in a more JavaScript friendly
   way and goes a long way towards improving the situation from SDP by
   taking out the mapping of the streams from the SDP but does not
   remove the reliance upon SDP.  This creates a dual format system
   where some information is initially carried over SDP and other
   information is signaled through an alternative approach (including
   the possibility of SDP offer/answer).  NoPlan could have been the
   sufficient approach if it took one step further and removed SDP

   PlanA, PlanB and NoPlan are a perfect example of why not to use SDP
   as the basis for WebRTC.  SDP has some arbitrary limitations as a
   description protocol for multiple streams whereas no such limitations
   exist at the lower layer transports themselves.  RTP allows for
   multiplexing multiple SSRCs.  In other words, the problem is SDP, not
   the real time transportation technologies.

   These drafts illustrate the limitations of SDP and attempt to solve
   it by introducing even more complex descriptions around SDP and / or
   by "relaxation" of the offer answer model combined with altering the
   description language of SDP.

   None of these drafts address most of the concerns outlined in this
   draft.  If anything, they further illustrate how divergent the SDP
   will become as more and more effort is put into working around
   problems inherent to the nature of utilizing SDP (or any universal

   The issue that SDP implementers face should be isolated to those who
   require SDP for their signaling protocols (namely SIP) where they can
   choose the best practices for their networks for interoperability.
   These complex approaches do not have to be forced on other signaling
   protocols that do not have or require such limitations.

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   Certainly JavaScript programmers and the W3C should not be impacted
   by such limitations by introducing SDP (or any universal format) into
   the mix when it adds zero value and fails in its primary objectives,
   namely: interoperability with existing SIP vendors & networks.

   This further illustrates why SDP baked into the browser binary is not
   beneficial for SIP vendors either.  They will be forced to upgrade
   their SIP infrastructure to support SDP packets from browsers with
   these kinds of extensions or be forced to utilize a JavaScript SDP
   re-write of SDP approach to "fix" these incompatibilities.

   With an object approach, newer signaling protocols could describe
   multiple media streams with ease and SIP providers could ensure they
   only generate compatible SDP with their networks and agree on their
   best practices and launch new features that incorporate approaches
   like as PlanA, PlanB or NoPlan in a manner they deem fit rather then
   when the browser vendors decide to upgrade the SDP arbitrarily.

2.12.  SIP Forking Issue

   The current SDP based API model does not allow for SIP parallel
   forking even though the RTC engine can allow for demuxing a media
   stream.  The current model does not allow for one offer to be
   transmitted but accepts multiple answers, which is legal in SIP.  A
   complex UPDATE process is described on how to work around the problem
   instead of fixing the original problem, i.e. the state machine being

   A WebRTC JavaScript object model is designed to easily allow forking
   but does not care if an upper shim supports SDP / SIP style forking
   in the negotiation or not, so long as the basic rules of the RTC
   media engine is respected.

3.  Alternatives to Fixing these Issues Now

3.1.  Waiting for WebRTC 2.0

   If we don't get WebRTC 1.0 correct, fixing the API in WebRTC 2.0 may
   become even more difficult.

   At this stage, prototypes are underway but to our knowledge there are
   no major commercial services deployed by more that one major vendor
   using the current WebRTC API.  Yet, the argument to even consider an
   alternative is that 'it's too late'.  Imagine trying to argue fixing
   it after major networks are reliant upon specific browser
   implementation.  Having a good but simple architecture from the start
   could alleviate a lot of pressure to fix a broken 1.0 in a 2.0
   release before APIs become entrenched.

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3.1.1.  Cost now to fix versus fixing later

   The cost of fixing the API issues today may pale in comparison to the
   cost of compatibility problems spread across entire sets of
   industries where constant fixes and work around may be required.

3.1.2.  If starting over, would even SIP people want SDP as a surface

   Even SIP providers and vendors have started to realize that baking
   SDP into the browser is not necessarily in their best interests, but
   they do have an interest in a simple API to use since they aren't
   specialized JavaScript developers but SIP integrators.

   If an alternative approach provides SIP providers a simple JavaScript
   API shim they desire and achieves greater interoperability because of
   predictable, controllable and tailored SDP for their network, would
   they not prefer such a model over the current "baked in the browser"

   If the current WebRTC specification was ever rebooted, the current
   mandated SDP based API would undoubtedly be scrapped in favor of a
   better approach without its inherent design and use case flaws with
   negative long term compatibility consequences.

3.1.3.  Incremental Approach may make Compatibility Worse

   One argument put forward, to keep the current SDP model, proposes the
   current WebRTC SDP-based API must be completed soon and an
   incremental improvement approach can be used to gradually move away
   from these obvious problems.

   The trouble with an incremental approach is that it may increase
   incompatibility further.  Not all browser vendors will match the
   incremental improvements in unison nor will all customers upgrade
   simultaneously.  This puts the onus on JavaScript developers to
   support multiple versions of the WebRTC API and increase the number
   of APIs they must learn and maintain.  The JavaScript developers must
   still perform all the workarounds required for the current API even
   if they support the increments.  This limits their willingness to use
   any additional APIs until all browsers universally support the
   incremental improvements.  This will likely slow innovation and
   adoption of future improvements.

   This will likely create a situation where browser vendors cannot
   easily achieve compliance because they too must support the existing
   API and incremental improvements along the way, or break those
   reliant upon the current methods.

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   Having a good solid simple foundation is key to ensuring basic
   compatibility while allowing for innovation to occur for those
   developers who are willing to give new APIs a trial without needing
   to support multiple sets of equivalent but incompatible APIs

3.2.  Session Description Format Construction API

   An alternative JavaScript model has in the past been floated around,
   other than the model advocated in this draft.  That model creates a
   JavaScript session description format construction API in the
   browser.  Such an API would use JavaScript objects to construct the
   session description format rather than allowing direct control of how
   media should be plumbed together from JavaScript.

   While using SDP as the chosen format for WebRTC highlights the issues
   described in this draft particularly well, using an alternative
   format like JSON instead of SDP does not remove many of the issues
   presented in this draft.  The issues expressed are not solely caused
   by the lack of expressiveness of the SDP format but the nature of
   creating a universal all-encompassing format to describe all
   transport, media, constraints, and negotiations with an attached
   inflexible state machine is the nature of the issue.  This format
   must do everything and encompass all concepts and becomes the
   effective mandate for signaling even if not explicitly required to
   perform signaling.

   A few years ago there was an attempt to create a new "SDP 2.0" format
   with a draft named Session Description and Capability Negotiation
   [I-D.ietf-mmusic-sdpng].  This effort to create the "ultimate" SDP
   format in XML was ultimately abandoned, in no small part because of
   the difficulties in coming up with a single solution that works for
   all scenarios.

   Given the difficulty in creating a universal all-encompassing format
   that works for all scenarios, the idea that creating a JavaScript
   based API that constructs a similar flexible, but well defined
   universal session description format using JavaScript objects is
   highly suspect to fail equally.  The reality is that such an effort
   is complex.

   Even if successful, this format is not necessarily the format that
   will be sent on-the-wire, especially for existing alternative
   signaling protocols.  As such, the format will still need to be
   transformed into alternative formats by JavaScript (or by a gateway).
   If the format must be parsed or interpreted by an intermediate then
   the format becomes an interaction point to the browser no matter how
   clever the JavaScript session description construction API

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   implementation.  Whatever format is selected, each browser or
   alternative protocol format will have to decide how to convert and
   interpret the output and generate new compatible inputs and deal with
   the variations that will undoubtedly arrive from browser to browser
   and from version to version.

   Even if JavaScript APIs are made available to simplify the
   construction or interpretation of a defined format, this format would
   still become a do-everything serialization access point for the
   browser and the defined exchange point for the local and remote
   browser.  Therefore the format itself must be described in meticulous

   The standardization requirements for such an approach would increase
   substantially over the WebRTC JavaScript object model advocated by
   this draft since not only would such a JavaScript format construction
   API have to be standardized (as any JavaScript Model would) but the
   formatting rules and state machine it relies upon needs to become
   standardized in detail as well.

   Every combination of this all-encompassing format would need to be
   outlined, rather than minimal definition of fixed properties needed
   on a scoped objects as used in the WebRTC JavaScript Object Model.
   Any slight variations would likely cause JavaScript developers or
   other browsers to break their implementations.  Obtaining 100%
   stability in such an output equally across all browsers, on all
   platforms with all versions is highly doubtful.

   While a JavaScript format construction API is merely hypothetical at
   the time of writing this draft, any proposal will need to be vetted
   to see if it addresses all the concerns and issues brought up in this

   This hypothetical JavaScript session description construction API
   still puts the emphasis in driving the developer towards building up
   a media signaling exchange format rather than in the logic of how the
   media should be controlled and pipelined.

   The WebRTC JavaScript object model is being proposed as the
   alternative.  In a follow-up to this draft the model will describe
   how the JavaScript developer gains control over the stream's
   pipelining for the browser's media/RTC engine and thus free the
   JavaScript developer to express signaling and state machines using
   whatever mechanism desired.  A simplified shim implemented entirely
   in JavaScript will allow easier interpretation to any format desired
   by the JavaScript developer in a way that can be updated
   independently of a browser's binary release.  Should any changes be
   needed in signaling, a JavaScript shim generating this custom format

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   is strictly under the control of the service provider and not the

4.  Example Difficult Usage Cases with Current Model

4.1.  On / off hold example usage case

   This is a typical scenario widely adopted SIP technique of an SDP
   attribute to place a stream on / off hold.  This is the accepted
   methodology and performing alternative approaches would deviate from
   the expected practices for use with SIP and its manipulation of SDP.
   Although not officially documented as supported, it is effectively
   supported in WebRTC implementations.  This is a typical use case need
   by media application:

   1.  Browser A establishes a connection with Browser B

   2.  Browser A and browser B are streaming media

   3.  JavaScript developer wants Browser A to put "on hold"

   These are the steps that must be performed by a JavaScript developer:

   1.  createOffer to obtain the SDP from Browser A

   2.  Parse the SDP

   3.  Add "a=sendonly" or "a=inactive" to all media

   4.  Regenerate the SDP, feed back to browser

   5.  Send the SDP to Browser B

   6.  Receive the answer from Browser B (which should respond with
       a=recvonly if it still wishes media)

   7.  Parse the received SDP and modify with "a=recvonly" if it did not
       respond correctly (to ensure the local side hold back its media)

   8.  Pass the modified SDP answer back into Browser A

   This also implies that:

   1.  All future SDP events received from Browser B must be mangled to
       ensure the "sendonly/recvonly/inactive" attribute is maintained
       while on hold

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   2.  All future createOffer/createAnswer calls from Browser A must be
       modified to ensure the "sendonly" property is maintained

   3.  We need to handle alternative formats to describe hold, e.g.
       "c=" from Browser B which may not utilize the latest SDP
       specifications depending on the remote device / platform

   Ironically, hold is a very SIP and telephony specific concept.  The
   better approach would be to allow the streams to be pause/unpaused at
   will as that does not require interaction with the SDP, and allow the
   higher layers to signal the desire to pause the session to the remote
   peer in whatever manner desired.

   This is a very basic use case that is extremely complex for a
   JavaScript developer, but it is the only way to perform this
   particular action which is effectively supported by the browsers,
   except only via the "SDP surface API".  Even if this particular use
   case ends up being an exposed JavaScript method to manipulate the SDP
   by the browser, there are countless other scenarios where tweaking a
   field to modify the behavior in the format will only be only
   available via SDP manipulation.

4.2.  One-Sided Constraints Negotiation use Case Scenario

   As WebRTC is a web API and not a SIP API, the API must be capable of
   allowing for alternative signaling methods without enforcing it's own
   signaling aspects (other than basic principles like ensure ICE
   agreement has been achieved for security reasons).

   Consider the following scenario:

   1.  Browser A and Browser B establish a connection

   2.  Browser A and Browser B use one-sided constraints negotiation
       where each party independently decides what "it expects to

   3.  Browser A decides that it wishes to alter the properties of the
       video it expects to receive

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   With this model, browser A must be capable of independently modifying
   its expectations without waiting for an answer from the remote side
   (as that's illegal by the nature of the offer / answer signaling),
   unless the rules are relaxed and special exceptions are made.  For
   the model to work, browser A's receive constraints must be applied to
   the send constraints of the remote peer.  This model does not require
   an SDP offer / answer exchange since the sending peer can monitor the
   expectations of the receiving peer and set its send constraints as

   To achieve this a for one-sided negotiation:

   1.  Browser A's JavaScript must respond to every SDP offer with an
       answer locally generated from JavaScript without a round trip,
       extracting out last known expectations from the remote SDP last
       received as part of the answer

   2.  The JavaScript must update the constraint signaling for the
       remote party

   3.  Browser B's JavaScript sees the constraints have changed from
       Browser A thus it initiates a fake offer from the remote party
       (generating the intentions of the constraint and generating an
       SDP format)

   4.  Browser B's JavaScript must examine the answer if any constraints
       have changed, and if so, it may trigger another reverse situation
       where step 1 is repeated, except with Browser A and B's role

   Is this really doable?  Maybe, with a great deal of difficulty and
   SDP mangling but it is unquestionably a hack and a violation of offer
   / answer (and relaxed rules create exceptions and exceptions require
   additional logic to handle).  The offer / answer rules are violated
   because no round trip was performed at the time when the constraints
   were changed.

   This is also fragile because if Browser B failed to accept the fake
   offer there is no way to enforce the constraint nor can the
   JavaScript rollback the expected constraint.  Likewise if the state
   machine in Browser A expected an offer to be generated before a new
   offer would be accepted, the conflict resolution process would be
   extremely difficult and messy.

   This offer / answer state machine is not even required to fulfill the
   mandate of the RTCWEB Working Group charter but it is currently
   mandated because it supposedly makes producing "SIP interoperability"
   easier (which is highly suspect at best).

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   A JavaScript shim approach on a WebRTC JavaScript object model and
   without offer / answer could achieve the same (or better) "SIP
   interoperability" without breaking other stateless negotiation
   models, such as one-sided negotiation.

4.3.  Meet-me Negotiation Use Case Scenario

   1.  WebRTC client A generates an offer and sends to a server

   2.  WebRTC client B generates an offer and sends to a server

   3.  WebRTC client C generates an offer and sends to a server

   4.  The server returns all the exchanges to each of these clients

   5.  WebRTC client A, B and C interconnect

   Technically, there is no need for independent SDP offer / answer
   negotiation amongst all these peers to achieve a mesh scenario for
   this use case.  Each client has enough information about the other
   clients to establish a peer connection.  The current WebRTC SDP API
   imposes independent round trip negotiations that are not technically
   necessary.  If WebRTC client D was added later, the original
   connection can be forked and re-use the same DTLS fingerprints to
   negotiate new encryptions keys for media or data.  Fingerprint or
   identity signature reuse should not introduce any additional security
   concerns since identities will be verified and keys negotiated for
   each peer-to-peer connection.

   A JavaScript object model approach would allow for this kind of
   scenario without independent round trip negotiations for each WebRTC
   client in the mesh.

4.4.  Browser to Browser Compatibility Extension Compatibility Issue

   Consider the following scenario:

   1.  Browser A has implemented an extension to SDP (which is allowed)

   2.  Browser B has no knowledge of such an extension

   3.  The JavaScript engine running on Browser A has no knowledge of
       the extension

   4.  The JavaScript engine packages up the SDP from Browser A and
       sends it to Browser B

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   Under this scenario, what should browser B do?  To reject the offer
   means communication cannot occur.  To accept the offer has ambiguous
   meaning because the answer might have misunderstood the extension's
   intention and does not allow for the appropriate behavior.

   The exact rules of what is allowed in SDP and what is not and how
   extensions are treated must be defined clearly and non ambiguously.
   Even though current SDP offer / answer API can deal with some
   extensions, like new codecs being introduced, it is ambiguous on how
   to deal with more major extensions such as new SDP profiles,
   transports, or encryption methods.

   Assuming that a lack of response to an extension is non-agreement to
   use the extension is not acceptable.  For example, if the extension
   was security related dictating some security precondition to opening
   a stream, the offer must be rejected as the precondition cannot be
   met.  Ignoring the extension would mean the offer was accepted where
   it cannot be accepted.  Another example would be introduction of new
   SDP profile, like AVPF2.  Offer/answer negotiation simply fails when
   it encounters an unknown profile even if it is backwards compatible,
   like for instance, most of the calls to current SIP devices will fail
   if AVPF is used instead of AVP.  A better approach is to define the
   rules for how extensions can be made, whereas SDP has no such rules.

   Currently, in SIP networks, such extensions are agreed upon in
   advanced and extensively tested before they are introduced.  SBCs
   (Session Border Controllers) are often used to make devices with
   different feature sets work with each other.  By allowing JavaScript
   control over the format generated on the wire, feature roll out is
   under strict control of the provider, and not whenever a browser
   vendor decides to produce an update.

4.5.  Building Interoperability between WebRTC and a SIP Service

   Consider the following scenario:

   1.  Developer takes SDP produced by browser and send to SIP gateway
       (which is supposed to be SIP "compatible")

   2.  Users happily use this service

   3.  Browser Vendor A updates the browser SDP generator and a slight
       variation in SDP changes

   4.  Users are now broken

   5.  SIP gateway must be updated to handle new SDP (and old SDP)

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   6.  Browser Vendor B updates their browser SDP generator (with a
       different SDP variation)

   7.  Users are now broken again

   8.  SIP gateway must be updated to handle another variation of SDP
       (and maintain the old variations)

   9.  Repeat to step 3, but add Browser Vendor C, D and multiple

   This is not an unrealistic scenario by any stretch of the
   imagination.  This currently happens in the SIP world, but at least
   in that world new devices are tested to ensure compatibility before
   roll outs occur on the network so issues can be addressed before the
   user's experience is broken.  Since the SIP provider and gateway
   vendor do not have control over the update cycle of the browsers,
   their users are much more prone to breakage by taking the SDP from
   the browser and sending to their network.

   Whereas this is what happens with a JavaScript Object API model with
   SDP shim written in JavaScript-only:

   1.  Developer uses shim to generate SDP by browser and sends to SIP
       gateway (with SDP that is compatible)

   2.  Users happily use this service

   3.  Browser Vendor A updates the browser with a new RTC feature.

   4.  Repeat to step 2

   The reason why the browser update does not affect the gateway is
   because the SDP is generated entirely in JavaScript and thus updates
   to the browser do not change the SDP generation logic.  The SDP is
   entirely in control of SIP network provider.  Any bugs with SDP
   compatibility can be addressed by the SIP provider without changes in
   the browser's binary.  Bugs, updates and improvements are completely
   within the boundary and control of the SIP network provider.

4.6.  Bit-rate Change Scenario

   Consider the follow scenario:

   1.  User is connected to a conference server

   2.  While user is listening, the user transmits a low bit-rate

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   3.  The users starts to communicate and the bit-rate is adjusted to
       maximum quality

   Using the current WebRTC API, this would require an offer / answer
   round trip to perform the change and thus the quality would be
   updated until the answer was acknowledged, although proposals have
   been made to alter the rules for offer / answer in this case and
   allow for an exception.  This round trip is unnecessary technically
   since the bit-rate can be dynamically adjusted without remote
   acknowledgment.  Yet, the current offer / answer model imposes a
   round trip (unless yet another exception to the SDP rules are

4.7.  Video Codec Option Change Scenario

   Consider the follow scenario:

   1.  JavaScript wishes to change a video codec option

   Using the current WebRTC API, this would require parsing the entire
   SDP, isolating the video codecs for a particular video media line,
   figuring the mapping and then reconstructing the original SDP with
   the newly incorporated changes.  Accessors have been suggested for
   these common use cases but do not exist yet.  If such accessors are
   created then a more involved API cannot be avoided out of necessity.
   One of the main justifications given by SDP proponents for only
   having an API that creates and accepts SDP is due to its supposed
   simplicity, as opposed to providing a more involved API.

4.8.  Video Upgrade Scenario

   1.  Alice and Bob are having an audio conversation

   2.  Alice presses the video button on her application and offers Bob

   3.  Bob does not wish to see Alice's video, so the application
       rejects the media (e.g. using "a=inactive" or "m=video 0")

   4.  Alice's web application successfully parses and interprets Bob's

   5.  As Alice's video window of herself is independent of the SDP
       negotiation, Alice's HTML5 application successfully renders
       Alice's video locally

   The current WebRTC implementation offers no event to indicate the
   rejection, thus Alice is given no feedback of the rejection.  She

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   incorrectly assumes she's in a video conversation.  In order to solve
   this scenario, custom signaling must be added to indicate of Bob's
   rejection of Alice's video.  Yet this is duplication of signaling as
   the video is already rejected in the SDP.  This leaves the JavaScript
   developer with a choice: either parse the SDP, understand the SDP and
   derive meaning, or duplicate the SDP efforts by introducing custom
   signaling for a common scenario when upgrading from audio to video
   and providing appropriate user feedback.

5.  Proposal: WebRTC JavaScript Object Model

5.1.  Overview

   The browser can expose simple object methods, properties and events
   representing the various RTC components at an abstracted level and
   provide a solid API for controlling how the media should be
   pipelined.  The properties needed to be exchanged is separated into
   the appropriate object rather than meshed into an all-encompassing

   A JavaScript-only shim can be layered on top of an object model to
   provide easy SDP offer / answer capability for those who want a
   similar "simple" API to the current WebRTC API for use with SIP.  A
   developer can chose to use this shim or not if they do not need SDP.
   Likewise, the object model could be used to produce alternative
   formats to SDP if the same do-everything format is needed but in an
   alternative on-the-wire session description format.

   The object model described in the solution is presented in a related
   draft.  This solution will allow for the RTCWEB Working Group to
   complete its chartered mandate without starting from scratch.  If
   adopted, all of the drafts proposed to solve issues in expressing SDP
   for WebRTC can be moved to more appropriate working groups.  For
   example, SDP for SIP issues can be moved to the appropriate SIP
   working groups and multi-party SDP to the MMUSIC (e.g. drafts like
   PlanA or PlanB).

5.2.  Benefits

5.2.1.  Greater compatibility

   By having a WebRTC JavaScript object model, the exact inputs,
   outputs, properties and events can be well defined on individual
   objects and each object will be designed to be a specific contract
   between browser vendors and JavaScript developers.

5.2.2.  Easier to extend

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   New objects and methods can be added without breaking existing
   compatibility.  Compliance can be verified with unit tests able to
   test each and every behavior across all browsers' versions on every
   platform.  JavaScript developers can expect their version of the API
   object contract to remain fixed to expected behaviors and not break
   (unless through well planned deprecation).

   Any extensions added to a JavaScript object model does not change the
   behavior expectation from JavaScript developers when using the
   current version of the API regardless of any extensions, unless
   explicitly deprecated.  This is unlike SDP where extensions could be
   silently added into the SDP produced by the browsers at will, even in
   minor browser version changes, where any component that consumes the
   SDP may be unaware what those additional feature behaviors imply or
   require as a result.

5.2.3.  Faster Reaction Time To Issues

   Signaling related bugs produced by the JavaScript shims can easily be
   fixed and updated at any time regardless of the browser's release
   cycle.  If a SIP provider discovers their SIP is not compatible
   within their JavaScript shim, the SIP provider can update the shim
   code to their own needs dynamically without lobbying the browser
   vendor and waiting for the browser to be patched and updated.

5.2.4.  Decreased surface API

   With a JavaScript object model, the features are well defined so the
   surface API is fixed to the agreed contract.  Once agreed, a browser
   vendor only has to ensure their compatibility with well defined
   limited scope unit tests, and need not worry about some free-form
   format that may introduce untold compatibility issues should another
   vendor issue an update.  This is also true of any non-browsers that
   may wish to implement and be compliant to the WebRTC API for
   JavaScript and provide their own JavaScript and WebRTC engines.

5.2.5.  Greater compatibility for SIP

   While SIP is not the main RTCWEB Working Group charter responsibility
   for WebRTC, SIP compatibility is highly desirable.  By exclusively
   generating SDP from a JavaScript shim, the SDP produced will be
   identical across all platforms and all devices with every browser
   version and entirely under the control of the SIP provider.  This
   increases compatibility for SIP providers.  The SDP produced from the
   shim can be custom tailored to a SIP network without affecting any
   other SIP vendor or harming compatibility with other utilizing

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5.2.6.  Alternative formats

   With a JavaScript shim approach on top of an object model, the
   information going over the wire can be transformed from the
   JavaScript object properties to alternative formats, including JSON,
   XML or SIP (or anything custom).  As the JavaScript shim to use is
   under control of the service provider and identical regardless of the
   platform, the output from the JavaScript format generation is
   consistent and controllable, thus ensuring maximum compatibility
   within a network.

   The party receiving this format can be sure the format is to an
   exacting specification of their choosing rather than relying on
   whatever format is produced by whatever browser vendor.

5.3.  Design Goals and Considerations

5.3.1.  Objects Model Kept Simple

   The JavaScript developer should not need to understand the mechanics
   of RTC other than understanding how to plumb the objects together.
   Those who need extended properties or events for finer control can
   obtain them with simple method access to an object, but those
   extended attributes should not be required for simple use cases.

5.3.2.  Simple to Gather Negotiation Information

   The objects model should allow a simple method for collecting
   information that will be needed for various alternative negotiation
   models, highly focused to the object.  One of the targets for
   negotiation must be SDP and SIP.

5.3.3.  Offer / Answer

   The proposed JavaScript object model should not require the offer /
   answer state machine but must not preclude this state machine being
   built in a layer above.  The offer / answer state machine must be
   possible to implement as a JavaScript shim without any additional
   built-in browser services needing to be implemented.

5.3.4.  Extensions

   Extending the object model for the expected common extension use
   cases without breaking the JavaScript API should be possible.  Such
   possible extension use cases should include items like local mixing
   and data synchronization, or extended properties, events or features.

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   As any design, there may be limitations but the design should hold up
   to various realistic scenarios that are likely to happen in the near

5.3.5.  Well Defined Behaviors

   An API must describe specific API behavior sets to the browser
   vendors so they have the appropriate guidelines for implementation,
   including the mapping to on-the-wire to RTC protocols.  The API
   presented in the related draft may be the input to a W3C efforts to
   define specific and exact expected behavior sets for an object based
   JavaScript API for an official WebRTC 1.0 release.

5.3.6.  Data Channel

   The proposed WebRTC JavaScript Object model will provide a definition
   for basic JavaScript usage of the data channel.

5.3.7.  Satisfy the expectations of the RTCWEB charter

   The object model must adhere to the expectations of the RTCWEB
   charter either directly, via extensions that can be defined by the
   working group on top of the object model or possibly via a JavaScript
   shim written to utilize the functionality of the object model but it
   must not preclude the RTCWEB charter from fulfilling its previously
   stated goals.

5.3.8.  SIP/SDP and current WebRTC API shim compatibility statement

   The goal of the object model is to allow for a JavaScript shim that
   provides a simple mechanism for parsing and generating SDP for basic
   compatibility with SIP networks (capable of supporting the WebRTC
   wire protocols).

   The goal of this object based model is not to provide working
   JavaScript shim on top that is a 1-for-1 matching of the current
   WebRTC API as a shim, including all behaviors, features, bugs and
   expectations since the definition of the current approach is not
   defined enough to be able to produce that level of compatibility.
   This would be an impossible goal as a result, and would add little

   Extensions are beyond the scope of the JavaScript shim, but it is
   possible for others to fork and modify the shim to their own needs
   specific to their own SIP/SDP network infrastructure.

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   Compatibility with the SDP used in all SIP networks is not a stated
   goal for any JavaScript shim since not even SIP providers can agree
   on a common agreed definitive standard set of RFCs and drafts.

5.3.9.  Greater Separation of RTCWEB Working Group and Other Working

   A JavasScript object model would remove much of the need for cross
   IETF working group coordination, which has become common place with
   the current movement because of utilizing SDP and its close ties to
   SIP.  By limiting the RTCWEB technologies used to only those required
   for Real-Time Communication from the browser (e.g. RTP, ICE/STUN/
   TURN, DTLS), the RTCWEB Working Group is freed from tight couplings
   with other IETF working groups, each having their own charters,
   schedules, agendas and interests and thus ensures more rapid progress
   between RTCWEB Working Group the W3C and developers who are to use
   this technology.

6.  Security Considerations

   While RTCWEB has it's own security considerations for protocols, a
   JavaScript object model has no additional requirements other than
   those already established for use within RTCWEB, e.g. ICE
   connectivity permission check or DTLS fingerprint checks.

   JavaScript as a browser language itself has security consideration
   but nothing inherent to using a JavaScript object model versus a
   JavaScript SDP API model, as any proposed implementations must have a
   JavaScript API.  The specifics of any API must list their own
   specific security considerations to their defined model and API,
   should any exist.

   Any specific issues for the proposed JavaScript object model will be
   outlined in the separated draft WebRTC JavaScript object model draft
   as needed and warranted.

7.  References

7.1.  Normative References

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

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

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7.2.  Informative References

              Kutscher, D., Ott, J., and C. Bormann, "Session
              Description and Capability Negotiation", draft-ietf-
              mmusic-sdpng-08 (work in progress), February 2005.

              Ivov, E., Marocco, E., and P. Thatcher, "No Plan:
              Economical Use of the Offer/Answer Model in WebRTC
              Sessions with Multiple Media Sources", draft-ivov-rtcweb-
              noplan-01 (work in progress), June 2013.

              Roach, A. and M. Thomson, "Using SDP with Large Numbers of
              Media Flows", draft-roach-rtcweb-plan-a-00 (work in
              progress), May 2013.

              Uberti, J., "Plan B: a proposal for signaling multiple
              media sources in WebRTC.", draft-uberti-rtcweb-plan-00
              (work in progress), May 2013.

              Burnett, D., "Media Capture and Streams", May 2013, <http:

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

   [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.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

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

   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              October 2008.

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   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, January 2012.

              Bergkvist, A., "WebRTC 1.0 Real-time Communication Between
              Browsers", August 2012,

Authors' Addresses

   Robin Raymond
   436, 3553 31 St. NW
   Calgary, Alberta  T2L 2K7


   Erik Lagerway
   436, 3553 31 St. NW
   Calgary, Alberta  T2L 2K7


   Inaki Baz Castillo
   Basque Country


   Roman Shpount
   4905 Del Ray Ave Suite 300
   Bethesda, MD  20814


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