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In: MissingRef
Network Working Group                                          J. Uberti
Internet-Draft                                                    Google
Intended status: Standards Track                             C. Jennings
Expires: March 22, 2014                                            Cisco
                                                      September 18, 2013


               Javascript Session Establishment Protocol
                       draft-ietf-rtcweb-jsep-04

Abstract

   This document describes the mechanisms for allowing a Javascript
   application to control the signaling plane of a multimedia session
   via the interface specified in the W3C RTCPeerConnection API, and
   discusses how this relates to existing signaling protocols.

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 http://datatracker.ietf.org/drafts/current/.

   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 March 22, 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
   (http://trustee.ietf.org/license-info) in effect on the date of
   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.



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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
     1.1.  General Design of JSEP  . . . . . . . . . . . . . . . . .   3
     1.2.  Other Approaches Considered . . . . . . . . . . . . . . .   5
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   6
   3.  Semantics and Syntax  . . . . . . . . . . . . . . . . . . . .   6
     3.1.  Signaling Model . . . . . . . . . . . . . . . . . . . . .   6
     3.2.  Session Descriptions and State Machine  . . . . . . . . .   7
     3.3.  Session Description Format  . . . . . . . . . . . . . . .   9
     3.4.  ICE . . . . . . . . . . . . . . . . . . . . . . . . . . .  10
       3.4.1.  ICE Candidate Trickling . . . . . . . . . . . . . . .  10
         3.4.1.1.  ICE Candidate Format  . . . . . . . . . . . . . .  10
     3.5.  Interactions With Forking . . . . . . . . . . . . . . . .  11
       3.5.1.  Sequential Forking  . . . . . . . . . . . . . . . . .  11
       3.5.2.  Parallel Forking  . . . . . . . . . . . . . . . . . .  12
     3.6.  Session Rehydration . . . . . . . . . . . . . . . . . . .  13
   4.  Interface . . . . . . . . . . . . . . . . . . . . . . . . . .  13
     4.1.  Methods . . . . . . . . . . . . . . . . . . . . . . . . .  14
       4.1.1.  createOffer . . . . . . . . . . . . . . . . . . . . .  14
       4.1.2.  createAnswer  . . . . . . . . . . . . . . . . . . . .  15
       4.1.3.  SessionDescriptionType  . . . . . . . . . . . . . . .  15
         4.1.3.1.  Use of Provisional Answers  . . . . . . . . . . .  16
         4.1.3.2.  Rollback  . . . . . . . . . . . . . . . . . . . .  17
       4.1.4.  setLocalDescription . . . . . . . . . . . . . . . . .  17
       4.1.5.  setRemoteDescription  . . . . . . . . . . . . . . . .  18
       4.1.6.  localDescription  . . . . . . . . . . . . . . . . . .  18
       4.1.7.  remoteDescription . . . . . . . . . . . . . . . . . .  18
       4.1.8.  updateIce . . . . . . . . . . . . . . . . . . . . . .  19
       4.1.9.  addIceCandidate . . . . . . . . . . . . . . . . . . .  19
   5.  SDP Interaction Procedures  . . . . . . . . . . . . . . . . .  19
     5.1.  SDP Requirements Overview . . . . . . . . . . . . . . . .  19
     5.2.  Constructing an Offer . . . . . . . . . . . . . . . . . .  21
       5.2.1.  Initial Offers  . . . . . . . . . . . . . . . . . . .  21
       5.2.2.  Subsequent Offers . . . . . . . . . . . . . . . . . .  25
       5.2.3.  Constraints Handling  . . . . . . . . . . . . . . . .  26
         5.2.3.1.  OfferToReceiveAudio . . . . . . . . . . . . . . .  26
         5.2.3.2.  OfferToReceiveVideo . . . . . . . . . . . . . . .  27
         5.2.3.3.  VoiceActivityDetection  . . . . . . . . . . . . .  27
         5.2.3.4.  IceRestart  . . . . . . . . . . . . . . . . . . .  27
     5.3.  Generating an Answer  . . . . . . . . . . . . . . . . . .  27
       5.3.1.  Initial Answers . . . . . . . . . . . . . . . . . . .  27
       5.3.2.  Subsequent Answers  . . . . . . . . . . . . . . . . .  31
       5.3.3.  Constraints Handling  . . . . . . . . . . . . . . . .  31
     5.4.  Parsing an Offer  . . . . . . . . . . . . . . . . . . . .  31
     5.5.  Parsing an Answer . . . . . . . . . . . . . . . . . . . .  31
     5.6.  Applying a Local Description  . . . . . . . . . . . . . .  31
     5.7.  Applying a Remote Description . . . . . . . . . . . . . .  31



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   6.  Configurable SDP Parameters . . . . . . . . . . . . . . . . .  31
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  33
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  33
   9.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  33
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  33
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  33
     10.2.  Informative References . . . . . . . . . . . . . . . . .  35
   Appendix A.  JSEP Implementation Examples . . . . . . . . . . . .  36
     A.1.  Example API Flows . . . . . . . . . . . . . . . . . . . .  36
       A.1.1.  Call using ROAP . . . . . . . . . . . . . . . . . . .  36
       A.1.2.  Call using XMPP . . . . . . . . . . . . . . . . . . .  37
       A.1.3.  Adding video to a call, using XMPP  . . . . . . . . .  38
       A.1.4.  Simultaneous add of video streams, using XMPP . . . .  39
       A.1.5.  Call using SIP  . . . . . . . . . . . . . . . . . . .  40
       A.1.6.  Handling early media (e.g. 1-800-GO FEDEX), using SIP  40
     A.2.  Example Session Descriptions  . . . . . . . . . . . . . .  41
       A.2.1.  createOffer . . . . . . . . . . . . . . . . . . . . .  41
       A.2.2.  createAnswer  . . . . . . . . . . . . . . . . . . . .  43
   Appendix B.  Change log . . . . . . . . . . . . . . . . . . . . .  44
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  45

1.  Introduction

   This document describes how the W3C WEBRTC RTCPeerConnection
   interface[W3C.WD-webrtc-20111027] is used to control the setup,
   management and teardown of a multimedia session.

1.1.  General Design of JSEP

   The thinking behind WebRTC call setup has been to fully specify and
   control the media plane, but to leave the signaling plane up to the
   application as much as possible.  The rationale is that different
   applications may prefer to use different protocols, such as the
   existing SIP or Jingle call signaling protocols, or something custom
   to the particular application, perhaps for a novel use case.  In this
   approach, the key information that needs to be exchanged is the
   multimedia session description, which specifies the necessary
   transport and media configuration information necessary to establish
   the media plane.

   The browser environment also has its own challenges that pose
   problems for an embedded signaling state machine.  One of these is
   that the user may reload the web page at any time.  If the browser is
   fully in charge of the signaling state, this will result in the loss
   of the call when this state is wiped by the reload.  However, if the
   state can be stored at the server, and pushed back down to the new
   page, the call can be resumed with minimal interruption.




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   With these considerations in mind, this document describes the
   Javascript Session Establishment Protocol (JSEP) that allows for full
   control of the signaling state machine from Javascript.  This
   mechanism effectively removes the browser almost completely from the
   core signaling flow; the only interface needed is a way for the
   application to pass in the local and remote session descriptions
   negotiated by whatever signaling mechanism is used, and a way to
   interact with the ICE state machine.

   In this document, the use of JSEP is described as if it always occurs
   between two browsers.  Note though in many cases it will actually be
   between a browser and some kind of server, such as a gateway or MCU.
   This distinction is invisible to the browser; it just follows the
   instructions it is given via the API.

   JSEP's handling of session descriptions is simple and
   straightforward.  Whenever an offer/answer exchange is needed, the
   initiating side creates an offer by calling a createOffer() API.  The
   application optionally modifies that offer, and then uses it to set
   up its local config via the setLocalDescription() API.  The offer is
   then sent off to the remote side over its preferred signaling
   mechanism (e.g., WebSockets); upon receipt of that offer, the remote
   party installs it using the setRemoteDescription() API.

   When the call is accepted, the callee uses the createAnswer() API to
   generate an appropriate answer, applies it using
   setLocalDescription(), and sends the answer back to the initiator
   over the signaling channel.  When the offerer gets that answer, it
   installs it using setRemoteDescription(), and initial setup is
   complete.  This process can be repeated for additional offer/answer
   exchanges.

   Regarding ICE [RFC5245], JSEP decouples the ICE state machine from
   the overall signaling state machine, as the ICE state machine must
   remain in the browser, because only the browser has the necessary
   knowledge of candidates and other transport info.  Performing this
   separation also provides additional flexibility; in protocols that
   decouple session descriptions from transport, such as Jingle, the
   transport information can be sent separately; in protocols that
   don't, such as SIP, the information can be used in the aggregated
   form.  Sending transport information separately can allow for faster
   ICE and DTLS startup, since the necessary roundtrips can occur while
   waiting for the remote side to accept the session.

   Through its abstraction of signaling, the JSEP approach does require
   the application to be aware of the signaling process.  While the
   application does not need to understand the contents of session
   descriptions to set up a call, the application must call the right



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   APIs at the right times, convert the session descriptions and ICE
   information into the defined messages of its chosen signaling
   protocol, and perform the reverse conversion on the messages it
   receives from the other side.

   One way to mitigate this is to provide a Javascript library that
   hides this complexity from the developer; said library would
   implement a given signaling protocol along with its state machine and
   serialization code, presenting a higher level call-oriented interface
   to the application developer.  For example, this library could easily
   adapt the JSEP API into the API that was proposed for the ROAP
   signaling protocol [I-D.jennings-rtcweb-signaling], which would
   perform a ROAP call setup under the covers, interacting with the
   application only when it needs a signaling message to be sent.  In
   the same fashion, one could also implement other popular signaling
   protocols, including SIP or Jingle.  This allow JSEP to provide
   greater control for the experienced developer without forcing any
   additional complexity on the novice developer.

1.2.  Other Approaches Considered

   One approach that was considered instead of JSEP was to include a
   lightweight signaling protocol.  Instead of providing session
   descriptions to the API, the API would produce and consume messages
   from this protocol.  While providing a more high-level API, this put
   more control of signaling within the browser, forcing the browser to
   have to understand and handle concepts like signaling glare.  In
   addition, it prevented the application from driving the state machine
   to a desired state, as is needed in the page reload case.

   A second approach that was considered but not chosen was to decouple
   the management of the media control objects from session
   descriptions, instead offering APIs that would control each component
   directly.  This was rejected based on a feeling that requiring
   exposure of this level of complexity to the application programmer
   would not be beneficial; it would result in an API where even a
   simple example would require a significant amount of code to
   orchestrate all the needed interactions, as well as creating a large
   API surface that needed to be agreed upon and documented.  In
   addition, these API points could be called in any order, resulting in
   a more complex set of interactions with the media subsystem than the
   JSEP approach, which specifies how session descriptions are to be
   evaluated and applied.

   One variation on JSEP that was considered was to keep the basic
   session description-oriented API, but to move the mechanism for
   generating offers and answers out of the browser.  Instead of
   providing createOffer/createAnswer methods within the browser, this



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   approach would instead expose a getCapabilities API which would
   provide the application with the information it needed in order to
   generate its own session descriptions.  This increases the amount of
   work that the application needs to do; it needs to know how to
   generate session descriptions from capabilities, and especially how
   to generate the correct answer from an arbitrary offer and the
   supported capabilities.  While this could certainly be addressed by
   using a library like the one mentioned above, it basically forces the
   use of said library even for a simple example.  Providing createOffer
   /createAnswer avoids this problem, but still allows applications to
   generate their own offers/answers (to a large extent) if they choose,
   using the description generated by createOffer as an indication of
   the browser's capabilities.

2.  Terminology

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

3.  Semantics and Syntax

3.1.  Signaling Model

   JSEP does not specify a particular signaling model or state machine,
   other than the generic need to exchange SDP media descriptions in the
   fashion described by [RFC3264] (offer/answer) in order for both sides
   of the session to know how to conduct the session.  JSEP provides
   mechanisms to create offers and answers, as well as to apply them to
   a session.  However, the browser is totally decoupled from the actual
   mechanism by which these offers and answers are communicated to the
   remote side, including addressing, retransmission, forking, and glare
   handling.  These issues are left entirely up to the application; the
   application has complete control over which offers and answers get
   handed to the browser, and when.

    +-----------+                               +-----------+
    |  Web App  |<--- App-Specific Signaling -->|  Web App  |
    +-----------+                               +-----------+
          ^                                            ^
          |  SDP                                       |  SDP
          V                                            V
    +-----------+                                +-----------+
    |  Browser  |<----------- Media ------------>|  Browser  |
    +-----------+                                +-----------+

                      Figure 1: JSEP Signaling Model




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3.2.  Session Descriptions and State Machine

   In order to establish the media plane, the user agent needs specific
   parameters to indicate what to transmit to the remote side, as well
   as how to handle the media that is received.  These parameters are
   determined by the exchange of session descriptions in offers and
   answers, and there are certain details to this process that must be
   handled in the JSEP APIs.

   Whether a session description applies to the local side or the remote
   side affects the meaning of that description.  For example, the list
   of codecs sent to a remote party indicates what the local side is
   willing to receive, which, when intersected with the set of codecs
   the remote side supports, specifies what the remote side should send.
   However, not all parameters follow this rule; for example, the SRTP
   parameters [RFC4568] sent to a remote party indicate what the local
   side will use to encrypt, and thereby what the remote party should
   expect to receive; the remote party will have to accept these
   parameters, with no option to choose a different value.

   In addition, various RFCs put different conditions on the format of
   offers versus answers.  For example, a offer may propose multiple
   SRTP configurations, but an answer may only contain a single SRTP
   configuration.

   Lastly, while the exact media parameters are only known only after a
   offer and an answer have been exchanged, it is possible for the
   offerer to receive media after they have sent an offer and before
   they have received an answer.  To properly process incoming media in
   this case, the offerer's media handler must be aware of the details
   of the offer before the answer arrives.

   Therefore, in order to handle session descriptions properly, the user
   agent needs:

   1.  To know if a session description pertains to the local or remote
       side.

   2.  To know if a session description is an offer or an answer.

   3.  To allow the offer to be specified independently of the answer.

   JSEP addresses this by adding both a setLocalDescription and a
   setRemoteDescription method and having session description objects
   contain a type field indicating the type of session description being
   supplied.  This satisfies the requirements listed above for both the
   offerer, who first calls setLocalDescription(sdp [offer]) and then
   later setRemoteDescription(sdp [answer]), as well as for the



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   answerer, who first calls setRemoteDescription(sdp [offer]) and then
   later setLocalDescription(sdp [answer]).

   JSEP also allows for an answer to be treated as provisional by the
   application.  Provisional answers provide a way for an answerer to
   communicate initial session parameters back to the offerer, in order
   to allow the session to begin, while allowing a final answer to be
   specified later.  This concept of a final answer is important to the
   offer/answer model; when such an answer is received, any extra
   resources allocated by the caller can be released, now that the exact
   session configuration is known.  These "resources" can include things
   like extra ICE components, TURN candidates, or video decoders.
   Provisional answers, on the other hand, do no such deallocation
   results; as a result, multiple dissimilar provisional answers can be
   received and applied during call setup.

   In [RFC3264], the constraint at the signaling level is that only one
   offer can be outstanding for a given session, but from the media
   stack level, a new offer can be generated at any point.  For example,
   when using SIP for signaling, if one offer is sent, then cancelled
   using a SIP CANCEL, another offer can be generated even though no
   answer was received for the first offer.  To support this, the JSEP
   media layer can provide an offer whenever the Javascript application
   needs one for the signaling.  The answerer can send back zero or more
   provisional answers, and finally end the offer-answer exchange by
   sending a final answer.  The state machine for this is as follows:

                       setRemote(OFFER)               setLocal(PRANSWER)
                           /-----\                               /-----\
                           |     |                               |     |
                           v     |                               v     |
            +---------------+    |                +---------------+    |
            |               |----/                |               |----/
            |               | setLocal(PRANSWER)  |               |
            |  Remote-Offer |------------------- >| Local-Pranswer|
            |               |                     |               |
            |               |                     |               |
            +---------------+                     +---------------+
                 ^   |                                   |
                 |   | setLocal(ANSWER)                  |
   setRemote(OFFER)  |                                   |
                 |   V                  setLocal(ANSWER) |
            +---------------+                            |
            |               |                            |
            |               |                            |
            |    Stable     |<---------------------------+
            |               |                            |
            |               |                            |



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            +---------------+          setRemote(ANSWER) |
                 ^   |                                   |
                 |   | setLocal(OFFER)                   |
   setRemote(ANSWER) |                                   |
                 |   V                                   |
            +---------------+                     +---------------+
            |               |                     |               |
            |               | setRemote(PRANSWER) |               |
            |  Local-Offer  |------------------- >|Remote-Pranswer|
            |               |                     |               |
            |               |----\                |               |----\
            +---------------+    |                +---------------+    |
                           ^     |                               ^     |
                           |     |                               |     |
                           \-----/                               \-----/
                       setLocal(OFFER)               setRemote(PRANSWER)

                       Figure 2: JSEP State Machine

   Aside from these state transitions, there is no other difference
   between the handling of provisional ("pranswer") and final ("answer")
   answers.

3.3.  Session Description Format

   In the WebRTC specification, session descriptions are formatted as
   SDP messages.  While this format is not optimal for manipulation from
   Javascript, it is widely accepted, and frequently updated with new
   features.  Any alternate encoding of session descriptions would have
   to keep pace with the changes to SDP, at least until the time that
   this new encoding eclipsed SDP in popularity.  As a result, JSEP
   currently uses SDP as the internal representation for its session
   descriptions.

   However, to simplify Javascript processing, and provide for future
   flexibility, the SDP syntax is encapsulated within a
   SessionDescription object, which can be constructed from SDP, and be
   serialized out to SDP.  If future specifications agree on a JSON
   format for session descriptions, we could easily enable this object
   to generate and consume that JSON.

   Other methods may be added to SessionDescription in the future to
   simplify handling of SessionDescriptions from Javascript.  In the
   meantime, Javascript libraries can be used to perform these
   manipulations.

   Note that most applications should be able to treat the
   SessionDescriptions produced and consumed by these various API calls



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   as opaque blobs; that is, the application will not need to read or
   change them.  The W3C API will provide appropriate APIs to allow the
   application to control various session parameters, which will provide
   the necessary information to the browser about what sort of
   SessionDescription to produce.

3.4.  ICE

   When a new ICE candidate is available, the ICE Agent will notify the
   application via a callback; these candidates will automatically be
   added to the local session description.  When all candidates have
   been gathered, the callback will also be invoked to signal that the
   gathering process is complete.

3.4.1.  ICE Candidate Trickling

   Candidate trickling is a technique through which a caller may
   incrementally provide candidates to the callee after the initial
   offer has been dispatched; the semantics of "Trickle ICE" are defined
   in [I-D.ivov-mmusic-trickle-ice].  This process allows the callee to
   begin acting upon the call and setting up the ICE (and perhaps DTLS)
   connections immediately, without having to wait for the caller to
   gather all possible candidates.  This results in faster call startup
   in cases where gathering is not performed prior to initiating the
   call.

   JSEP supports optional candidate trickling by providing APIs that
   provide control and feedback on the ICE candidate gathering process.
   Applications that support candidate trickling can send the initial
   offer immediately and send individual candidates when they get the
   notified of a new candidate; applications that do not support this
   feature can simply wait for the indication that gathering is
   complete, and then create and send their offer, with all the
   candidates, at this time.

   Upon receipt of trickled candidates, the receiving application will
   supply them to its ICE Agent.  This triggers the ICE Agent to start
   using the new remote candidates for connectivity checks.

3.4.1.1.  ICE Candidate Format

   As with session descriptions, the syntax of the IceCandidate object
   provides some abstraction, but can be easily converted to and from
   the SDP candidate lines.

   The candidate lines are the only SDP information that is contained
   within IceCandidate, as they represent the only information needed
   that is not present in the initial offer (i.e.  for trickle



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   candidates).  This information is carried with the same syntax as the
   "candidate-attribute" field defined for ICE.  For example:

   candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host


   The IceCandidate object also contains fields to indicate which m=
   line it should be associated with.  The m line can be identified in
   one of two ways; either by a m-line index, or a MID.  The m-line
   index is a zero-based index, referring to the Nth m-line in the SDP.
   The MID uses the "media stream identification", as defined in
   [RFC5888] , to identify the m-line.  WebRTC implementations creating
   an ICE Candidate object MUST populate both of these fields.
   Implementations receiving an ICE Candidate object SHOULD use the MID
   if they implement that functionality, or the m-line index, if not.

3.5.  Interactions With Forking

   Some call signaling systems allow various types of forking where an
   SDP Offer may be provided to more than one device.  For example, SIP
   [RFC3261] defines both a "Parallel Search" and "Sequential Search".
   Although these are primarily signaling level issues that are outside
   the scope of JSEP, they do have some impact on the configuration of
   the media plane which is relevant.  When forking happens at the
   signaling layer, the Javascript application responsible for the
   signaling needs to make the decisions about what media should be sent
   or received at any point of time, as well as which remote endpoint it
   should communicate with; JSEP is used to make sure the media engine
   can make the RTP and media perform as required by the application.
   The basic operations that the applications can have the media engine
   do are:

      Start exchanging media to a given remote peer, but keep all the
      resources reserved in the offer.

      Start exchanging media with a given remote peer, and free any
      resources in the offer that are not being used.

3.5.1.  Sequential Forking

   Sequential forking involves a call being dispatched to multiple
   remote callees, where each callee can accept the call, but only one
   active session ever exists at a time; no mixing of received media is
   performed.

   JSEP handles sequential forking well, allowing the application to
   easily control the policy for selecting the desired remote endpoint.
   When an answer arrives from one of the callees, the application can



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   choose to apply it either as a provisional answer, leaving open the
   possibility of using a different answer in the future, or apply it as
   a final answer, ending the setup flow.

   In a "first-one-wins" situation, the first answer will be applied as
   a final answer, and the application will reject any subsequent
   answers.  In SIP parlance, this would be ACK + BYE.

   In a "last-one-wins" situation, all answers would be applied as
   provisional answers, and any previous call leg will be terminated.
   At some point, the application will end the setup process, perhaps
   with a timer; at this point, the application could reapply the
   existing remote description as a final answer.

3.5.2.  Parallel Forking

   Parallel forking involves a call being dispatched to multiple remote
   callees, where each callee can accept the call, and multiple
   simultaneous active signaling sessions can be established as a
   result.  If multiple callees send media at the same time, the
   possibilities for handling this are described in Section 3.1 of
   [RFC3960].  Most SIP devices today only support exchanging media with
   a single device at a time, and do not try to mix multiple early media
   audio sources, as that could result in a confusing situation.  For
   example, consider having a European ringback tone mixed together with
   the North American ringback tone - the resulting sound would not be
   like either tone, and would confuse the user.  If the signaling
   application wishes to only exchange media with one of the remote
   endpoints at a time, then from a media engine point of view, this is
   exactly like the sequential forking case.

   In the parallel forking case where the Javascript application wishes
   to simultaneously exchange media with multiple peers, the flow is
   slightly more complex, but the Javascript application can follow the
   strategy that [RFC3960] describes using UPDATE.  (It is worth noting
   that use cases where this is the desired behavior are very unusual.)
   The UPDATE approach allows the signaling to set up a separate media
   flow for each peer that it wishes to exchange media with.  In JSEP,
   this offer used in the UPDATE would be formed by simply creating a
   new PeerConnection and making sure that the same local media streams
   have been added into this new PeerConnection.  Then the new
   PeerConnection object would produce a SDP offer that could be used by
   the signaling to perform the UPDATE strategy discussed in [RFC3960].

   As a result of sharing the media streams, the application will end up
   with N parallel PeerConnection sessions, each with a local and remote
   description and their own local and remote addresses.  The media flow
   from these sessions can be managed by specifying SDP direction



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   attributes in the descriptions, or the application can choose to play
   out the media from all sessions mixed together.  Of course, if the
   application wants to only keep a single session, it can simply
   terminate the sessions that it no longer needs.

3.6.  Session Rehydration

   In the event that the local application state is reinitialized,
   either due to a user reload of the page, or a decision within the
   application to reload itself (perhaps to update to a new version), it
   is possible to keep an existing session alive, via a process called
   "rehydration".  The explicit goal of rehydration is to carry out this
   session resumption with no interaction with the remote side other
   than normal call signaling messages.

   With rehydration, the current signaling state is persisted somewhere
   outside of the page, perhaps on the application server, or in browser
   local storage.  The page is then reloaded, the saved signaling state
   is retrieved, and a new PeerConnection object is created for the
   session.  The previously obtained MediaStreams are re-acquired, and
   are given the same IDs as the original session; this ensures the IDs
   in use by the remote side continue to work.  Next, a new offer is
   generated by the new PeerConnection; this offer will have new ICE and
   possibly new DTLS-SRTP certificate fingerprints (since the old ICE
   and SRTP state has been lost).  Finally, this offer is used to re-
   initiate the session with the existing remote endpoint, who simply
   sees the new offer as an in-call renegotiation, and replies with an
   answer that can be supplied to setRemoteDescription.  ICE processing
   proceeds as usual, and as soon as connectivity is established, the
   session will be back up and running again.

   [OPEN ISSUE: EKR proposed an alternative rehydration approach where
   the actual internal PeerConnection object in the browser was kept
   alive for some time after the web page was killed and provided some
   way for a new page to acquire the old PeerConnection object.]

4.  Interface

   This section details the basic operations that must be present to
   implement JSEP functionality.  The actual API exposed in the W3C API
   may have somewhat different syntax, but should map easily to these
   concepts.









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4.1.  Methods

4.1.1.  createOffer

   The createOffer method generates a blob of SDP that contains a
   [RFC3264] offer with the supported configurations for the session,
   including descriptions of the local MediaStreams attached to this
   PeerConnection, the codec/RTP/RTCP options supported by this
   implementation, and any candidates that have been gathered by the ICE
   Agent.  A constraints parameters may be supplied to provide
   additional control over the generated offer.  This constraints
   parameter should allow for the following manipulations to be
   performed:

   o  To indicate support for a media type even if no MediaStreamTracks
      of that type have been added to the session (e.g., an audio call
      that wants to receive video.)

   o  To trigger an ICE restart, for the purpose of reestablishing
      connectivity.

   o  For re-offer cases, to request an offer that contains the full set
      of supported capabilities, as opposed to just the currently
      negotiated parameters.

   In the initial offer, the generated SDP will contain all desired
   functionality for the session (certain parts that are supported but
   not desired by default may be omitted); for each SDP line, the
   generation of the SDP will follow the process defined for generating
   an initial offer from the document that specifies the given SDP line.
   The exact handling of initial offer generation is detailed in
   Section 5.2.1. below.

   In the event createOffer is called after the session is established,
   createOffer will generate an offer to modify the current session
   based on any changes that have been made to the session, e.g. adding
   or removing MediaStreams, or requesting an ICE restart.  For each
   existing stream, the generation of each SDP line must follow the
   process defined for generating an updated offer from the document
   that specifies the given SDP line.  For each new stream, the
   generation of the SDP must follow the process of generating an
   initial offer, as mentioned above.  If no changes have been made, or
   for SDP lines that are unaffected by the requested changes, the offer
   will only contain the parameters negotiated by the last offer-answer
   exchange.  The exact handling of subsequent offer generation is
   detailed in Section 5.2.2. below.





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   Session descriptions generated by createOffer must be immediately
   usable by setLocalDescription; if a system has limited resources
   (e.g. a finite number of decoders), createOffer should return an
   offer that reflects the current state of the system, so that
   setLocalDescription will succeed when it attempts to acquire those
   resources.  Because this method may need to inspect the system state
   to determine the currently available resources, it may be implemented
   as an async operation.

   Calling this method may do things such as generate new ICE
   credentials, but does not result in candidate gathering, or cause
   media to start or stop flowing.

4.1.2.  createAnswer

   The createAnswer method generates a blob of SDP that contains a
   [RFC3264] SDP answer with the supported configuration for the session
   that is compatible with the parameters supplied in the offer.  Like
   createOffer, the returned blob contains descriptions of the local
   MediaStreams attached to this PeerConnection, the codec/RTP/RTCP
   options negotiated for this session, and any candidates that have
   been gathered by the ICE Agent.  A constraints parameter may be
   supplied to provide additional control over the generated answer.

   As an answer, the generated SDP will contain a specific configuration
   that specifies how the media plane should be established; for each
   SDP line, the generation of the SDP must follow the process defined
   for generating an answer from the document that specifies the given
   SDP line.  The exact handling of answer generation is detailed in
   Section 5.3. below.

   Session descriptions generated by createAnswer must be immediately
   usable by setLocalDescription; like createOffer, the returned
   description should reflect the current state of the system.  Because
   this method may need to inspect the system state to determine the
   currently available resources, it may need to be implemented as an
   async operation.

   Calling this method may do things such as generate new ICE
   credentials, but does not trigger candidate gathering or change media
   state.

4.1.3.  SessionDescriptionType

   Session description objects (RTCSessionDescription) may be of type
   "offer", "pranswer", and "answer".  These types provide information
   as to how the description parameter should be parsed, and how the
   media state should be changed.



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   "offer" indicates that a description should be parsed as an offer;
   said description may include many possible media configurations.  A
   description used as an "offer" may be applied anytime the
   PeerConnection is in a stable state, or as an update to a previously
   supplied but unanswered "offer".

   "pranswer" indicates that a description should be parsed as an
   answer, but not a final answer, and so should not result in the
   freeing of allocated resources.  It may result in the start of media
   transmission, if the answer does not specify an inactive media
   direction.  A description used as a "pranswer" may be applied as a
   response to an "offer", or an update to a previously sent "answer".

   "answer" indicates that a description should be parsed as an answer,
   the offer-answer exchange should be considered complete, and any
   resources (decoders, candidates) that are no longer needed can be
   released.  A description used as an "answer" may be applied as a
   response to a "offer", or an update to a previously sent "pranswer".

   The only difference between a provisional and final answer is that
   the final answer results in the freeing of any unused resources that
   were allocated as a result of the offer.  As such, the application
   can use some discretion on whether an answer should be applied as
   provisional or final, and can change the type of the session
   description as needed.  For example, in a serial forking scenario, an
   application may receive multiple "final" answers, one from each
   remote endpoint.  The application could choose to accept the initial
   answers as provisional answers, and only apply an answer as final
   when it receives one that meets its criteria (e.g. a live user
   instead of voicemail).

4.1.3.1.  Use of Provisional Answers

   Most web applications will not need to create answers using the
   "pranswer" type.  The preferred handling for a web application would
   be to create and send an "inactive" answer more or less immediately
   after receiving the offer, instead of waiting for a human user to
   physically answer the call.  Later, when the human input is received,
   the application can create a new "sendrecv" offer to update the
   previous offer/answer pair and start the media flow.  This approach
   is preferred because it minimizes the amount of time that the offer-
   answer exchange is left open, in addition to avoiding media clipping
   by ensuring the transport is ready to go by the time the call is
   physically answered.  However, some applications may not be able to
   do this, particularly ones that are attempting to gateway to other
   signaling protocols.  In these cases, "pranswer" can still allow the
   application to warm up the transport.




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   Consider a typical web application that will set up a data channel,
   an audio channel, and a video channel.  When an endpoint receives an
   offer with these channels, it could send an answer accepting the data
   channel for two-way data, and accepting the audio and video tracks as
   inactive or receive-only.  It could then ask the user to accept the
   call, acquire the local media streams, and send a new offer to the
   remote side moving the audio and video to be two-way media.  By the
   time the human has accepted the call and sent the new offer, it is
   likely that the ICE and DTLS handshaking for all the channels will
   already be set up.

4.1.3.2.  Rollback

   In certain situations it may be desirable to "undo" a change made to
   setLocalDescription or setRemoteDescription.  Consider a case where a
   call is ongoing, and one side wants to change some of the session
   parameters; that side generates an updated offer and then calls
   setLocalDescription.  However, the remote side, either before or
   after setRemoteDescription, decides it does not want to accept the
   new parameters, and sends a reject message back to the offerer.  Now,
   the offerer, and possibly the answerer as well, need to return to a
   stable state and the previous local/remote description.  To support
   this, we introduce the concept of "rollback".

   A rollback returns the state machine to its previous state, and the
   local or remote description to its previous value.  Any resources or
   candidates that were allocated by the new local description are
   discarded; any media that is received will be processed according to
   the previous session description.

   A rollback is performed by supplying a session description of type
   "rollback" to either setLocalDescription or setRemoteDescription,
   depending on which needs to be rolled back (i.e. if the new offer was
   supplied to setLocalDescription, the rollback should be done on
   setLocalDescription as well.)

4.1.4.  setLocalDescription

   The setLocalDescription method instructs the PeerConnection to apply
   the supplied SDP blob as its local configuration.  The type field
   indicates whether the blob should be processed as an offer,
   provisional answer, or final answer; offers and answers are checked
   differently, using the various rules that exist for each SDP line.

   This API changes the local media state; among other things, it sets
   up local resources for receiving and decoding media.  In order to
   successfully handle scenarios where the application wants to offer to
   change from one media format to a different, incompatible format, the



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   PeerConnection must be able to simultaneously support use of both the
   old and new local descriptions (e.g. support codecs that exist in
   both descriptions) until a final answer is received, at which point
   the PeerConnection can fully adopt the new local description, or roll
   back to the old description if the remote side denied the change.

   This API indirectly controls the candidate gathering process.  When a
   local description is supplied, and the number of transports currently
   in use does not match the number of transports needed by the local
   description, the PeerConnection will create transports as needed and
   begin gathering candidates for them.

   If setRemoteDescription was previous called with an offer, and
   setLocalDescription is called with an answer (provisional or final),
   and the media directions are compatible, and media are available to
   send, this will result in the starting of media transmission.

4.1.5.  setRemoteDescription

   The setRemoteDescription method instructs the PeerConnection to apply
   the supplied SDP blob as the desired remote configuration.  As in
   setLocalDescription, the type field of the indicates how the blob
   should be processed.

   This API changes the local media state; among other things, it sets
   up local resources for sending and encoding media.

   If setRemoteDescription was previously called with an offer, and
   setLocalDescription is called with an answer (provisional or final),
   and the media directions are compatible, and media are available to
   send, this will result in the starting of media transmission.

4.1.6.  localDescription

   The localDescription method returns a copy of the current local
   configuration, i.e. what was most recently passed to
   setLocalDescription, plus any local candidates that have been
   generated by the ICE Agent.

   TODO: Do we need to expose accessors for both the current and
   proposed local description?

   A null object will be returned if the local description has not yet
   been established, or if the PeerConnection has been closed.

4.1.7.  remoteDescription





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   The remoteDescription method returns a copy of the current remote
   configuration, i.e. what was most recently passed to
   setRemoteDescription, plus any remote candidates that have been
   supplied via processIceMessage.

   TODO: Do we need to expose accessors for both the current and
   proposed remote description?

   A null object will be returned if the remote description has not yet
   been established, or if the PeerConnection has been closed.

4.1.8.  updateIce

   The updateIce method allows the configuration of the ICE Agent to be
   changed during the session, primarily for changing which types of
   local candidates are provided to the application and used for
   connectivity checks.  A callee may initially configure the ICE Agent
   to use only relay candidates, to avoid leaking location information,
   but update this configuration to use all candidates once the call is
   accepted.

   Regardless of the configuration, the gathering process collects all
   available candidates, but excluded candidates will not be surfaced in
   onicecandidate callback or used for connectivity checks.

   This call may result in a change to the state of the ICE Agent, and
   may result in a change to media state if it results in connectivity
   being established.

4.1.9.  addIceCandidate

   The addIceCandidate method provides a remote candidate to the ICE
   Agent, which, if parsed successfully, will be added to the remote
   description according to the rules defined for Trickle ICE.
   Connectivity checks will be sent to the new candidate.

   This call will result in a change to the state of the ICE Agent, and
   may result in a change to media state if it results in connectivity
   being established.

5.  SDP Interaction Procedures

   This section describes the specific procedures to be followed when
   creating and parsing SDP objects.

5.1.  SDP Requirements Overview





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   The key specifications that govern creation and processing of offers
   and answers are listed below.  This list is derived from
   [I-D.ietf-rtcweb-rtp-usage].

   R-1   [RFC4566] is the base SDP specification and MUST be
      implemented.

   R-2   The [RFC5888] grouping framework MUST be implemented for
      signaling grouping information, and MUST be used to identify m=
      lines via the a=mid attribute.

   R-3   [RFC5124] MUST be supported for signaling RTP/SAVPF RTP
      profile.

   R-4   [RFC4585] MUST be implemented to signal RTCP based feedback.

   R-5   [RFC5245] MUST be implemented for signaling the ICE candidate
      lines corresponding to each media stream.

   R-6   [RFC5761] MUST be implemented to signal multiplexing of RTP and
      RTCP.

   R-7   The SDP atributes of "sendonly", "recvonly", "inactive", and
      "sendrecv" from [RFC4566] MUST be implemented to signal
      information about media direction.

   R-8   [RFC5576] MUST be implemented to signal RTP SSRC values.

   R-9   [RFC5763] MUST be implemented to signal DTLS certificate
      fingerprints.

   R-10  [RFC5506] MAY be implemented to signal Reduced-Size RTCP
      messages.

   R-11  [RFC3556] with bandwidth modifiers MAY be supported for
      specifying RTCP bandwidth as a fraction of the media bandwidth,
      RTCP fraction allocated to the senders and setting maximum media
      bit-rate boundaries.

   R-12  [RFC4568] MUST NOT be implemented to signal SDES SRTP keying
      information.

   R-13  A [I-D.ietf-mmusic-msid] MUST be supported, in order to signal
      associations between RTP objects and W3C MediaStreams and
      MediaStreamTracks in a standard way.

   R-14  The bundle mechanism in
      [I-D.ietf-mmusic-sdp-bundle-negotiation] MUST be supported to



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      signal the use or multiplexing RTP somethings on a single UDP
      port, in order to avoid excessive use of port number resources.

   As required by [RFC4566] Section 5.13 JSEP implementations MUST
   ignore unknown attributes (a=) lines.

   Example SDP for RTCWeb call flows can be found in
   [I-D.nandakumar-rtcweb-sdp].  [TODO: since we are starting to specify
   how to handle SDP in this document, should these call flows be merged
   into this document, or this link moved to the examples section?]

5.2.  Constructing an Offer

   When createOffer is called, a new SDP description must be created
   that includes the functionality specified in
   [I-D.ietf-rtcweb-rtp-usage].  The exact details of this process are
   explained below.

5.2.1.  Initial Offers

   When createOffer is called for the first time, the result is known as
   the initial offer.

   The first step in generating an initial offer is to generate session-
   level attributes, as specified in [RFC4566], Section 5.
   Specifically:

   o  The first SDP line MUST be "v=0", as specified in [RFC4566],
      Section 5.1

   o  The second SDP line MUST be an "o=" line, as specified in
      [RFC4566], Section 5.2.  The value of the <username> field SHOULD
      be "-".  The value of the <sess-id> field SHOULD be a
      cryptographically random number.  To ensure uniqueness, this
      number SHOULD be at least 64 bits long.  The value of the <sess-
      version> field SHOULD be zero.  The value of the <nettype>
      <addrtype> <unicast-address> tuple SHOULD be set to a non-
      meaningful address, such as IN IP4 0.0.0.0, to prevent leaking the
      local address in this field.  As mentioned in [RFC4566], the
      entire o= line needs to be unique, but selecting a random number
      for <sess-id> is sufficient to accomplish this.

   o  The third SDP line MUST be a "s=" line, as specified in [RFC4566],
      Section 5.3; a single space SHOULD be used as the session name,
      e.g. "s= "

   o  Session Information ("i="), URI ("u="), Email Address ("e="),
      Phone Number ("p="), Bandwidth ("b="), Repeat Times ("r="), and



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      Time Zones ("z=") lines are not useful in this context and SHOULD
      NOT be included.

   o  Encryption Keys ("k=") lines do not provide sufficient security
      and MUST NOT be included.

   o  A "t=" line MUST be added, as specified in [RFC4566], Section 5.9;
      both <start-time> and <stop-time> SHOULD be set to zero, e.g. "t=0
      0".

   The next step is to generate m= sections for each MediaStreamTrack
   that has been added to the PeerConnection via the addStream method.
   Note that this method takes a MediaStream, which can contain multiple
   MediaStreamTracks, and therefore multiple m= sections can be
   generated even if addStream is only called once.

   Each m= section should be generated as specified in [RFC4566],
   Section 5.14.  The <proto> field MUST be set to "RTP/SAVPF".  If a m=
   section is not being bundled into another m= section, it MUST
   generate a unique set of ICE credentials and gather its own set of
   candidates.  Otherwise, it MUST use the same ICE credentials and
   candidates that were used in the m= section that it is being bundled
   into.  For DTLS, all m= sections MUST use the same certificate [OPEN
   ISSUE: how this is configured] and will therefore have the same
   fingerprint values.

   Each m= section MUST include the following:

   o  An "a=mid" line, as specified in [RFC5888], Section 4.

   o  An "a=msid" line, as specified in [I-D.ietf-mmusic-msid],
      Section 2.

   o  [OPEN ISSUE: Use of App Token versus stream-correlator ]

   o  An "a=sendrecv" line, as specified in [RFC3264], Section 5.1.

   o  For each supported codec, "a=rtpmap" and "a=fmtp" lines, as
      specified in [RFC4566], Section 6.  For audio, the codecs
      specified in [I-D.ietf-rtcweb-audio], Section 3, MUST be be
      supported.

   o  For each primary codec where RTP retransmission should be used, a
      corresponding "a=rtpmap" line indicating "rtx" with the clock rate
      of the primary codec and an "a=fmtp" line that references the
      payload type fo the primary codec, as specified in [RFC4588],
      Section 8.1.




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   o  For each supported FEC mechanism, a corresponding "a=rtpmap" line
      indicating the desired FEC codec.

   o  "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245],
      Section 15.4.

   o  An "a=ice-options" line, with the "trickle" option, as specified
      in [I-D.ivov-mmusic-trickle-ice], Section 4.

   o  For each candidate that has been gathered during the most recent
      gathering phase, an "a=candidate" line, as specified in [RFC5245],
      Section 4.3., paragraph 3.

   o  For the current default candidate, a "c=" line, as specific in
      [RFC5245], Section 4.3., paragraph 6.  [OPEN ISSUE, pending
      resolution in mmusic: If no candidates have yet been gathered yet,
      the default candidate should be set to the null value defined in
      [I-D.ivov-mmusic-trickle-ice], Section 5.1.]

   o  An "a=fingerprint" line, as specified in [RFC4572], Section 5.
      Use of the SHA-256 algorithm for the fingerprint is REQUIRED; if
      the browser also supports stronger hashes, additional
      "a=fingerprint" lines with these hashes MAY also be added.

   o  An "a=setup" line, as specified in [RFC4145], Section 4, and
      clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
      The role value in the offer MUST be "actpass".

   o  An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.1.

   o  An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5.

   o  For each supported RTP header extension, an "a=extmap" line, as
      specified in [RFC5285], Section 5.  The list of header extensions
      that SHOULD/MUST be supported is specified in
      [I-D.ietf-rtcweb-rtp-usage], Section 5.2.  Any header extensions
      that require encryption MUST be specified as indicated in
      [RFC6904], Section 4.

   o  For each supported RTCP feedback mechanism, an "a=rtcp-fb"
      mechanism, as specified in [RFC4585], Section 4.2.  The list of
      RTCP feedback mechanisms that SHOULD/MUST be supported is
      specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.1.

   o  An "a=ssrc" line, as specified in [RFC5576], Section 4.1,
      indicating the SSRC to be used for sending media.





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   o  If RTX is supported for this media type, another "a=ssrc" line
      with the RTX SSRC, and an "a=ssrc-group" line, as specified in
      [RFC5576], section 4.2, with semantics set to "FID" and including
      the primary and RTX SSRCs.

   o  If FEC is supported for this media type, another "a=ssrc" line
      with the FEC SSRC, and an "a=ssrc-group" line, as specified in
      [RFC5576], section 4.2, with semantics set to "FEC" and including
      the primary and FEC SSRCs.

   o  [OPEN ISSUE: Handling of a=imageattr]

   o  [TODO: bundle-only]

   Lastly, if a data channel has been created, a m= section MUST be
   generated for data.  The <media> field MUST be set to "application"
   and the <proto> field MUST be set to "DTLS/SCTP", as specified in
   [I-D.ietf-mmusic-sctp-sdp], Section 3.  The "a=mid", "a=ice-ufrag",
   "a=ice-passwd", "a=ice-options", "a=candidate", "a=fingerprint", and
   "a=setup" lines MUST be included as mentioned above.  [OPEN ISSUE:
   additional SCTP-specific stuff to be included, as indicated in
   [I-D.jesup-rtcweb-data-protocol] (currently none)]

   Once all m= sections have been generated, a session-level "a=group"
   attribute MUST be added as specified in [RFC5888].  This attribute
   MUST have semantics "BUNDLE", and identify the m= sections to be
   bundled.  [OPEN ISSUE: Need to determine exactly how this decision is
   made.]

   Attributes that are common between all m= sections MAY be moved to
   session-level, if desired.

   Attributes other than the ones specified above MAY be included,
   except for the following attributes which are specifically
   incompatible with the requirements of [I-D.ietf-rtcweb-rtp-usage],
   and MUST NOT be included:

   o  "a=crypto"

   o  "a=key-mgmt"

   o  "a=ice-lite"

   Note that when BUNDLE is used, any additional attributes that are
   added MUST follow the advice in
   [I-D.nandakumar-mmusic-sdp-mux-attributes] on how those attributes
   interact with BUNDLE.




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5.2.2.  Subsequent Offers

   When createOffer is called a second (or later) time, the processing
   is different, depending on the current signaling state.

   If the initial offer was not applied using setLocalDescription,
   meaning the PeerConnection is still in the "stable" state, the steps
   for generating an initial offer should be followed, with this
   exception:

   o  The "o=" line MUST stay the same.

   If the initial offer was applied using setLocalDescription, but an
   answer from the remote side has not yet been applied, meaning the
   PeerConnection is still in the "local-offer" state, the steps for
   generating an initial offer should be followed, with these
   exceptions:

   o  The "o=" line MUST stay the same, except for the <session-version>
      field, which MUST increase by 1 from the previously applied local
      description.

   o  The "s=" and "t=" lines MUST stay the same.

   o  Each "a=mid" line MUST stay the same.

   o  Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same.

   o  For MediaStreamTracks that are still present, the "a=msid",
      "a=ssrc", and "a=ssrc-group" lines MUST stay the same.

   o  If any MediaStreamTracks have been removed, either through the
      removeStream method or by removing them from an added MediaStream,
      their m= sections MUST be marked as recvonly by changing the value
      of the [RFC3264] directional attribute to "a=recvonly".  The
      "a=msid", "a=ssrc", and "a=ssrc-group" lines MUST be removed from
      the associated m= sections.

   If the initial offer was applied using setLocalDescription, and an
   answer from the remote side has been applied using
   setRemoteDescription, meaning the PeerConnection is in the "remote-
   pranswer" or "stable" states, an offer is generated based on the
   negotiated session descriptions by following the steps mentioned for
   the "local-offer" state above, along with these exceptions: [OPEN
   ISSUE: should this be permitted in the remote-pranswer state?]

   o  If a m= section was rejected, i.e. has had its port set to zero in
      either the local or remote description, it MUST remain rejected



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      and have a zero port in the new offer, as indicated in RFC3264,
      Section 5.1.

   o  If a m= section exists in the current local description, but has
      its state set to inactive or recvonly, and a new MediaStreamTrack
      is added, the previously existing m= section MUST be recycled
      instead of creating a new m= section.  [OPEN ISSUE: Nail down
      exactly what this means.  Should the codecs remain the same?
      (No.)  Should ICE restart?  (No.)  Can the "a=mid" attribute be
      changed?  (Yes?)]

   o  If a m= section exists in the current local description, but does
      not have an associated MediaStreamTrack (i.e. it is inactive or
      recvonly), a corresponding m= section MUST be generated in the new
      offer, but without "a=msid", "a=ssrc", or "a=ssrc-group"
      attributes, and the appropriate directional attribute must be
      specified.

   In addition, for each previously existing, non-rejected m= section in
   the new offer, the following adjustments are made based on the
   contents of the corresponding m= section in the current remote
   description:

   o  The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
      only include codecs present in the remote description.

   o  The RTP header extensions MUST only include those that are present
      in the remote description.

   o  The RTCP feedback extensions MUST only include those that are
      present in the remote description.

   o  The "a=rtcp-mux" line MUST only be added if present in the remote
      description.

   o  The "a=rtcp-rsize" line MUST only be added if present in the
      remote description.

5.2.3.  Constraints Handling

   The createOffer method takes as a parameter a MediaConstraints
   object.  Special processing is performed when generating a SDP
   description if the following constraints are present.

5.2.3.1.  OfferToReceiveAudio

   If the "OfferToReceiveAudio" constraint is specified, with a value of
   "true", the offer MUST include a non-rejected m= section with media



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   type "audio", even if no audio MediaStreamTrack has been added to the
   PeerConnection.  This allows the offerer to receive audio even when
   not sending it; accordingly, the directional attribute on the audio
   m= section MUST be set to recvonly.  If this constraint is specified
   when an audio MediaStreamTrack has already been added to the
   PeerConnection, or a non-rejected m= section with media type "audio"
   previously existed, it has no effect.

5.2.3.2.  OfferToReceiveVideo

   If the "OfferToReceiveAudio" constraint is specified, with a value of
   "true", the offer MUST include a m= section with media type "video",
   even if no video MediaStreamTrack has been added to the
   PeerConnection.  This allows the offerer to receive video even when
   not sending it; accordingly, the directional attribute on the video
   m= section MUST be set to recvonly.  If this constraint is specified
   when an video MediaStreamTrack has already been added to the
   PeerConnection, or a non-rejected m= section with media type "video"
   previously existed, it has no effect.

5.2.3.3.  VoiceActivityDetection

   If the "VoiceActivityDetection" constraint is specified, with a value
   of "true", the offer MUST indicate support for silence suppression by
   including comfort noise ("CN") codecs for each supported clock rate,
   as specified in [RFC3389], Section 5.1.  [OPEN issue: should this do
   anything in signaling, or should it just control built-in DTX modes
   in audio codecs?  Opus has built-in DTX, but G.711 does not.]

5.2.3.4.  IceRestart

   If the "IceRestart" constraint is specified, with a value of "true",
   the offer MUST indicate an ICE restart by generating new ICE ufrag
   and pwd attributes, as specified in RFC5245, Section 9.1.1.1.  If
   this constraint is specified on an initial offer, it has no effect
   (since a new ICE ufrag and pwd are already generated).

5.3.  Generating an Answer

   When createAnswer is called, a new SDP description must be created
   that is compatible with the supplied remote description as well as
   the requirements specified in [I-D.ietf-rtcweb-rtp-usage].  The exact
   details of this process are explained below.

5.3.1.  Initial Answers

   When createAnswer is called for the first time after a remote
   description has been provided, the result is known as the initial



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   answer.  If no remote description has been installed, an answer
   cannot be generated, and an error MUST be returned.

   Note that the remote description SDP may not have been created by a
   WebRTC endpoint and may not conform to all the requirements listed in
   Section 5.2.  For many cases, this is not a problem.  However, if any
   mandatory SDP attributes are missing, or functionality listed as
   mandatory-to-use is not present (e.g. ICE, DTLS) [TODO: find
   reference for this], this MUST be treated as an error.  [OPEN ISSUE:
   Should this cause setRemoteDescription to fail, or should this cause
   createAnswer to reject those particular m= sections?]

   The first step in generating an initial answer is to generate
   session-level attributes.  The process here is identical to that
   indicated in the Initial Offers section above, with the addition that

   The next step is to generate m= sections for each m= section that is
   present in the remote offer, as specified in [RFC3264], Section 6.
   For the purposes of this discussion, any session-level attributes in
   the offer that are also valid as media-level attributes SHALL be
   considered to be present in each m= section.

   If any of the offered m= sections have been rejected, by stopping the
   associated remote MediaStreamTrack, the corresponding m= section in
   the answer MUST be marked as rejected by setting the port in the m=
   line to zero, as indicated in [RFC3264], Section 6., and processing
   continues with the next m= section.

   For each non-rejected m= section of a given media type, if there is a
   local MediaStreamTrack of the specified type which has been added to
   the PeerConnection via addStream and not yet associated with a m=
   section, the MediaStreamTrack is associated with the m= section at
   this time.  If there are more m= sections of a certain type than
   MediaStreamTracks, some m= sections will not have an associated
   MediaStreamTrack.  If there are more MediaStreamTracks of a certain
   type than m= sections, only the first N MediaStreamTracks will be
   able to be associated in the constructed answer.  The remainder will
   need to be associated in a subsequent offer.

   Each m= section should then generated as specified in [RFC3264],
   Section 6.1.  The <proto> field MUST be set to "RTP/SAVPF".  If the
   offer supports BUNDLE, all m= sections to be BUNDLEd must use the
   same ICE credentials and candidates; all m= sections not being
   BUNDLEd must use unique ICE credentials and candidates.  Each m=
   section MUST include the following:






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   o  If present in the offer, an "a=mid" line, as specified in
      [RFC5888], Section 9.1.  The "mid" value MUST match that specified
      in the offer.

   o  If a local MediaStreamTrack has been associated, an "a=msid" line,
      as specified in [I-D.ietf-mmusic-msid], Section 2.

   o  [OPEN ISSUE: Use of App Token versus stream-correlator ]

   o  If a local MediaStreamTrack has been associated, an "a=sendrecv"
      line, as specified in [RFC3264], Section 6.1.  If no local
      MediaStreamTrack has been associated, an "a=recvonly" line.
      [TODO: handle non-sendrecv offered m= sections]

   o  For each supported codec that is present in the offer, "a=rtpmap"
      and "a=fmtp" lines, as specified in [RFC4566], Section 6, and
      [RFC3264], Section 6.1.  For audio, the codecs specified in
      [I-D.ietf-rtcweb-audio], Section 3, MUST be be supported.  Note
      that for simplicity, the answerer MAY use different payload types
      for codecs than the offerer, as it is not prohibited by
      Section 6.1.

   o  If "rtx" is present in the offer, for each primary codec where RTP
      retransmission should be used, a corresponding "a=rtpmap" line
      indicating "rtx" with the clock rate of the primary codec and an
      "a=fmtp" line that references the payload type fo the primary
      codec, as specified in [RFC4588], Section 8.1.

   o  For each supported FEC mechanism that is present in the offer, a
      corresponding "a=rtpmap" line indicating the desired FEC codec.

   o  "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245],
      Section 15.4.

   o  If the "trickle" ICE option is present in the offer, an "a=ice-
      options" line, with the "trickle" option, as specified in
      [I-D.ivov-mmusic-trickle-ice], Section 4.

   o  For each candidate that has been gathered during the most recent
      gathering phase, an "a=candidate" line, as specified in [RFC5245],
      Section 4.3., paragraph 3.

   o  For the current default candidate, a "c=" line, as specific in
      [RFC5245], Section 4.3., paragraph 6.  [OPEN ISSUE, pending
      resolution in mmusic: If no candidates have yet been gathered yet,
      the default candidate should be set to the null value defined in
      [I-D.ivov-mmusic-trickle-ice], Section 5.1.]




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   o  An "a=fingerprint" line, as specified in [RFC4572], Section 5.
      Use of the SHA-256 algorithm for the fingerprint is REQUIRED; if
      the browser also supports stronger hashes, additional
      "a=fingerprint" lines with these hashes MAY also be added.

   o  An "a=setup" line, as specified in [RFC4145], Section 4, and
      clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
      The role value in the answer MUST be "active" or "passive"; the
      "active" role is RECOMMENDED.

   o  If present in the offer, an "a=rtcp-mux" line, as specified in
      [RFC5761], Section 5.1.1.

   o  If present in the offer, an "a=rtcp-rsize" line, as specified in
      [RFC5506], Section 5.

   o  For each supported RTP header extension that is present in the
      offer, an "a=extmap" line, as specified in [RFC5285], Section 5.
      The list of header extensions that SHOULD/MUST be supported is
      specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.2.  Any header
      extensions that require encryption MUST be specified as indicated
      in [RFC6904], Section 4.

   o  For each supported RTCP feedback mechanism that is present in the
      offer, an "a=rtcp-fb" mechanism, as specified in [RFC4585],
      Section 4.2.  The list of RTCP feedback mechanisms that SHOULD/
      MUST be supported is specified in [I-D.ietf-rtcweb-rtp-usage],
      Section 5.1.

   o  If a local MediaStreamTrack has been associated, an "a=ssrc" line,
      as specified in [RFC5576], Section 4.1, indicating the SSRC to be
      used for sending media.

   o  If a local MediaStreamTrack has been associated, and RTX has been
      negotiated for this m= section, another "a=ssrc" line with the RTX
      SSRC, and an "a=ssrc-group" line, as specified in [RFC5576],
      section 4.2, with semantics set to "FID" and including the primary
      and RTX SSRCs.

   o  If a local MediaStreamTrack has been associated, and FEC has been
      negotiated for this m= section, another "a=ssrc" line with the FEC
      SSRC, and an "a=ssrc-group" line, as specified in [RFC5576],
      section 4.2, with semantics set to "FEC" and including the primary
      and FEC SSRCs.

   o  [OPEN ISSUE: Handling of a=imageattr]

   o  [TODO: bundle-only]



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   If a data channel m= section has been offered, a m= section MUST also
   be generated for data.  The <media> field MUST be set to
   "application" and the <proto> field MUST be set to "DTLS/SCTP", as
   specified in [I-D.ietf-mmusic-sctp-sdp], Section 3.  The "a=mid", "a
   =ice-ufrag", "a=ice-passwd", "a=ice-options", "a=candidate",
   "a=fingerprint", and "a=setup" lines MUST be included as mentioned
   above.  [OPEN ISSUE: additional SCTP-specific stuff to be included,
   as indicated in [I-D.jesup-rtcweb-data-protocol] (currently none)]

   [TODO: processing of BUNDLE group]

   Attributes that are common between all m= sections MAY be moved to
   session-level, if desired.

   The attributes prohibited in creation of offers are also prohibited
   in the creation of answers.

5.3.2.  Subsequent Answers

5.3.3.  Constraints Handling

5.4.  Parsing an Offer

5.5.  Parsing an Answer

5.6.  Applying a Local Description

5.7.  Applying a Remote Description

6.  Configurable SDP Parameters

   Note: This section is still very early and is likely to significantly
   change as we get a better understanding of a) the use cases for this
   b) the implications at the protocol level c) feedback from
   implementors on what they can do.

   The following elements of the SDP media description MUST NOT be
   changed between the createOffer and the setLocalDescription, since
   they reflect transport attributes that are solely under browser
   control, and the browser MUST NOT honor an attempt to change them:

   o  The number, type and port number of m-lines.

   o  The generated ICE credentials (a=ice-ufrag and a=ice-pwd).

   o  The set of ICE candidates and their parameters (a=candidate).





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   The following modifications, if done by the browser to a description
   between createOffer/createAnswer and the setLocalDescription, MUST be
   honored by the browser:

   o  Remove or reorder codecs (m=)

   The following parameters may be controlled by constraints passed into
   createOffer/createAnswer.  As an open issue, these changes may also
   be be performed by manipulating the SDP returned from createOffer/
   createAnswer, as indicated above, as long as the capabilities of the
   endpoint are not exceeded (e.g. asking for a resolution greater than
   what the endpoint can encode):

   o  disable BUNDLE (a=group)

   o  disable RTCP mux (a=rtcp-mux)

   o  change send resolution or frame rate

   o  change desired recv resolution or frame rate

   o  change maximum total bandwidth (b=) [OPEN ISSUE: need to clarify
      if this is CT or AS - see section 5.8 of [RFC4566]]

   o  remove desired AVPF mechanisms (a=rtcp-fb)

   o  remove RTP header extensions (a=extmap)

   o  change media send/recv state (a=sendonly/recvonly/inactive)

   For example, an application could implement call hold by adding an
   a=inactive attribute to its local description, and then applying and
   signaling that description.

   The application can also modify the SDP to reduce the capabilities in
   the offer it sends to the far side in any way the application sees
   fit, as long as it is a valid SDP offer and specifies a subset of
   what the browser is expecting to do.

   As always, the application is solely responsible for what it sends to
   the other party, and all incoming SDP will be processed by the
   browser to the extent of its capabilities.  It is an error to assume
   that all SDP is well-formed; however, one should be able to assume
   that any implementation of this specification will be able to
   process, as a remote offer or answer, unmodified SDP coming from any
   other implementation of this specification.





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7.  Security Considerations

   The intent of the WebRTC protocol suite is to provide an environment
   that is securable by default: all media is encrypted, keys are
   exchanged in a secure fashion, and the Javascript API includes
   functions that can be used to verify the identity of communication
   partners.

8.  IANA Considerations

   This document requires no actions from IANA.

9.  Acknowledgements

   Significant text incorporated in the draft as well and review was
   provided by Harald Alvestrand and Suhas Nandakumar.  Dan Burnett,
   Neil Stratford, Eric Rescorla, Anant Narayanan, Andrew Hutton,
   Richard Ejzak, and Adam Bergkvist all provided valuable feedback on
   this proposal.  Matthew Kaufman provided the observation that keeping
   state out of the browser allows a call to continue even if the page
   is reloaded.

10.  References

10.1.  Normative References

   [I-D.ietf-mmusic-msid]
              Alvestrand, H., "Cross Session Stream Identification in
              the Session Description Protocol", draft-ietf-mmusic-
              msid-01 (work in progress), August 2013.

   [I-D.ietf-mmusic-sctp-sdp]
              Loreto, S. and G. Camarillo, "Stream Control Transmission
              Protocol (SCTP)-Based Media Transport in the Session
              Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-04
              (work in progress), June 2013.

   [I-D.ietf-mmusic-sdp-bundle-negotiation]
              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Multiplexing Negotiation Using Session Description
              Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp-
              bundle-negotiation-04 (work in progress), June 2013.

   [I-D.ietf-rtcweb-audio]
              Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
              Requirements", draft-ietf-rtcweb-audio-02 (work in
              progress), August 2013.




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   [I-D.ietf-rtcweb-rtp-usage]
              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
              Communication (WebRTC): Media Transport and Use of RTP",
              draft-ietf-rtcweb-rtp-usage-09 (work in progress),
              September 2013.

   [I-D.nandakumar-mmusic-sdp-mux-attributes]
              Nandakumar, S., "A Framework for SDP Attributes when
              Multiplexing", draft-nandakumar-mmusic-sdp-mux-
              attributes-03 (work in progress), July 2013.

   [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.

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

   [RFC4145]  Yon, D. and G. Camarillo, "TCP-Based Media Transport in
              the Session Description Protocol (SDP)", RFC 4145,
              September 2005.

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

   [RFC4572]  Lennox, J., "Connection-Oriented Media Transport over the
              Transport Layer Security (TLS) Protocol in the Session
              Description Protocol (SDP)", RFC 4572, July 2006.

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
              2006.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, February 2008.

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




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   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, July 2008.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761, April 2010.

   [RFC5888]  Camarillo, G. and H. Schulzrinne, "The Session Description
              Protocol (SDP) Grouping Framework", RFC 5888, June 2010.

   [RFC6904]  Lennox, J., "Encryption of Header Extensions in the Secure
              Real-time Transport Protocol (SRTP)", RFC 6904, April
              2013.

10.2.  Informative References

   [I-D.ivov-mmusic-trickle-ice]
              Ivov, E., Rescorla, E., and J. Uberti, "Trickle ICE:
              Incremental Provisioning of Candidates for the Interactive
              Connectivity Establishment (ICE) Protocol", draft-ivov-
              mmusic-trickle-ice-01 (work in progress), March 2013.

   [I-D.jennings-rtcweb-signaling]
              Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/
              Answer Protocol (ROAP)", draft-jennings-rtcweb-
              signaling-01 (work in progress), October 2011.

   [I-D.jesup-rtcweb-data-protocol]
              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
              Protocol", draft-jesup-rtcweb-data-protocol-04 (work in
              progress), February 2013.

   [I-D.nandakumar-rtcweb-sdp]
              Nandakumar, S. and C. Jennings, "SDP for the WebRTC",
              draft-nandakumar-rtcweb-sdp-02 (work in progress), July
              2013.

   [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
              Comfort Noise (CN)", RFC 3389, September 2002.

   [RFC3556]  Casner, S., "Session Description Protocol (SDP) Bandwidth
              Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC
              3556, July 2003.

   [RFC3960]  Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
              Tone Generation in the Session Initiation Protocol (SIP)",
              RFC 3960, December 2004.





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   [RFC4568]  Andreasen, F., Baugher, M., and D. Wing, "Session
              Description Protocol (SDP) Security Descriptions for Media
              Streams", RFC 4568, July 2006.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              July 2006.

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, April 2009.

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, June 2009.

   [RFC5763]  Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
              for Establishing a Secure Real-time Transport Protocol
              (SRTP) Security Context Using Datagram Transport Layer
              Security (DTLS) ", RFC 5763, May 2010.

   [W3C.WD-webrtc-20111027]
              Bergkvist, A., Burnett, D., Narayanan, A., and C.
              Jennings, "WebRTC 1.0: Real-time Communication Between
              Browsers", World Wide Web Consortium WD WD-
              webrtc-20111027, October 2011,
              <http://www.w3.org/TR/2011/WD-webrtc-20111027>.

Appendix A.  JSEP Implementation Examples

A.1.  Example API Flows

   Below are several sample flows for the new PeerConnection and library
   APIs, demonstrating when the various APIs are called in different
   situations and with various transport protocols.  For clarity and
   simplicity, the createOffer/createAnswer calls are assumed to be
   synchronous in these examples, whereas the actual APIs are async.

A.1.1.  Call using ROAP

   This example demonstrates a ROAP call, without the use of trickle
   candidates.

   // Call is initiated toward Answerer
   OffererJS->OffererUA:   pc = new PeerConnection();
   OffererJS->OffererUA:   pc.addStream(localStream, null);
   OffererUA->OffererJS:   iceCallback(candidate);
   OffererJS->OffererUA:   offer = pc.createOffer(null);



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   OffererJS->OffererUA:   pc.setLocalDescription("offer", offer);
   OffererJS->AnswererJS:  {"type":"OFFER", "sdp":offer }

   // OFFER arrives at Answerer
   AnswererJS->AnswererUA: pc = new PeerConnection();
   AnswererJS->AnswererUA: pc.setRemoteDescription("offer", msg.sdp);
   AnswererUA->AnswererJS: onaddstream(remoteStream);
   AnswererUA->OffererUA:  iceCallback(candidate);

   // Answerer accepts call
   AnswererJS->AnswererUA: pc.addStream(localStream, null);
   AnswererJS->AnswererUA: answer = pc.createAnswer(msg.sdp, null);
   AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
   AnswererJS->OffererJS:  {"type":"ANSWER","sdp":answer }

   // ANSWER arrives at Offerer
   OffererJS->OffererUA:   pc.setRemoteDescription("answer", answer);
   OffererUA->OffererJS:   onaddstream(remoteStream);

   // ICE Completes (at Answerer)
   AnswererUA->OffererUA:  Media

   // ICE Completes (at Offerer)
   OffererJS->AnswererJS:  {"type":"OK" }
   OffererUA->AnswererUA:  Media


A.1.2.  Call using XMPP

   This example demonstrates an XMPP call, making use of trickle
   candidates.

   // Call is initiated toward Answerer
   OffererJS->OffererUA:   pc = new PeerConnection();
   OffererJS->OffererUA:   pc.addStream(localStream, null);
   OffererJS->OffererUA:   offer = pc.createOffer(null);
   OffererJS->OffererUA:   pc.setLocalDescription("offer", offer);
   OffererJS:              xmpp = createSessionInitiate(offer);
   OffererJS->AnswererJS:  <jingle action="session-initiate"/>

   OffererJS->OffererUA:   pc.startIce();
   OffererUA->OffererJS:   onicecandidate(cand);
   OffererJS:              createTransportInfo(cand);
   OffererJS->AnswererJS:  <jingle action="transport-info"/>

   // session-initiate arrives at Answerer
   AnswererJS->AnswererUA: pc = new PeerConnection();
   AnswererJS:             offer = parseSessionInitiate(xmpp);



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   AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
   AnswererUA->AnswererJS: onaddstream(remoteStream);

   // transport-infos arrive at Answerer
   AnswererJS->AnswererUA: candidate = parseTransportInfo(xmpp);
   AnswererJS->AnswererUA: pc.addIceCandidate(candidate);
   AnswererUA->AnswererJS: onicecandidate(cand)
   AnswererJS:             createTransportInfo(cand);
   AnswererJS->OffererJS:  <jingle action="transport-info"/>

   // transport-infos arrive at Offerer
   OffererJS->OffererUA:   candidates = parseTransportInfo(xmpp);
   OffererJS->OffererUA:   pc.addIceCandidate(candidates);

   // Answerer accepts call
   AnswererJS->AnswererUA: pc.addStream(localStream, null);
   AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
   AnswererJS:             xmpp = createSessionAccept(answer);
   AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
   AnswererJS->OffererJS:  <jingle action="session-accept"/>

   // session-accept arrives at Offerer
   OffererJS:              answer = parseSessionAccept(xmpp);
   OffererJS->OffererUA:   pc.setRemoteDescription("answer", answer);
   OffererUA->OffererJS:   onaddstream(remoteStream);

   // ICE Completes (at Answerer)
   AnswererUA->OffererUA:  Media

   // ICE Completes (at Offerer)
   OffererUA->AnswererUA:  Media


A.1.3.  Adding video to a call, using XMPP

   This example demonstrates an XMPP call, where the XMPP content-add
   mechanism is used to add video media to an existing session.  For
   simplicity, candidate exchange is not shown.

   Note that the offerer for the change to the session may be different
   than the original call offerer.

   // Offerer adds video stream
   OffererJS->OffererUA:   pc.addStream(videoStream)
   OffererJS->OffererUA:   offer = pc.createOffer(null);
   OffererJS:              xmpp = createContentAdd(offer);
   OffererJS->OffererUA:   pc.setLocalDescription("offer", offer);
   OffererJS->AnswererJS:  <jingle action="content-add"/>



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   // content-add arrives at Answerer
   AnswererJS:             offer = parseContentAdd(xmpp);
   AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
   AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
   AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
   AnswererJS:             xmpp = createContentAccept(answer);
   AnswererJS->OffererJS:  <jingle action="content-accept"/>

   // content-accept arrives at Offerer
   OffererJS:              answer = parseContentAccept(xmpp);
   OffererJS->OffererUA:   pc.setRemoteDescription("answer", answer);


A.1.4.  Simultaneous add of video streams, using XMPP

   This example demonstrates an XMPP call, where new video sources are
   added at the same time to a call that already has video; since adding
   these sources only affects one side of the call, there is no
   conflict.  The XMPP description-info mechanism is used to indicate
   the new sources to the remote side.

   // Offerer and "Answerer" add video streams at the same time
   OffererJS->OffererUA:   pc.addStream(offererVideoStream2)
   OffererJS->OffererUA:   offer = pc.createOffer(null);
   OffererJS:              xmpp = createDescriptionInfo(offer);
   OffererJS->OffererUA:   pc.setLocalDescription("offer", offer);
   OffererJS->AnswererJS:  <jingle action="description-info"/>

   AnswererJS->AnswererUA: pc.addStream(answererVideoStream2)
   AnswererJS->AnswererUA: offer = pc.createOffer(null);
   AnswererJS:             xmpp = createDescriptionInfo(offer);
   AnswererJS->AnswererUA: pc.setLocalDescription("offer", offer);
   AnswererJS->OffererJS:  <jingle action="description-info"/>

   // description-info arrives at "Answerer", and is acked
   AnswererJS:             offer = parseDescriptionInfo(xmpp);
   AnswererJS->OffererJS:  <iq type="result"/>  // ack

   // description-info arrives at Offerer, and is acked
   OffererJS:              offer = parseDescriptionInfo(xmpp);
   OffererJS->AnswererJS:  <iq type="result"/>  // ack

   // ack arrives at Offerer; remote offer is used as an answer
   OffererJS->OffererUA:   pc.setRemoteDescription("answer", offer);

   // ack arrives at "Answerer"; remote offer is used as an answer
   AnswererJS->AnswererUA: pc.setRemoteDescription("answer", offer);




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A.1.5.  Call using SIP

   This example demonstrates a simple SIP call (e.g. where the client
   talks to a SIP proxy over WebSockets).

   // Call is initiated toward Answerer
   OffererJS->OffererUA:   pc = new PeerConnection();
   OffererJS->OffererUA:   pc.addStream(localStream, null);
   OffererUA->OffererJS:   onicecandidate(candidate);
   OffererJS->OffererUA:   offer = pc.createOffer(null);
   OffererJS->OffererUA:   pc.setLocalDescription("offer", offer);
   OffererJS:              sip = createInvite(offer);
   OffererJS->AnswererJS:  SIP INVITE w/ SDP

   // INVITE arrives at Answerer
   AnswererJS->AnswererUA: pc = new PeerConnection();
   AnswererJS:             offer = parseInvite(sip);
   AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
   AnswererUA->AnswererJS: onaddstream(remoteStream);
   AnswererUA->OffererUA:  onicecandidate(candidate);

   // Answerer accepts call
   AnswererJS->AnswererUA: pc.addStream(localStream, null);
   AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
   AnswererJS:             sip = createResponse(200, answer);
   AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
   AnswererJS->OffererJS:  200 OK w/ SDP

   // 200 OK arrives at Offerer
   OffererJS:              answer = parseResponse(sip);
   OffererJS->OffererUA:   pc.setRemoteDescription("answer", answer);
   OffererUA->OffererJS:   onaddstream(remoteStream);
   OffererJS->AnswererJS:  ACK

   // ICE Completes (at Answerer)
   AnswererUA->OffererUA:  Media

   // ICE Completes (at Offerer)
   OffererUA->AnswererUA:  Media


A.1.6.  Handling early media (e.g. 1-800-GO FEDEX), using SIP

   This example demonstrates how early media could be handled; for
   simplicity, only the offerer side of the call is shown.

   // Call is initiated toward Answerer
   OffererJS->OffererUA:   pc = new PeerConnection();



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   OffererJS->OffererUA:   pc.addStream(localStream, null);
   OffererUA->OffererJS:   onicecandidate(candidate);
   OffererJS->OffererUA:   offer = pc.createOffer(null);
   OffererJS->OffererUA:   pc.setLocalDescription("offer", offer);
   OffererJS:              sip = createInvite(offer);
   OffererJS->AnswererJS:  SIP INVITE w/ SDP

   // 180 Ringing is received by offerer, w/ SDP
   OffererJS:              answer = parseResponse(sip);
   OffererJS->OffererUA:   pc.setRemoteDescription("pranswer", answer);
   OffererUA->OffererJS:   onaddstream(remoteStream);

   // ICE Completes (at Offerer)
   OffererUA->AnswererUA:  Media

   // 200 OK arrives at Offerer
   OffererJS:              answer = parseResponse(sip);
   OffererJS->OffererUA:   pc.setRemoteDescription("answer", answer);
   OffererJS->AnswererJS:  ACK


A.2.  Example Session Descriptions

A.2.1.  createOffer

   This SDP shows a typical initial offer, created by createOffer for a
   PeerConnection with a single audio MediaStreamTrack, a single video
   MediaStreamTrack, and a single data channel.  Host candidates have
   also already been gathered.  Note some lines have been broken into
   two lines for formatting reasons.

   v=0
   o=- 4962303333179871722 1 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=group:BUNDLE audio video data
   m=audio 56500 RTP/SAVPF 111 0 8 126
   c=IN IP4 192.0.2.1
   a=rtcp:56501 IN IP4 192.0.2.1
   a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56500
               typ host generation 0
   a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56501
               typ host generation 0
   a=ice-ufrag:ETEn1v9DoTMB9J4r
   a=ice-pwd:OtSK0WpNtpUjkY4+86js7ZQl
   a=ice-options:trickle
   a=mid:audio
   a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level



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   a=sendrecv
   a=rtcp-mux
   a=rtcp-rsize
   a=fingerprint:sha-256
                 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
                :BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
   a=setup:actpass
   a=rtpmap:111 opus/48000/2
   a=fmtp:111 minptime=10
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:126 telephone-event/8000
   a=maxptime:60
   a=ssrc:1732846380 cname:EocUG1f0fcg/yvY7
   a=msid:47017fee-b6c1-4162-929c-a25110252400
          f83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
   m=video 56502 RTP/SAVPF 100 115 116 117
   c=IN IP4 192.0.2.1
   a=rtcp:56503 IN IP4 192.0.2.1
   a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56502
               typ host generation 0
   a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56503
               typ host generation 0
   a=ice-ufrag:BGKkWnG5GmiUpdIV
   a=ice-pwd:mqyWsAjvtKwTGnvhPztQ9mIf
   a=ice-options:trickle
   a=mid:video
   a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
   a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
   a=sendrecv
   a=rtcp-mux
   a=rtcp-rsize
   a=fingerprint:sha-256
                 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
                :BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
   a=setup:actpass
   a=rtpmap:100 VP8/90000
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 goog-remb
   a=rtpmap:115 rtx/90000
   a=fmtp:115 apt=100
   a=rtpmap:116 red/90000
   a=rtpmap:117 ulpfec/90000
   a=ssrc:1366781083 cname:EocUG1f0fcg/yvY7
   a=ssrc:1366781084 cname:EocUG1f0fcg/yvY7
   a=ssrc:1366781085 cname:EocUG1f0fcg/yvY7
   a=ssrc-group:FID 1366781083 1366781084



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   a=ssrc-group:FEC 1366781083 1366781085
   a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
          f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0
   m=application 56504 DTLS/SCTP 5000
   c=IN IP4 192.0.2.1
   a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56504
               typ host generation 0
   a=ice-ufrag:VD5v2BnbZm3mgP3d
   a=ice-pwd:+Jlkuox+VVIUDqxcfIDuTZMH
   a=ice-options:trickle
   a=mid:data
   a=fingerprint:sha-256 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
                        :BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
   a=setup:actpass
   a=fmtp:5000 protocol=webrtc-datachannel; streams=10


A.2.2.  createAnswer

   This SDP shows a typical initial answer to the above offer, created
   by createAnswer for a PeerConnection with a single audio
   MediaStreamTrack, a single video MediaStreamTrack, and a single data
   channel.  Host candidates have also already been gathered.  Note some
   lines have been broken into two lines for formatting reasons.

   v=0
   o=- 6729291447651054566 1 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=group:BUNDLE audio video data
   m=audio 20000 RTP/SAVPF 111 0 8 126
   c=IN IP4 192.0.2.2
   a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
               typ host generation 0
   a=ice-ufrag:6sFvz2gdLkEwjZEr
   a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2
   a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
                        :DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
   a=setup:active
   a=mid:audio
   a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=sendrecv
   a=rtcp-mux
   a=rtpmap:111 opus/48000/2
   a=fmtp:111 minptime=10
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:126 telephone-event/8000



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   a=maxptime:60
   a=ssrc:3429951804 cname:Q/NWs1ao1HmN4Xa5
   a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
          PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1a0
   m=video 20000 RTP/SAVPF 100 115 116 117
   c=IN IP4 192.0.2.2
   a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
               typ host generation 0
   a=ice-ufrag:6sFvz2gdLkEwjZEr
   a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2
   a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
                        :DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
   a=setup:active
   a=mid:video
   a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
   a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
   a=sendrecv
   a=rtcp-mux
   a=rtpmap:100 VP8/90000
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 goog-remb
   a=rtpmap:115 rtx/90000
   a=fmtp:115 apt=100
   a=rtpmap:116 red/90000
   a=rtpmap:117 ulpfec/90000
   a=ssrc:3229706345 cname:Q/NWs1ao1HmN4Xa5
   a=ssrc:3229706346 cname:Q/NWs1ao1HmN4Xa5
   a=ssrc:3229706347 cname:Q/NWs1ao1HmN4Xa5
   a=ssrc-group:FID 3229706345 3229706346
   a=ssrc-group:FEC 3229706345 3229706347
   a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
          PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1v0
   m=application 20000 DTLS/SCTP 5000
   c=IN IP4 192.0.2.2
   a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
               typ host generation 0
   a=ice-ufrag:6sFvz2gdLkEwjZEr
   a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2
   a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
                        :DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
   a=setup:active
   a=mid:data
   a=fmtp:5000 protocol=webrtc-datachannel; streams=10


Appendix B.  Change log




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   Changes in draft-04:

   o  Filled in sections on createOffer and createAnswer.

   o  Added SDP examples.

   o  Fixed references.

   Changes in draft-03:

   o  Added text describing relationship to W3C specification

   Changes in draft-02:

   o  Converted from nroff

   o  Removed comparisons to old approaches abandoned by the working
      group

   o  Removed stuff that has moved to W3C specification

   o  Align SDP handling with W3C draft

   o  Clarified section on forking.

   Changes in draft-01:

   o  Added diagrams for architecture and state machine.

   o  Added sections on forking and rehydration.

   o  Clarified meaning of "pranswer" and "answer".

   o  Reworked how ICE restarts and media directions are controlled.

   o  Added list of parameters that can be changed in a description.

   o  Updated suggested API and examples to match latest thinking.

   o  Suggested API and examples have been moved to an appendix.

   Changes in draft -00:

   o  Migrated from draft-uberti-rtcweb-jsep-02.

Authors' Addresses





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   Justin Uberti
   Google
   747 6th Ave S
   Kirkland, WA  98033
   USA

   Email: justin@uberti.name


   Cullen Jennings
   Cisco
   170 West Tasman Drive
   San Jose, CA  95134
   USA

   Email: fluffy@iii.ca



































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