Network Working Group                                          J. Uberti
Internet-Draft                                                    Google
Intended status: Standards Track                             C. Jennings
Expires:  August 29, 2013 March 22, 2014                                            Cisco
                                                       February 25,
                                                      September 18, 2013

               Javascript Session Establishment Protocol
                       draft-ietf-rtcweb-jsep-03
                       draft-ietf-rtcweb-jsep-04

Abstract

   This document describes the mechanisms for allowing a Javascript
   application to fully 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 This Memo

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   This Internet-Draft will expire on August 29, 2013. March 22, 2014.

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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . . .  4   3
     1.1.  General Design of JSEP  . . . . . . . . . . . . . . . . . .  4   3
     1.2.  Other Approaches Considered . . . . . . . . . . . . . . .  6   5
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   6
   3.  Semantics and Syntax  . . . . . . . . . . . . . . . . . . . . .  7   6
     3.1.  Signaling Model . . . . . . . . . . . . . . . . . . . . .  7   6
     3.2.  Session Descriptions and State Machine  . . . . . . . . . .   7
     3.3.  Session Description Format  . . . . . . . . . . . . . . . . 10   9
     3.4.  ICE . . . . . . . . . . . . . . . . . . . . . . . . . . .  10
       3.4.1.  ICE Candidate Trickling . . . . . . . . . . . . . . .  10
         3.4.1.1.  ICE Candidate Format  . . . . . . . . . . . . . . . 11  10
     3.5.  Interactions With Forking . . . . . . . . . . . . . . . .  11
       3.5.1.  Sequential Forking  . . . . . . . . . . . . . . . . . . 12  11
       3.5.2.  Parallel Forking  . . . . . . . . . . . . . . . . . . .  12
     3.6.  Session Rehydration . . . . . . . . . . . . . . . . . . .  13
   4.  Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 14  13
     4.1.  SDP Requirements . . . . . . . . . . . . . . . . . . . . . 14
     4.2.  Methods . . . . . . . . . . . . . . . . . . . . . . . . . 15
       4.2.1.  14
       4.1.1.  createOffer . . . . . . . . . . . . . . . . . . . . . 15
       4.2.2.  14
       4.1.2.  createAnswer  . . . . . . . . . . . . . . . . . . . . . 16
       4.2.3.  15
       4.1.3.  SessionDescriptionType  . . . . . . . . . . . . . . . . 17
         4.2.3.1.  15
         4.1.3.1.  Use of Provisional Answers  . . . . . . . . . . . . 18
         4.2.3.2.  16
         4.1.3.2.  Rollback  . . . . . . . . . . . . . . . . . . . . . 18
       4.2.4.  17
       4.1.4.  setLocalDescription . . . . . . . . . . . . . . . . . 19
       4.2.5.  17
       4.1.5.  setRemoteDescription  . . . . . . . . . . . . . . . . . 19
       4.2.6.  18
       4.1.6.  localDescription  . . . . . . . . . . . . . . . . . . . 20
       4.2.7.  18
       4.1.7.  remoteDescription . . . . . . . . . . . . . . . . . . 20
       4.2.8.  18
       4.1.8.  updateIce . . . . . . . . . . . . . . . . . . . . . . 20
       4.2.9.  19
       4.1.9.  addIceCandidate . . . . . . . . . . . . . . . . . . . 21  19
   5.  SDP Interaction Procedures  . . . . . . . . . . . . . . . . . . 21  19
     5.1.  Constructing an Offer  . .  SDP Requirements Overview . . . . . . . . . . . . . . . . 21  19
     5.2.  Generating  Constructing an Answer . Offer . . . . . . . . . . . . . . . . . .  21
     5.3.  Parsing an Offer
       5.2.1.  Initial Offers  . . . . . . . . . . . . . . . . . . .  21
       5.2.2.  Subsequent Offers . . 21
     5.4.  Parsing an Answer . . . . . . . . . . . . . . . .  25
       5.2.3.  Constraints Handling  . . . . 21
     5.5.  Applying a Local Description . . . . . . . . . . . .  26
         5.2.3.1.  OfferToReceiveAudio . . . 21
     5.6.  Applying a Remote Description . . . . . . . . . . . .  26
         5.2.3.2.  OfferToReceiveVideo . . 21
   6.  Configurable SDP Parameters . . . . . . . . . . . . .  27
         5.2.3.3.  VoiceActivityDetection  . . . . 21
   7.  Security Considerations . . . . . . . . .  27
         5.2.3.4.  IceRestart  . . . . . . . . . . 22
   8.  IANA Considerations . . . . . . . . .  27
     5.3.  Generating an Answer  . . . . . . . . . . . . 23
   9.  Acknowledgements . . . . . .  27
       5.3.1.  Initial Answers . . . . . . . . . . . . . . . . . 23
   10. References . .  27
       5.3.2.  Subsequent Answers  . . . . . . . . . . . . . . . . .  31
       5.3.3.  Constraints Handling  . . . . . . . 23
     10.1. Normative References . . . . . . . . .  31
     5.4.  Parsing an Offer  . . . . . . . . . . 23
     10.2. Informative References . . . . . . . . . .  31
     5.5.  Parsing an Answer . . . . . . . . 24
   Appendix A.  JSEP Implementation Examples . . . . . . . . . . . . 25
     A.1.  Example API Flows  31
     5.6.  Applying a Local Description  . . . . . . . . . . . . . .  31
     5.7.  Applying a Remote Description . . . . . . 25
       A.1.1.  Call using ROAP . . . . . . . .  31

   6.  Configurable SDP Parameters . . . . . . . . . . . 26
       A.1.2.  Call using XMPP . . . . . .  31
   7.  Security Considerations . . . . . . . . . . . . . 26
       A.1.3.  Adding video to a call, using XMPP . . . . . .  33
   8.  IANA Considerations . . . . 28
       A.1.4.  Simultaneous add of video streams, using XMPP . . . . 28
       A.1.5.  Call using SIP . . . . . . . . . . . . .  33
   9.  Acknowledgements  . . . . . . . 29
       A.1.6.  Handling early media (e.g. 1-800-GO FEDEX), using
               SIP . . . . . . . . . . . . . . .  33
   10. References  . . . . . . . . . . 30
   Appendix B.  Change log . . . . . . . . . . . . . . .  33
     10.1.  Normative References . . . . . . 31
   Authors' Addresses . . . . . . . . . . . .  33
     10.2.  Informative References . . . . . . . . . . . . 32

1. . . . . .  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 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.

   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
   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
   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

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 W3C WEBRTC RTCPeerConnection
   interface[W3C.WD-webrtc-20111027] is used to control requirements listed above for both the setup,
   management
   offerer, who first calls setLocalDescription(sdp [offer]) and teardown of a multimedia session.

1.1.  General Design of then
   later setRemoteDescription(sdp [answer]), as well as for the
   answerer, who first calls setRemoteDescription(sdp [offer]) and then
   later setLocalDescription(sdp [answer]).

   JSEP

   The thinking behind WebRTC call setup has been also allows for an answer to fully specify and
   control be treated as provisional by the media plane, but
   application.  Provisional answers provide a way for an answerer to
   communicate initial session parameters back to leave the signaling plane up offerer, in order
   to allow the
   application as much as possible.  The rationale session to begin, while allowing a final answer to be
   specified later.  This concept of a final answer is that different
   applications may prefer important to use different protocols, the
   offer/answer model; when such as an answer is received, any extra
   resources allocated by the
   existing SIP caller can be released, now that the exact
   session configuration is known.  These "resources" can include things
   like extra ICE components, TURN candidates, or Jingle 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 protocols, or something custom
   to level is that only one
   offer can be outstanding for a given session, but from the particular application, perhaps 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 novel use case.  In this
   approach, the key information that needs to SIP CANCEL, another offer can be exchanged is generated even though no
   answer was received for the
   multimedia session description, which specifies first offer.  To support this, the necessary
   transport and JSEP
   media configuration information necessary to establish layer can provide an offer whenever the media plane.

   The browser environment also has its own challenges that pose
   problems Javascript application
   needs one for an embedded signaling 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.  One of 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     |<---------------------------+
            |               |                            |
            |               |                            |
            +---------------+          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
   that the user may reload the web page at any time.  If no other difference
   between the browser is
   fully in charge handling of provisional ("pranswer") and final ("answer")
   answers.

3.3.  Session Description Format

   In the signaling state, WebRTC specification, session descriptions are formatted as
   SDP messages.  While this will result in the loss 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 call when changes to SDP, at least until the time that
   this state is wiped by new encoding eclipsed SDP in popularity.  As a result, JSEP
   currently uses SDP as the reload. internal representation for its session
   descriptions.

   However, if to simplify Javascript processing, and provide for future
   flexibility, the
   state SDP syntax is encapsulated within a
   SessionDescription object, which can be stored at the server, constructed from SDP, and pushed back down 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 new
   page, future to
   simplify handling of SessionDescriptions from Javascript.  In the call
   meantime, Javascript libraries can be resumed with minimal interruption.

   With used to perform these considerations in mind, this document describes
   manipulations.

   Note that most applications should be able to treat the
   Javascript Session Establishment Protocol (JSEP)
   SessionDescriptions produced and consumed by these various API calls
   as opaque blobs; that allows for full is, the application will not need to read or
   change them.  The W3C API will provide appropriate APIs to allow the
   application to control of various session parameters, which will provide
   the signaling state machine from Javascript.  This
   mechanism effectively removes necessary information to the browser almost completely from the
   core signaling flow; the only interface needed is about what sort of
   SessionDescription to produce.

3.4.  ICE

   When a way for new ICE candidate is available, the ICE Agent will notify the
   application via a callback; these candidates will automatically be
   added to pass in the local and remote session descriptions
   negotiated by whatever signaling mechanism is used, and a way description.  When all candidates have
   been gathered, the callback will also be invoked to
   interact with signal that the
   gathering process is complete.

3.4.1.  ICE state machine.

   In this document, the use of JSEP Candidate Trickling

   Candidate trickling 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 technique through which a gateway or MCU.
   This distinction is invisible caller may
   incrementally provide candidates to the browser; it just follows callee after the
   instructions it is given via initial
   offer has been dispatched; the API.

   JSEP's handling semantics of session descriptions is simple "Trickle ICE" are defined
   in [I-D.ivov-mmusic-trickle-ice].  This process allows the callee to
   begin acting upon the call and
   straightforward.  Whenever an offer/answer exchange is needed, 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 side creates an offer the
   call.

   JSEP supports optional candidate trickling by calling a createOffer() API.  The
   application optionally modifies providing APIs that offer,
   provide control and then uses it to set
   up its local config via feedback on the setLocalDescription() API.  The ICE candidate gathering process.
   Applications that support candidate trickling can send the initial
   offer is
   then sent off to immediately and send individual candidates when they get the remote side over its preferred signaling
   mechanism (e.g., WebSockets); upon receipt
   notified of a new candidate; applications that offer, the remote
   party installs it using the setRemoteDescription() API.

   When do not support this
   feature can simply wait for the call indication that gathering is accepted,
   complete, and then create and send their offer, with all the callee uses
   candidates, at this time.

   Upon receipt of trickled candidates, the createAnswer() API receiving application will
   supply them to
   generate an appropriate answer, applies it using
   setLocalDescription(), and sends its ICE Agent.  This triggers the answer back ICE Agent to start
   using the initiator
   over new remote candidates for connectivity checks.

3.4.1.1.  ICE Candidate Format

   As with session descriptions, the signaling channel.  When syntax of the offerer gets that answer, it
   installs it using setRemoteDescription(), and initial setup is
   complete.  This process IceCandidate object
   provides some abstraction, but can be repeated for additional offer/answer
   exchanges.

   Regarding ICE [RFC5245], JSEP decouples the ICE state machine easily converted to and from
   the overall signaling state machine, as the ICE state machine must
   remain in SDP candidate lines.

   The candidate lines are 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 SDP information that
   don't, such is contained
   within IceCandidate, as SIP, they represent the only information can be used needed
   that is not present in the aggregated
   form.  Sending transport information separately can allow for faster
   ICE and DTLS startup, since the necessary roundtrips can occur while
   waiting initial offer (i.e.  for trickle
   candidates).  This information is carried with the remote side to accept the session.

   Through its abstraction of signaling, the JSEP approach does require same syntax as the application
   "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 aware of the signaling process.  While the
   application does not need to understand the contents associated with.  The m line can be identified in
   one of session
   descriptions to set up two ways; either by a call, m-line index, or a MID.  The m-line
   index is a zero-based index, referring to the application must call Nth m-line in the right
   APIs at SDP.
   The MID uses the right times, convert "media stream identification", as defined in
   [RFC5888] , to identify the session descriptions and m-line.  WebRTC implementations creating
   an ICE
   information into the defined messages Candidate object MUST populate both of its chosen 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
   protocol, 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 perform "Sequential Search".
   Although these are primarily signaling level issues that are outside
   the reverse conversion scope of JSEP, they do have some impact on the messages it
   receives from configuration of
   the other side.

   One way to mitigate this media plane which is to provide a Javascript library that
   hides this complexity from relevant.  When forking happens at 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 layer, the Javascript application developer.  For example, this library could easily
   adapt the JSEP API into the API that was proposed responsible 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 make the same fashion, one could also implement other popular signaling
   protocols, including SIP decisions about what media should be sent
   or Jingle.  This allow JSEP to provide
   greater control for the experienced developer without forcing received at any
   additional complexity on the novice developer.

1.2.  Other Approaches Considered

   One approach that was considered instead point of time, as well as which remote endpoint it
   should communicate with; JSEP was to include a
   lightweight signaling protocol.  Instead of providing session
   descriptions is used to make sure 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 media engine
   can make the browser to
   have to understand RTP and handle concepts like signaling glare.  In
   addition, it prevented the application from driving the state machine
   to a desired state, media perform as is needed in required by the page reload case.

   A second approach application.
   The basic operations that was considered but not chosen was to decouple the management of applications can have 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 engine
   do are:

      Start exchanging media 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 given remote peer, but keep all the needed interactions, as well as creating
      resources reserved in the offer.

      Start exchanging media with a large
   API surface that needed to be agreed upon given remote peer, and documented.  In
   addition, these API points could be called in free any order, resulting
      resources 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 offer that was considered was are not being used.

3.5.1.  Sequential Forking

   Sequential forking involves a call being dispatched to keep multiple
   remote callees, where each callee can accept the basic
   session description-oriented API, 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 move
   easily control the mechanism policy for
   generating offers and answers out of the browser.  Instead of
   providing createOffer/createAnswer methods within selecting the browser, this
   approach would instead expose a getCapabilities API which would
   provide desired remote endpoint.
   When an answer arrives from one of the application with callees, the information it needed in order application can
   choose to
   generate its own session descriptions.  This increases apply it either as a provisional answer, leaving open the amount
   possibility of
   work that using a different answer in the application needs to do; future, or apply it needs to know how to
   generate session descriptions from capabilities, and especially how
   to generate as
   a final answer, ending the correct setup flow.

   In a "first-one-wins" situation, the first answer from an arbitrary offer will be applied as
   a final answer, and the
   supported capabilities.  While application will reject any subsequent
   answers.  In SIP parlance, this could certainly would be addressed by
   using ACK + BYE.

   In a library like "last-one-wins" situation, all answers would be applied as
   provisional answers, and any previous call leg will be terminated.
   At some point, the one mentioned above, it basically forces application will end the
   use of said library even for setup process, perhaps
   with a simple example.  Providing
   createOffer/createAnswer avoids timer; at this problem, but still allows
   applications to generate their own offers/answers (to a large extent)
   if they choose, using point, the application could reapply the
   existing remote description generated by createOffer as an
   indication of 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 browser's capabilities.

2.  Terminology

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

3.  Semantics Section 3.1 of
   [RFC3960].  Most SIP devices today only support exchanging media with
   a single device at a time, and Syntax

3.1.  Signaling Model

   JSEP does do not specify a particular signaling model or state machine,
   other than the generic need try to exchange SDP mix multiple early media descriptions in the
   fashion described by [RFC3264] (offer/answer)
   audio sources, as that could result in order for both sides
   of a confusing situation.  For
   example, consider having a European ringback tone mixed together with
   the session to know how to conduct North American ringback tone - the session.  JSEP provides
   mechanisms to create offers resulting sound would not be
   like either tone, and answers, as well as to apply them to
   a session.  However, would confuse the browser is totally decoupled from user.  If the actual
   mechanism by which these offers and answers are communicated signaling
   application wishes to only exchange media with one of the remote side, including addressing, retransmission, forking, and glare
   handling.  These issues are left entirely up to
   endpoints at a time, then from a media engine point of view, this is
   exactly like the application; sequential forking case.

   In the
   application has complete control over which offers and answers get
   handed to parallel forking case where the browser, and when.

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

                      Figure 1: JSEP Signaling Model

3.2.  Session Descriptions and State Machine

   In order Javascript application wishes
   to establish the simultaneously exchange media plane, with multiple peers, the user agent needs specific
   parameters to indicate what to transmit to flow is
   slightly more complex, but the remote side, as well
   as how to handle Javascript application can follow the media
   strategy that [RFC3960] describes using UPDATE.  (It is worth noting
   that use cases where this is received.  These parameters are
   determined by the exchange of session descriptions in offers and
   answers, and there desired behavior are certain details very unusual.)
   The UPDATE approach allows the signaling to this process set up a separate media
   flow for each peer that must be
   handled it wishes to exchange media with.  In JSEP,
   this offer used in the JSEP APIs.

   Whether UPDATE would be formed by simply creating a session description applies to
   new PeerConnection and making sure that the same local side or the remote
   side affects the meaning of that description.  For example, media streams
   have been added into this new PeerConnection.  Then the list
   of codecs sent to new
   PeerConnection object would produce a remote party indicates what SDP offer that could be used by
   the local side is
   willing signaling to receive, which, when intersected with perform the set UPDATE strategy discussed in [RFC3960].

   As a result of codecs
   the remote side supports, specifies what sharing the remote side should send.
   However, not all parameters follow this rule; for example, media streams, the SRTP
   parameters [RFC4568] sent to application will end up
   with N parallel PeerConnection sessions, each with a local and remote party indicate what
   description and their own local and remote addresses.  The media flow
   from these sessions can be managed by specifying SDP direction
   attributes in the local
   side will use to encrypt, and thereby what descriptions, or the remote party should
   expect application can choose to receive; play
   out the remote party will have to accept these
   parameters, with no option media from all sessions mixed together.  Of course, if the
   application wants to choose only keep a different value. single session, it can simply
   terminate the sessions that it no longer needs.

3.6.  Session Rehydration

   In addition, various RFCs put different conditions on the format of
   offers versus answers.  For example, event that the local application state is reinitialized,
   either due to a offer may propose multiple
   SRTP configurations, but an answer may only contain user reload of the page, or a single SRTP
   configuration.

   Lastly, while decision within the exact media parameters are only known only after
   application to reload itself (perhaps to update to a
   offer and an answer have been exchanged, new version), it
   is possible for the
   offerer to receive media after they have sent an offer and before
   they have received keep an answer.  To properly existing session alive, via a process incoming media in called
   "rehydration".  The explicit goal of rehydration is to carry out this case,
   session resumption with no interaction with the offerer's media handler must be aware of remote side other
   than normal call signaling messages.

   With rehydration, the details current signaling state is persisted somewhere
   outside of the offer before 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 answer arrives.

   Therefore, IDs
   in order to handle session descriptions properly, the user
   agent needs:
   1.  To know if a session description pertains to use by the local or remote
       side.
   2.  To know if side continue to work.  Next, a session description is an offer or an answer.
   3.  To allow the new offer to be specified independently of is
   generated by the answer.
   JSEP addresses new PeerConnection; this by adding both a setLocalDescription offer will have new ICE and a
   setRemoteDescription method
   possibly new DTLS-SRTP certificate fingerprints (since the old ICE
   and having session description objects
   contain a type field indicating SRTP state has been lost).  Finally, this offer is used to re-
   initiate the type of session description being
   supplied.  This satisfies the requirements listed above for both with the
   offerer, existing remote endpoint, who first calls setLocalDescription(sdp [offer]) and then
   later setRemoteDescription(sdp [answer]), as well as for simply
   sees the
   answerer, who first calls setRemoteDescription(sdp [offer]) new offer as an in-call renegotiation, and then
   later setLocalDescription(sdp [answer]).

   JSEP also allows for replies with an
   answer to that can be treated supplied to setRemoteDescription.  ICE processing
   proceeds as provisional by usual, and as soon as connectivity is established, the
   application.  Provisional answers provide a
   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 an answerer a new page to
   communicate initial session parameters back acquire the old PeerConnection object.]

4.  Interface

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

4.1.  Methods

4.1.1.  createOffer

   The createOffer method generates a final answer to be
   specified later.  This concept blob of SDP that contains a final answer is important
   [RFC3264] offer with the supported configurations for the session,
   including descriptions of the local MediaStreams attached to this
   PeerConnection, the
   offer/answer model; when such an answer is received, any extra
   resources allocated codec/RTP/RTCP options supported by the caller can be released, now this
   implementation, and any candidates that have been gathered by 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
   Agent.  A constraints parameters may be
   received and applied during call setup.

   In [RFC3264], the constraint at supplied to provide
   additional control over the signaling level is that only one
   offer can be outstanding generated offer.  This constraints
   parameter should allow for a given session, but from the media
   stack level, a new offer can following manipulations to be generated at any point.  For example,
   when using SIP
   performed:

   o  To indicate support for signaling, if one offer is sent, then cancelled
   using a SIP CANCEL, another offer can be generated media type even though if no
   answer was received for MediaStreamTracks
      of that type have been added to the first offer. session (e.g., an audio call
      that wants to receive video.)

   o  To support this, trigger an ICE restart, for the JSEP
   media layer can provide purpose of reestablishing
      connectivity.

   o  For re-offer cases, to request an offer whenever that contains the Javascript application
   needs one for full set
      of supported capabilities, as opposed to just the signaling.  The answerer can send back zero or more
   provisional answers, and finally end currently
      negotiated parameters.

   In the offer-answer exchange initial offer, the generated SDP will contain all desired
   functionality for the session (certain parts that are supported but
   not desired by
   sending a final answer.  The state machine default may be omitted); 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     |<---------------------------+
            |               |                            |
            |               |                            |
            +---------------+          setRemote(ANSWER) |
                 ^   |                                   |
                 |   | setLocal(OFFER)                   |
setRemote(ANSWER)|   |                                   |
                 |   V                                   |
            +---------------+                     +---------------+
            |               |                     |               |
            |               | setRemote(PRANSWER) |               |
            |  Local-Offer  |------------------- >|Remote-Pranswer|
            |               |                     |               |
            |               |----\                |               |----\
            +---------------+    |                +---------------+    |
                           ^     |                               ^     |
                           |     |                               |     |
                           \-----/                               \-----/
                       setLocal(OFFER)               setRemote(PRANSWER)

                       Figure 2: JSEP State Machine

   Aside each SDP line, the
   generation of the SDP will follow the process defined for generating
   an initial offer from these state transitions, there is no other difference
   between the document that specifies the given SDP line.
   The exact handling of provisional ("pranswer") and final ("answer")
   answers.

3.3.  Session Description Format initial offer generation is detailed in
   Section 5.2.1. below.

   In the WebRTC specification, session descriptions are formatted as
   SDP messages.  While this format is not optimal for manipulation from
   Javascript, it event createOffer is widely accepted, and frequently updated with new
   features.  Any alternate encoding of called after the session descriptions would have is established,
   createOffer will generate an offer to keep pace with modify the current session
   based on any changes that have been made to SDP, at least until the time 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
   this new encoding eclipsed specifies the given SDP in popularity.  As a result, JSEP
   currently uses line.  For each new stream, the
   generation of the SDP as must follow the internal representation for its session
   descriptions.

   However, to simplify Javascript processing, and provide process of generating an
   initial offer, as mentioned above.  If no changes have been made, or
   for future
   flexibility, the SDP syntax 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 encapsulated within a
   SessionDescription object, which can be constructed from SDP, and
   detailed in Section 5.2.2. below.

   Session descriptions generated by createOffer must be
   serialized out 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 SDP.  If future specifications agree on a JSON
   format for session descriptions, we could easily enable acquire those
   resources.  Because this object
   to generate and consume that JSON.

   Other methods method may be added need to SessionDescription in inspect the future system state
   to
   simplify handling of SessionDescriptions from Javascript.  In determine the
   meantime, currently available resources, it would may be simple to write a Javascript library to perform
   these manipulations.

3.4.  ICE

   When a implemented
   as an async operation.

   Calling this method may do things such as generate new ICE
   credentials, but does not result in 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 gathering, or cause
   media to signal that the
   gathering process is complete.

3.4.1.  ICE Candidate Trickling

   Candidate trickling is start or stop flowing.

4.1.2.  createAnswer

   The createAnswer method generates a technique through which blob of SDP that contains a caller may
   incrementally provide candidates to
   [RFC3264] SDP answer with the callee after supported configuration for the initial
   offer has been dispatched; session
   that is compatible with the semantics of "Trickle ICE" are defined parameters supplied in [I-D.rescorla-mmusic-ice-trickle].  This process allows the callee
   to begin acting upon the call and setting up offer.  Like
   createOffer, the ICE (and perhaps
   DTLS) connections immediately, without having to wait for returned blob contains descriptions of the caller
   to gather all possible candidates.  This results in faster call
   startup in cases where gathering is not performed prior local
   MediaStreams attached to initating this PeerConnection, the call.

   JSEP supports optional candidate trickling by providing APIs 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 and feedback on over the ICE candidate gathering process.
   Applications generated answer.

   As an answer, the generated SDP will contain a specific configuration
   that support candidate trickling can send specifies how the initial
   offer immediately and send individual candidates when they get media plane should be established; for each
   SDP line, the
   notified generation of a new candidate; applications that do not support this
   feature can simply wait the SDP must follow the process defined
   for generating an answer from the indication document that gathering specifies the given
   SDP line.  The exact handling of answer generation is
   complete, and then create and send their offer, with all detailed in
   Section 5.3. below.

   Session descriptions generated by createAnswer must be immediately
   usable by setLocalDescription; like createOffer, the
   candidates, at this time.

   Upon receipt returned
   description should reflect the current state of trickled candidates, the receiving application will
   supply them system.  Because
   this method may need to its ICE Agent.  This triggers inspect the ICE Agent system state to start
   using 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 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,
   credentials, but can does not trigger candidate gathering or change media
   state.

4.1.3.  SessionDescriptionType

   Session description objects (RTCSessionDescription) may be easily converted of type
   "offer", "pranswer", and "answer".  These types provide information
   as to and from how the SDP candidate lines.

   The candidate lines are description parameter should be parsed, and how the only SDP information
   media state should be changed.

   "offer" indicates that is contained
   within IceCandidate, a description should be parsed as they represent an offer;
   said description may include many possible media configurations.  A
   description used as an "offer" may be applied anytime the only information needed
   that
   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 present a final answer, and so should not result in the initial offer (i.e. for trickle
   candidates).  This information is carried with
   freeing of allocated resources.  It may result in the same syntax as start of media
   transmission, if 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 answer does not specify an inactive media
   direction.  A description used as a "pranswer" may be applied as a
   response to indicate which m=
   line it an "offer", or an update to a previously sent "answer".

   "answer" indicates that a description should be associated with.  The m line parsed as an answer,
   the offer-answer exchange should be considered complete, and any
   resources (decoders, candidates) that are no longer needed can be identified in
   one of two ways; either by
   released.  A description used as an "answer" may be applied as a m-line index,
   response to a "offer", or an update to a MID. previously sent "pranswer".

   The m-line
   index only difference between a provisional and final answer is a zero-based index, referring to that
   the Nth m-line final answer results in the SDP.
   The MID uses the "media stream identification", freeing of any unused resources that
   were allocated as defined in
   [RFC3388] , to identify the m-line.  WebRTC implementations creating
   an ICE Candidate object MUST populate both a result of these fields.
   Implementations receiving an ICE Candidate object SHOULD use the MID
   if they implement that functionality, offer.  As such, the application
   can use some discretion on whether an answer should be applied as
   provisional or final, and can change the m-line index, if not.

3.5.  Interactions With Forking

   Some call signaling systems allow various types type of the session
   description as needed.  For example, in a serial forking where scenario, an
   SDP Offer
   application may be provided receive multiple "final" answers, one from each
   remote endpoint.  The application could choose to more than accept the initial
   answers as provisional answers, and only apply an answer as final
   when it receives one device.  For example, SIP
   [RFC3261] defines both that meets its criteria (e.g. a "Parallel Search" 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 "Sequential Search".
   Although these are primarily signaling level issues that are outside send an "inactive" answer more or less immediately
   after receiving the scope offer, instead of JSEP, they do have some impact on 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 configuration
   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 plane which clipping
   by ensuring the transport is relevant.  When forking happens at 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 layer, protocols.  In these cases, "pranswer" can still allow the Javascript
   application responsible for the
   signaling needs to make warm up the decisions about what media should be sent
   or received at any point of time, as well as which remote transport.

   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
   should communicate with; JSEP is used to make sure could send an answer accepting the media engine
   can make data
   channel for two-way data, and accepting the RTP audio and media perform video tracks as required by
   inactive or receive-only.  It could then ask the application.
   The basic operations that user to accept the applications can have
   call, acquire the local media engine
   do are:
      Start exchanging media to streams, and send a given new offer to the
   remote peer, but keep all side moving the
      resources reserved in audio and video to be two-way media.  By the offer.
      Start exchanging media with a given remote peer,
   time the human has accepted the call and free any
      resources in sent the new offer, it is
   likely that the ICE and DTLS handshaking for all the offer that are not being used.

3.5.1.  Sequential Forking

   Sequential forking involves channels will
   already be set up.

4.1.3.2.  Rollback

   In certain situations it may be desirable to "undo" a call being dispatched change made to multiple
   remote callees,
   setLocalDescription or setRemoteDescription.  Consider a case where each callee can accept the call, but only one
   active session ever exists at a time; no mixing of received media
   call 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 ongoing, and one side wants to change some of the callees, session
   parameters; that side generates an updated offer and then calls
   setLocalDescription.  However, the application can
   choose to apply it remote side, either as a provisional answer, leaving open the
   possibility of using a different answer in the future, before or apply
   after setRemoteDescription, decides it as
   a final answer, ending does not want to accept the setup flow.

   In
   new parameters, and sends a "first-one-wins" situation, reject message back to the first answer will be applied offerer.  Now,
   the offerer, and possibly the answerer as well, need to return to a final answer,
   stable state 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, 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 previous call leg media that is received will be terminated.
   At some point, the application will end processed according to
   the setup process, perhaps
   with previous session description.

   A rollback is performed by supplying a timer; at this point, the application could reapply the
   existing remote session description as a final answer.

3.5.2.  Parallel Forking

   Parallel forking involves a call being dispatched of type
   "rollback" to multiple remote
   callees, where each callee can accept either setLocalDescription or setRemoteDescription,
   depending on which needs to be rolled back (i.e. if the call, and multiple
   simultaneous active signaling sessions can new offer was
   supplied to setLocalDescription, the rollback should be established done on
   setLocalDescription as a
   result.  If multiple callees send media at the same time, well.)

4.1.4.  setLocalDescription

   The setLocalDescription method instructs 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 PeerConnection 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 apply
   the North American ringback tone - supplied SDP blob as its local configuration.  The type field
   indicates whether the resulting sound would not blob should be
   like either tone, processed as an offer,
   provisional answer, or final answer; offers and would confuse answers are checked
   differently, using the user.  If 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 signaling application wishes wants to only exchange media with offer to
   change from one media format to a different, incompatible format, the
   PeerConnection must be able to simultaneously support use of both the remote
   endpoints at a time, then from
   old and new local descriptions (e.g. support codecs that exist in
   both descriptions) until a media engine point of view, this final answer is
   exactly like the sequential forking case.

   In received, at which point
   the parallel forking case where PeerConnection can fully adopt the Javascript application wishes new local description, or roll
   back to simultaneously exchange media with multiple peers, the flow is
   slightly more complex, but old description if the Javascript application can follow remote side denied the
   strategy that [RFC3960] describes using UPDATE.  (It change.

   This API indirectly controls the candidate gathering process.  When a
   local description is worth noting
   that supplied, and the number of transports currently
   in use cases where this 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 desired behavior media directions are very unusual.)
   The UPDATE approach allows the signaling to set up a separate compatible, and media
   flow for each peer that it wishes are available to exchange media with.  In JSEP,
   send, this offer used will result in the UPDATE would be formed by simply creating a
   new PeerConnection and making sure that the same local starting of media streams
   have been added into this new PeerConnection.  Then transmission.

4.1.5.  setRemoteDescription

   The setRemoteDescription method instructs the new PeerConnection object would produce a SDP offer that could be used by
   the signaling to perform apply
   the UPDATE strategy discussed in [RFC3960] . supplied SDP blob as the desired remote configuration.  As a result in
   setLocalDescription, the type field of sharing the media streams, indicates how the application will end blob
   should be processed.

   This API changes the local media state; among other things, it sets
   up
   with N parallel PeerConnection sessions, each with a local resources for sending and remote
   description and their own local encoding media.

   If setRemoteDescription was previously called with an offer, and remote addresses.  The media flow
   from these sessions can be managed by specifying SDP direction
   attributes in the descriptions,
   setLocalDescription is called with an answer (provisional or final),
   and the application can choose media directions are compatible, and media are available to play
   out
   send, this will result in the starting of media from all sessions mixed together.  Of course, if the
   application wants to only keep transmission.

4.1.6.  localDescription

   The localDescription method returns a single session, it can simply
   terminate the sessions that it no longer needs.

3.6.  Session Rehydration

   In the event that copy of the current local application state is reinitialized,
   either due
   configuration, i.e. what was most recently passed to a user reload of the page, or a decision within
   setLocalDescription, plus any local candidates that have been
   generated by the
   application to reload itself (perhaps to update to a new version), it
   is possible ICE Agent.

   TODO: Do we need to keep an existing session alive, via a process called
   "rehydration". 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
   The explicit goal remoteDescription method returns a copy of rehydration is to carry out this
   session resumption with no interaction with the current remote side other
   than normal call signaling messages.

   With rehydration,
   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 signaling state is persisted somewhere
   outside of the page, perhaps on and
   proposed remote description?

   A null object will be returned if the application server, remote description has not yet
   been established, or in browser
   local storage.  The page is then reloaded, if the saved signaling state
   is retrieved, and a new PeerConnection object is created for the
   session. has been closed.

4.1.8.  updateIce

   The previously obtained MediaStreams are re-acquired, and
   are given the same IDs as updateIce method allows the original session; this ensures configuration of the IDs
   in use by ICE Agent to be
   changed during the remote side continue session, primarily for changing which types of
   local candidates are provided to work.  Next, a new offer is
   generated by the new PeerConnection; this offer will have new ICE application and
   possibly new SDES credentials (since used for
   connectivity checks.  A callee may initially configure the old ICE and SRTP state has
   been lost).  Finally, Agent
   to use only relay candidates, to avoid leaking location information,
   but update this offer 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 re-initiate the session
   with the existing remote endpoint, who simply sees state of the new offer as
   an in-call renegotiation, and replies with an answer that can be
   supplied to setRemoteDescription. ICE processing proceeds as usual, Agent, and as soon as
   may result in a change to media state if it results in connectivity is established,
   being established.

4.1.9.  addIceCandidate

   The addIceCandidate method provides a remote candidate to the session ICE
   Agent, which, if parsed successfully, will be back
   up and running again.

   [OPEN ISSUE:  EKR proposed an alternative rehydration approach where
   the actual internal PeerConnection object in added to the browser was kept
   alive for some time after remote
   description according to the web page was killed and provided some
   way rules defined for a new page Trickle ICE.
   Connectivity checks will be sent to acquire the old PeerConnection object.]

4.  Interface new candidate.

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

4.1.  SDP Requirements

   Note:  The text in this section may not represent working group
   consensus and is put here so that the working group can discuss it state of the ICE Agent, and find out how to
   may result in a change to media state if it such that it does have consensus.

   When generating SDP blobs, either for offers or answers, the
   generated results in connectivity
   being established.

5.  SDP needs to conform to Interaction Procedures

   This section describes the following specifications.
   Similarly, in order specific procedures to properly process received be followed when
   creating and parsing SDP blobs,
   implementations need to implement the functionality described in the
   following specifications. objects.

5.1.  SDP Requirements Overview
   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.  [OPEN
         ISSUE; depends on BUNDLE and how we choose to represent
         multiple media sources]

   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] MAY MUST NOT be implemented to signal SDES SRTP keying
      information.

   R-13  A TBD-draft [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.  Though there is not yet WG consensus standard way.

   R-14  The bundle mechanism in
      [I-D.ietf-mmusic-sdp-bundle-negotiation] MUST be supported to
      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 area, this TBD-draft is very likely field.  As mentioned in [RFC4566], the
      entire o= line needs to be
         [I-D.alvestrand-mmusic-msid].
   R-14  A TBD-draft unique, but selecting a random number
      for <sess-id> is sufficient to accomplish this.

   o  The third SDP line MUST be supported to signal the use or multiplexing
         RTP somethings on a single UDP port, "s=" line, as specified in order to avoid
         excessive use of port number resources.  Though there is [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
      Time Zones ("z=") lines are not
         yet WG consensus useful in this area, this TBD-draft is very likely to context and SHOULD
      NOT be [I-D.holmberg-mmusic-sdp-bundle-negotiation].

   As required by [RFC4566] Section 5.13 JSEP implementations included.

   o  Encryption Keys ("k=") lines do not provide sufficient security
      and MUST
   ignore unknown attributes (a=) lines.

   Example SDP for RTCWeb call flows can NOT be found included.

   o  A "t=" line MUST be added, as specified in
   [I-D.nandakumar-rtcweb-sdp].  [TODO:  since we are starting [RFC4566], Section 5.9;
      both <start-time> and <stop-time> SHOULD be set to
   specify how zero, e.g. "t=0
      0".

   The next step is to handle SDP in generate m= sections for each MediaStreamTrack
   that has been added to the PeerConnection via the addStream method.
   Note that this document, 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 these call flows be merged into this document, or this link moved to the examples
   section?]

4.2.  Methods

4.2.1.  createOffer generated as specified in [RFC4566],
   Section 5.14.  The createOffer method generates <proto> field MUST be set to "RTP/SAVPF".  If a blob of SDP that contains m=
   section is not being bundled into another m= section, it MUST
   generate a
   [RFC3264] offer with the supported configurations for the session,
   including descriptions unique set of ICE credentials and gather its own set of
   candidates.  Otherwise, it MUST use the local MediaStreams attached to this
   PeerConnection, the codec/RTP/RTCP options supported by this
   implementation, same ICE credentials and any
   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 been gathered by the ICE
   Agent.  A constraints parameters may be supplied to provide
   additional control over same
   fingerprint values.

   Each m= section MUST include the generated offer.  This constraints
   parameter should allow for 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 following manipulations to codecs
      specified in [I-D.ietf-rtcweb-audio], Section 3, MUST be
   performed: be
      supported.

   o  To indicate support for  For each primary codec where RTP retransmission should be used, a media type even if no MediaStreamTracks
      corresponding "a=rtpmap" line indicating "rtx" with the clock rate
      of that type have been added to the session (e.g., primary codec and an audio call "a=fmtp" line that wants to receive video.)
   o  To trigger an ICE restart, for references the purpose of reestablishing
      connectivity.
      payload type fo the primary codec, as specified in [RFC4588],
      Section 8.1.

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

   In the initial offer, the generated SDP will contain all desired
   functionality for FEC mechanism, a corresponding "a=rtpmap" line
      indicating the session (certain parts that are supported but
   not desired by default may be omitted); for each SDP 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
   generation of the SDP must follow the process defined for generating
   an initial offer from the document (listed "trickle" option, as specified
      in [I-D.ivov-mmusic-trickle-ice], Section 4.1) 4.

   o  For each candidate that
   specifies the given SDP line.

   In the event createOffer is called after has been gathered during the session is established,
   createOffer will generate most recent
      gathering phase, an offer to modify "a=candidate" line, as specified in [RFC5245],
      Section 4.3., paragraph 3.

   o  For the current session
   based on any changes that 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 made to gathered yet,
      the session, e.g. adding
   or removing MediaStreams, or requesting an ICE restart.  For each
   existing stream, default candidate should be set to the generation 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 each SDP line must follow the
   process defined SHA-256 algorithm for generating an updated offer from the document
   that specfies fingerprint is REQUIRED; if
      the given SDP line. 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 new stream, the
   generation 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 SDP must follow SSRC to be used for sending media.

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

   o  If no changes have been made, or FEC is supported for SDP lines that are unaffected by the requested changes, the offer
   will only contain this media type, another "a=ssrc" line
      with the parameters negotiated by FEC SSRC, and an "a=ssrc-group" line, as specified in
      [RFC5576], section 4.2, with semantics set to "FEC" and including
      the last offer-answer
   exchange.

   Session descriptions generated by createOffer must be immediately
   usable by setLocalDescription; primary and FEC SSRCs.

   o  [OPEN ISSUE: Handling of a=imageattr]

   o  [TODO: bundle-only]

   Lastly, if a system data channel has limited resources
   (e.g. been created, 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 m= section MUST be
   generated for data.  The <media> field MUST be set to inspect "application"
   and the system state <proto> field MUST be set to determine the currently available resources, it may "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 implemented included as an async operation.

   Calling this method may do things such mentioned above.  [OPEN ISSUE:
   additional SCTP-specific stuff to be included, as generate new ICE
   credentials, but does not result indicated in candidate gathering, or cause
   media to start or stop flowing.

4.2.2.  createAnswer

   The createAnswer method generates
   [I-D.jesup-rtcweb-data-protocol] (currently none)]

   Once all m= sections have been generated, a blob of SDP 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 contains a
   [RFC3264] SDP answer with are common between all m= sections MAY be moved to
   session-level, if desired.

   Attributes other than the supported configuration ones specified above MAY be included,
   except for the session 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 compatible with used, any additional attributes that are
   added MUST follow the parameters supplied advice in
   [I-D.nandakumar-mmusic-sdp-mux-attributes] on how those attributes
   interact with BUNDLE.

5.2.2.  Subsequent Offers

   When createOffer is called a second (or later) time, the offer.  Like
   createOffer, processing
   is different, depending on the returned blob contains descriptions of current signaling state.

   If the local
   MediaStreams attached to this PeerConnection, initial offer was not applied using setLocalDescription,
   meaning the codec/RTP/RTCP
   options negotiated PeerConnection is still in the "stable" state, the steps
   for generating an initial offer should be followed, with this session, and any candidates that have
   been gathered by
   exception:

   o  The "o=" line MUST stay the ICE Agent.  A constraints parameter may be
   supplied to provide additional control over same.

   If the generated answer.

   As initial offer was applied using setLocalDescription, but an answer, the generated SDP will contain a specific configuration
   that specifies how
   answer from the media plane should be established; for each
   SDP line, remote side has not yet been applied, meaning the generation of
   PeerConnection is still in the SDP must follow "local-offer" state, the process defined steps for
   generating an answer from initial offer should be followed, with these
   exceptions:

   o  The "o=" line MUST stay the document that specifies same, except for the given
   SDP line.

   Session descriptions generated by createAnswer must be immediately
   usable <session-version>
      field, which MUST increase by setLocalDescription; like createOffer, 1 from the returned
   description should reflect previously applied local
      description.

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

   o  Each "a=mid" line MUST stay the system.  Because
   this method may need to inspect same.

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

   o  For MediaStreamTracks that are still present, the system state to determine "a=msid",
      "a=ssrc", and "a=ssrc-group" lines MUST stay the
   currently available resources, it may need to be implemented as an
   async operation.

   Calling this same.

   o  If any MediaStreamTracks have been removed, either through the
      removeStream method may do things such as generate new ICE
   credentials, but does not trigger candidate gathering or change media
   state.

4.2.3.  SessionDescriptionType

   Session description objects (RTCSessionDescription) may by removing them from an added MediaStream,
      their m= sections MUST be of type
   "offer", "pranswer", and "answer".  These types provide information marked as to how recvonly by changing the description parameter should be parsed, and how value
      of the
   media state should be changed.

   "offer" indicates that a description should [RFC3264] directional attribute to "a=recvonly".  The
      "a=msid", "a=ssrc", and "a=ssrc-group" lines MUST be parsed as an offer;
   said description may include many possible media configurations.  A
   description used as removed from
      the associated m= sections.

   If the initial offer was applied using setLocalDescription, and an "offer" may be
   answer from the remote side has been applied anytime using
   setRemoteDescription, meaning the PeerConnection is in a stable state, the "remote-
   pranswer" or as "stable" states, an update to a previously
   supplied but unanswered "offer".

   "pranswer" indicates that a description 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 parsed as an
   answer, but not permitted in the remote-pranswer state?]

   o  If a final answer, m= section was rejected, i.e. has had its port set to zero in
      either the local or remote description, it MUST remain rejected
      and so should not result have a zero port in the new offer, as indicated in RFC3264,
      Section 5.1.

   o  If a m= section exists in the
   freeing 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 allocated resources.  It may result in creating a new m= section.  [OPEN ISSUE: Nail down
      exactly what this means.  Should the start of media
   transmission, if codecs remain the answer 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 specify have an associated MediaStreamTrack (i.e. it is 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
      recvonly), a description should corresponding m= section MUST be parsed as an answer, generated in 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", new
      offer, but without "a=msid", "a=ssrc", or an update to a previously sent "pranswer".

   The only difference between a provisional "a=ssrc-group"
      attributes, and final answer is that the final answer results appropriate directional attribute must be
      specified.

   In addition, for each previously existing, non-rejected m= section in
   the freeing of any unused resources that
   were allocated as a result new offer, the following adjustments are made based on the
   contents of the offer.  As such, corresponding m= section in the application
   can use some discretion on whether an answer should be applied as
   provisional or final, current remote
   description:

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

   o  The RTP header extensions MUST only include those that are present
      in the session
   description as needed.  For example, remote description.

   o  The RTCP feedback extensions MUST only include those that are
      present in a serial forking scenario, an
   application may receive multiple "final" answers, one from each the remote endpoint. description.

   o  The application could choose to accept "a=rtcp-mux" line MUST only be added if present in the initial
   answers as provisional answers, and remote
      description.

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

5.2.3.  Constraints Handling

   The createOffer method takes as final a parameter a MediaConstraints
   object.  Special processing is performed when it receives one that meets its criteria (e.g. generating a live user
   instead of voicemail).

4.2.3.1.  Use of Provisional Answers

   Most web applications will not need to create answers using SDP
   description if 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 following constraints are present.

5.2.3.1.  OfferToReceiveAudio

   If the offer, instead "OfferToReceiveAudio" constraint is specified, with a value of waiting for
   "true", the offer MUST include a human user non-rejected m= section with media
   type "audio", even if no audio MediaStreamTrack has been added to
   physically answer the call.  Later,
   PeerConnection.  This allows the offerer to receive audio even when
   not sending it; accordingly, the human input is received, directional attribute on the application can create a new "sendrecv" offer audio
   m= section MUST be set to recvonly.  If this constraint is specified
   when an audio MediaStreamTrack has already been added to update the
   previous offer/answer pair and start the
   PeerConnection, or a non-rejected m= section with media flow.  This approach
   is preferred because type "audio"
   previously existed, it minimizes has no effect.

5.2.3.2.  OfferToReceiveVideo

   If the amount "OfferToReceiveAudio" constraint is specified, with a value of time that
   "true", the offer-
   answer exchange is left open, in addition offer MUST include a m= section with media type "video",
   even if no video MediaStreamTrack has been added to avoiding media clipping
   by ensuring the transport is ready
   PeerConnection.  This allows the offerer to go by receive video even when
   not sending it; accordingly, the time directional attribute on the call is
   phyiscally answered.  However, some applications may not video
   m= section MUST be able to
   do this, particularly ones that are attempting to gateway set to other
   signaling protocols.  In these cases, "pranswer" can still allow the
   application recvonly.  If this constraint is specified
   when an video MediaStreamTrack has already been added to warm up the transport.

   Consider
   PeerConnection, or a typical web application that will set up 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 data channel,
   an 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 channel, and codecs?  Opus has built-in DTX, but G.711 does not.]

5.2.3.4.  IceRestart

   If the "IceRestart" constraint is specified, with a video channel.  When value of "true",
   the offer MUST indicate an endpoint receives 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
   offer with these channels, initial offer, it could send an answer accepting the data
   channel for two-way data, has no effect
   (since a new ICE ufrag and accepting 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 audio and video tracks supplied remote description as well as
   inactive or receive-only.  It could then ask the user to accept
   the
   call, acquire 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 local media streams, and send first time after a new offer to the remote side moving the audio and video to be two-way media.  By
   description has been provided, the
   time result is known as the human initial
   answer.  If no remote description has accepted the call been installed, an answer
   cannot be generated, and sent the new offer, it is
   likely an error MUST be returned.

   Note that the ICE remote description SDP may not have been created by a
   WebRTC endpoint and DTLS handshaking for all the channels will
   already be set up.

4.2.3.2.  Rollback

   In certain situations it may be desirable not conform to "undo" all the requirements listed in
   Section 5.2.  For many cases, this is not a change made 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
   setLocalDescription fail, or setRemoteDescription.  Consider a case where a
   call should this cause
   createAnswer to reject those particular m= sections?]

   The first step in generating an initial answer is ongoing, and one side wants to change some of generate
   session-level attributes.  The process here is identical to that
   indicated in the session
   parameters; Initial Offers section above, with the addition that side generates an updated offer and then calls
   setLocalDescription.  However,

   The next step is to generate m= sections for each m= section that is
   present in the remote side, either before or
   after setRemoteDescription, decides it does not want to accept offer, as specified in [RFC3264], Section 6.
   For the
   new parameters, and sends a reject message back 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 offerer.  Now, offered m= sections have been rejected, by stopping the offerer, and possibly
   associated remote MediaStreamTrack, the answerer as well, need to return to a
   stable state and corresponding m= section in
   the previous local/remote description.  To support
   this, we introduce answer MUST be marked as rejected by setting the concept of "rollback".

   A rollback returns port in the state machine m=
   line to its previous state, zero, as indicated in [RFC3264], Section 6., and processing
   continues with the
   local or remote description to its previous value.  Any resources or
   candidates that were allocated by the new local description are
   discarded; any next m= section.

   For each non-rejected m= section of a given media that type, if there is received will be processed according a
   local MediaStreamTrack of the specified type which has been added to
   the previous session description.

   A rollback PeerConnection via addStream and not yet associated with a m=
   section, the MediaStreamTrack is performed by supplying associated with the m= section at
   this time.  If there are more m= sections of a session description certain type than
   MediaStreamTracks, some m= sections will not have an associated
   MediaStreamTrack.  If there are more MediaStreamTracks of a certain
   type
   "rollback" than m= sections, only the first N MediaStreamTracks will be
   able to either setLocalDescription or setRemoteDescription,
   depending on which needs be associated in the constructed answer.  The remainder will
   need to be rolled back (i.e. if 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 new
   offer was
   supplied supports BUNDLE, all m= sections to setLocalDescription, 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:

   o  If present in the rollback should be done on
   setLocalDescription offer, an "a=mid" line, as well.)

4.2.4.  setLocalDescription specified in
      [RFC5888], Section 9.1.  The setLocalDescription method instructs the PeerConnection to apply "mid" value MUST match that specified
      in the supplied SDP blob offer.

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

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

   o  If a local configuration.  The type field
   indicates whether the blob should be processed 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,
   provisional answer, or final answer; offers "a=rtpmap"
      and answers are checked
   differently, using "a=fmtp" lines, as specified in [RFC4566], Section 6, and
      [RFC3264], Section 6.1.  For audio, the various rules codecs specified in
      [I-D.ietf-rtcweb-audio], Section 3, MUST be be supported.  Note
      that exist for each SDP line.

   This API changes simplicity, the local media state; among other things, answerer MAY use different payload types
      for codecs than the offerer, as it sets
   up local resources is not prohibited by
      Section 6.1.

   o  If "rtx" is present in the offer, for receiving and decoding media.  In order to
   successfully handle scenarios each primary codec where the application wants to offer to
   change from one media format to RTP
      retransmission should be used, a different, incompatible format, corresponding "a=rtpmap" line
      indicating "rtx" with the
   PeerConnection must be able to simultaneously support use clock rate of both the
   old primary codec and new local descriptions (e.g. support codecs an
      "a=fmtp" line that exist references the payload type fo the primary
      codec, as specified in
   both descriptions) until a final answer [RFC4588], Section 8.1.

   o  For each supported FEC mechanism that is received, at which point
   the PeerConnection can fully adopt present in the new local description, or roll
   back to offer, a
      corresponding "a=rtpmap" line indicating the old description if desired FEC codec.

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

   o  If the remote side denied "trickle" ICE option is present in the change.

   This API indirectly controls 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 process.  When a
   local description is supplied, and phase, an "a=candidate" line, as specified in [RFC5245],
      Section 4.3., paragraph 3.

   o  For the number of transports currently current default candidate, a "c=" line, as specific in use does not match
      [RFC5245], Section 4.3., paragraph 6.  [OPEN ISSUE, pending
      resolution in mmusic: If no candidates have yet been gathered yet,
      the number 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 transports needed by the local
   description, SHA-256 algorithm for the PeerConnection will create transports 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 needed specified in [RFC4145], Section 4, and
   begin gathering candidates
      clarified for them.

   If setRemoteDescription was previous called with an offer, and
   setLocalDescription is called with an use in DTLS-SRTP scenarios in [RFC5763], Section 5.
      The role value in the answer (provisional MUST be "active" or final),
   and "passive"; the media directions are compatible, and media are available to
   send, this will result
      "active" role is RECOMMENDED.

   o  If present in the starting of media transmission.

4.2.5.  setRemoteDescription

   The setRemoteDescription method instructs the PeerConnection to apply
   the supplied SDP blob offer, an "a=rtcp-mux" line, as specified in
      [RFC5761], Section 5.1.1.

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

   o  For each supported RTP header extension that is present in
   setLocalDescription, the type field
      offer, an "a=extmap" line, as specified in [RFC5285], Section 5.
      The list of the indicates how the blob
   should 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 processed.

   This API changes specified as indicated
      in [RFC6904], Section 4.

   o  For each supported RTCP feedback mechanism that is present in 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 "a=rtcp-fb" mechanism, as specified in the starting of media transmission.

4.2.6.  localDescription [RFC4585],
      Section 4.2.  The localDescription method returns a copy list of the current local
   configuration, i.e. what was most recently passed to
   setLocalDescription, plus any local candidates RTCP feedback mechanisms 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 SHOULD/
      MUST be returned if the supported is specified in [I-D.ietf-rtcweb-rtp-usage],
      Section 5.1.

   o  If a local description has not yet
   been established, or if the PeerConnection MediaStreamTrack has been closed.

4.2.7.  remoteDescription

   The remoteDescription method returns a copy of associated, an "a=ssrc" line,
      as specified in [RFC5576], Section 4.1, indicating 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 SSRC to expose accessors for both the current and
   proposed remote description?

   A null object will be returned if the remote description
      used for sending media.

   o  If a local MediaStreamTrack has not yet been established, or if the PeerConnection associated, and RTX has been closed.

4.2.8.  updateIce

   The updateIce method allows the configuration of
      negotiated for this m= section, another "a=ssrc" line with the ICE Agent RTX
      SSRC, and an "a=ssrc-group" line, as specified in [RFC5576],
      section 4.2, with semantics set to be
   changed during "FID" and including the session, primarily for changing which types of primary
      and RTX SSRCs.

   o  If a local candidates are provided to the application MediaStreamTrack has been associated, and used FEC has been
      negotiated 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, m= section, another "a=ssrc" line with 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 FEC
      SSRC, and an "a=ssrc-group" line, as specified in a change [RFC5576],
      section 4.2, with semantics set to "FEC" and including the state of the ICE Agent, primary
      and
   may result in FEC SSRCs.

   o  [OPEN ISSUE: Handling of a=imageattr]

   o  [TODO: bundle-only]
   If a change to media state if it results in connectivity
   being established.

4.2.9.  addIceCandidate

   The addIceCandidate method provides data channel m= section has been offered, a remote candidate to the ICE
   Agent, which, if parsed successfully, will m= section MUST also
   be added to the remote
   description according to the rules defined generated for Trickle ICE.
   Connectivity checks will data.  The <media> field MUST be sent set to
   "application" and the new candidate.

   This call will result in a change <proto> field MUST be set to the state of the ICE Agent, "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
   may result "a=setup" lines MUST be included as mentioned
   above.  [OPEN ISSUE: additional SCTP-specific stuff to be included,
   as indicated in a change [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 media state
   session-level, if it results desired.

   The attributes prohibited in creation of offers are also prohibited
   in connectivity
   being established.

5.  SDP Interaction Procedures

   This section describes the specific procedures to be followed when
   creating and parsing SDP objects.  [Work In Progress]

5.1.  Constructing an Offer

5.2.  Generating an Answer

5.3. creation of answers.

5.3.2.  Subsequent Answers

5.3.3.  Constraints Handling

5.4.  Parsing an Offer

5.4.

5.5.  Parsing an Answer

5.5.

5.6.  Applying a Local Description

5.6.

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).

   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 framerate frame rate

   o  change desired recv resolution or framerate 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] [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.

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.rescorla-mmusic-ice-trickle]
              Rescorla, E., Uberti, J.,

   [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 E. Ivov, "Trickle ICE:
              Incremental Provisioning 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.

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

   [I-D.nandakumar-mmusic-sdp-mux-attributes]
              Nandakumar, S., "A Framework for the Interactive
              Connectivity Establishment (ICE) Protocol",
              draft-rescorla-mmusic-ice-trickle-00 SDP Attributes when
              Multiplexing", draft-nandakumar-mmusic-sdp-mux-
              attributes-03 (work in progress),
              October 2012. 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.

   [RFC3388]  Camarillo, G., Eriksson, G., Holler, J.,

   [RFC4145]  Yon, D. and H.
              Schulzrinne, "Grouping of G. Camarillo, "TCP-Based Media Lines Transport in
              the Session Description Protocol (SDP)", RFC 3388, December 2002.

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

   [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.

10.2.  Informative References

   [I-D.alvestrand-mmusic-msid]
              Alvestrand, H., "Cross Session Stream Identification in
              the Session Description Protocol",
              draft-alvestrand-mmusic-msid-01 (work in progress),
              October 2012.

   [I-D.holmberg-mmusic-sdp-bundle-negotiation]
              Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
              Using Session Description Protocol (SDP) Port Numbers",
              draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work Grouping Framework", RFC 5888, June 2010.

   [RFC6904]  Lennox, J., "Encryption of Header Extensions in
              progress), October 2011.

   [I-D.ietf-rtcweb-rtp-usage]
              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
              Communication (WebRTC): Media 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 Use J. Uberti, "Trickle ICE:
              Incremental Provisioning of RTP",
              draft-ietf-rtcweb-rtp-usage-04 Candidates for the Interactive
              Connectivity Establishment (ICE) Protocol", draft-ivov-
              mmusic-trickle-ice-01 (work in progress),
              July 2012. March 2013.

   [I-D.jennings-rtcweb-signaling]
              Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/
              Answer Protocol (ROAP)",
              draft-jennings-rtcweb-signaling-01 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-00
              draft-nandakumar-rtcweb-sdp-02 (work in progress),
              October 2012. 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.

   [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)", (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, 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);
   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);
   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"/>
   // 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);

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();
   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
   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
   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
   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
   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: 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 specificaiton specification

   o  Align SDP handling with W3C draft

   o  Clarified section on forking.

   Changes in draft -01: 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
   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