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Versions: (draft-uberti-rtcweb-jsep) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25

Network Working Group                                          J. Uberti
Internet-Draft                                                    Google
Intended status:  Standards Track                            C. Jennings
Expires:  April 25, 2013                                           Cisco
                                                        October 22, 2012


               Javascript Session Establishment Protocol
                       draft-ietf-rtcweb-jsep-02

Abstract

   This document proposes a mechanism for allowing a Javascript
   application to fully control the signaling plane of a multimedia
   session, and discusses how this would work with existing signaling
   protocols.

Status of this Memo

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

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

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

   This Internet-Draft will expire on April 25, 2013.

Copyright Notice

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

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.



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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Other Approaches Considered  . . . . . . . . . . . . . . . . .  5
   3.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  6
   4.  Semantics and Syntax . . . . . . . . . . . . . . . . . . . . .  7
     4.1.  Signaling Model  . . . . . . . . . . . . . . . . . . . . .  7
     4.2.  Session Descriptions and State Machine . . . . . . . . . .  7
     4.3.  Session Description Format . . . . . . . . . . . . . . . .  9
     4.4.  ICE  . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
       4.4.1.  ICE Candidate Trickling  . . . . . . . . . . . . . . . 10
         4.4.1.1.  ICE Candidate Format . . . . . . . . . . . . . . . 10
     4.5.  Interactions With Forking  . . . . . . . . . . . . . . . . 11
       4.5.1.  Sequential Forking . . . . . . . . . . . . . . . . . . 11
       4.5.2.  Parallel Forking . . . . . . . . . . . . . . . . . . . 12
     4.6.  Session Rehydration  . . . . . . . . . . . . . . . . . . . 13
   5.  Interface  . . . . . . . . . . . . . . . . . . . . . . . . . . 14
     5.1.  SDP Requirements . . . . . . . . . . . . . . . . . . . . . 14
     5.2.  Methods  . . . . . . . . . . . . . . . . . . . . . . . . . 15
       5.2.1.  createOffer  . . . . . . . . . . . . . . . . . . . . . 15
       5.2.2.  createAnswer . . . . . . . . . . . . . . . . . . . . . 15
       5.2.3.  SessionDescriptionType . . . . . . . . . . . . . . . . 16
         5.2.3.1.  Creating Answers . . . . . . . . . . . . . . . . . 17
       5.2.4.  setLocalDescription  . . . . . . . . . . . . . . . . . 17
       5.2.5.  setRemoteDescription . . . . . . . . . . . . . . . . . 18
       5.2.6.  localDescription . . . . . . . . . . . . . . . . . . . 18
       5.2.7.  remoteDescription  . . . . . . . . . . . . . . . . . . 18
       5.2.8.  updateIce  . . . . . . . . . . . . . . . . . . . . . . 18
       5.2.9.  addIceCandidate  . . . . . . . . . . . . . . . . . . . 19
   6.  Configurable SDP Parameters  . . . . . . . . . . . . . . . . . 20
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 21
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 22
   9.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23
   10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 24
     10.1. Normative References . . . . . . . . . . . . . . . . . . . 24
     10.2. Informative References . . . . . . . . . . . . . . . . . . 24
   Appendix A.  JSEP Implementation Examples  . . . . . . . . . . . . 26
     A.1.  Example API Flows  . . . . . . . . . . . . . . . . . . . . 26
       A.1.1.  Call using ROAP  . . . . . . . . . . . . . . . . . . . 26
       A.1.2.  Call using XMPP  . . . . . . . . . . . . . . . . . . . 27
       A.1.3.  Adding video to a call, using XMPP . . . . . . . . . . 28
       A.1.4.  Simultaneous add of video streams, using XMPP  . . . . 28
       A.1.5.  Call using SIP . . . . . . . . . . . . . . . . . . . . 29
       A.1.6.  Handling early media (e.g. 1-800-GO FEDEX), using
               SIP  . . . . . . . . . . . . . . . . . . . . . . . . . 30
   Appendix B.  Change log  . . . . . . . . . . . . . . . . . . . . . 32
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 33




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1.  Introduction

   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 cause
   problems for an embedded signaling state machine.  One of these is
   that the user may reload the web page at any time.  If this happens,
   and the state machine is being run at a server, the server can simply
   push the current state back down to the page and resume the call
   where it left off.

   This document describes the Javascript Session Establishment Protocol
   (JSEP) that pulls the signaling state machine out of the browser and
   into 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.

   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, 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



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   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.

   The JSEP approach does come with a minor downside.  As the
   application now is responsible for driving the signaling state
   machine, slightly more application code is necessary to perform call
   setup; the application must call the right APIs at the right times,
   and convert the session descriptions and ICE information into the
   defined messages of its chosen signaling protocol, instead of simply
   forwarding the messages emitted from the browser.

   One way to mitigate this is to provide a Javascript library that
   hides this complexity from the developer, which would implement the
   state machine and serialization of the desired signaling protocol.
   For example, this library could convert easily adapt the JSEP API
   into the exact ROAP API [I-D.jennings-rtcweb-signaling], thereby
   implementing the ROAP signaling protocol.  Such a library could of
   course also implement other popular signaling protocols, including
   SIP or Jingle.  In this fashion we can enable greater control for the
   experienced developer without forcing any additional complexity on
   the novice developer.
























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2.  Other Approaches Considered

   Another approach that was considered for JSEP was to move the
   mechanism for generating offers and answers out of the browser as
   well.  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.  Exposing createOffer/
   createAnswer avoids that problem, but still allows applications to
   generate their own offers/answers if they choose, using the
   description generated by createOffer as an indication of the
   browser's capabilities.

   Note also that while JSEP transfers more control to Javascript, it is
   not intended to be an example of a "low-level" API.  The general
   argument against a low-level API is that there are too many necessary
   API points, and they can be called in any order, leading to something
   that is hard to specify and test.  In the approach proposed here,
   control is performed via session descriptions; this requires only a
   few APIs to handle these descriptions, and they are evaluated in a
   specific fashion, which reduces the number of possible states and
   interactions.























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3.  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 RFC 2119 [RFC2119].














































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4.  Semantics and Syntax

4.1.  Signaling Model

   JSEP does not specify a particular signaling model or state machine,
   other than the generic need to exchange RFC 3264 offers and answers
   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 actual mechanism by
   which these offers and answers are communicated to the remote side,
   including addressing, retransmission, forking, and glare handling, is
   left entirely up to the application.

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

                      Figure 1: JSEP Signaling Model

4.2.  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 was sent or received 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 decode,
   and what the remote party should send.  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 how the remote party should expect to receive.

   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



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   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 offerer before the answer arrives.

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

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

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

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

   JSEP addresses this by adding both a setLocalDescription and a
   setRemoteDescription method and having session description objects
   contain a type field indicating the type of session description being
   supplied.  This satisfies the requirements listed above for both the
   offerer, who first calls setLocalDescription(sdp [offer]) and then
   later setRemoteDescription(sdp [answer]), as well as for the
   answerer, who first calls setRemoteDescription(sdp [offer]) and then
   later setLocalDescription(sdp [answer]).  While it could be possible
   to implicitly determine the value of the offer/answer argument,
   requiring it to be specified explicitly is more robust, allowing
   invalid combinations (i.e. an answer before an offer) to generate an
   appropriate error.

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

   In [RFC3264], the constraints at the signaling level is that only one
   offer can be outstanding for a given session but from the media stack
   level, a new offer can be generated at any point.  For example, when
   using SIP for signaling, if one offer is sent, then cancelled using a
   SIP CANCEL, another offer can be generated even though no answer was
   received for the first offer.  To support this, the JSEP media layer



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   can provide an offer whenever the Javascript application needs one
   for the signaling.  The answerer can send back zero or more
   provisional answers, and finally end the offer-answer exchange by
   sending a final answer.  The state machine for this is as follows:

         +-----------+
         |           |
         |           |
         |  Stable   |<---------------\
         |           |                |
         |           |                |
         +-----------+                |
             ^   |                    |
             |   | OFFER              |
      ANSWER |   |                    | ANSWER
             |   V                    |
         +-----------+          +-----------+
         |           |          |           |
         |           | PRANSWER |           |
         |   Offer   |-------- >| Pranswer  |
         |           |          |           |
         |           |----\     |           |----\
         +-----------+    |     +-----------+    |
                    ^     |                ^     |
                    |     |                |     |
                    \-----/                \-----/
                     OFFER                 PRANSWER

                       Figure 2: JSEP State Machine

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

4.3.  Session Description Format

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

   However, to simplify Javascript processing, and provide for future
   flexibility, the SDP syntax is encapsulated within a
   SessionDescription object, which can be constructed from SDP, and be



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   serialized out to SDP.  If future specifications agree on a JSON
   format for session descriptions, we could easily enable this object
   to generate and consume that JSON.

   Other methods may be added to SessionDescription in the future to
   simplify handling of SessionDescriptions from Javascript.  Though it
   is unclear exactly what manipulations developer will commonly want to
   do to SDP, it would be simple to write a Javascript library to
   perform these manipulations.

4.4.  ICE

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

4.4.1.  ICE Candidate Trickling

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

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

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

4.4.1.1.  ICE Candidate Format

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



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   The a=candidate lines are the only SDP information that is contained
   within IceCandidate, as they represent the only information needed
   that is not present in the initial offer (i.e. for trickle
   candidates).  This information is carried with the same syntax as the
   "a=candidate" line in SDP.  For example:

   a=candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host

   The IceCandidate object also contains fields to indicate which m=
   line it should be associated with.  The m line can be identified in
   one of two ways; either by a m-line index, or a MID.  The m-line
   index is a zero-based index, referring to the Nth m-line in the SDP.
   The MID uses the "media stream identification", as defined in [RFC
   3388], to identify the m-line.  WebRTC implementations creating an
   ICE Candidate object MUST populate both of these fields.
   Implementations receiving an ICE Candidate object SHOULD use the MID
   if they implement that functionality, or the m-line index, if not.

4.5.  Interactions With Forking

   Some call signaling systems allow various types of forking where an
   SDP Offer may be provided to more than one device.  For example, SIP
   RFC 3261 defines both a "Parallel Search" and "Sequential Search".
   Although these are primarily signaling level issues that are outside
   the scope of JSEP, they do have some impact on the configuration of
   the media plane, which is relevant.  When forking is happening at the
   signaling layer, the Javascript application responsible for the
   signaling needs to make the decisions about what media should be sent
   or received at any point of time and which remote endpoint it should
   communicate with.  JSEP is used to make sure the media engine can
   make the RTP and media perform as required by the application.  The
   basic operations that the applications can have the media engine do
   are:

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

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

4.5.1.  Sequential Forking

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

   JSEP handles serial forking well, allowing the application to easily



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   control the policy for selecting the desired remote endpoint.  When
   an answer arrives from one of the callees, the application can choose
   to apply it either as a provisional answer, leaving open the
   possibility of using a different answer in the future, or apply it as
   a final answer, ending the setup flow.

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

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

4.5.2.  Parallel Forking

   Parallel forking involves a call being dispatched to multiple remote
   callees, where each callee can accept the call, and multiple
   simultaneous active signaling sessions can be established as a
   result.  If multiple callees send media at the same time, the
   possibilities for handling this are described in Section 3.1 of RFC
   3960.  Most SIP devices today only support exchanging media with a
   single device at a time, and do not try to mix multiple early media
   audio sources, as that could result in a confusing situation.  For
   example. consider having a European ringback tone mixed together with
   the North American ringback tone - the resulting sound would not be
   like either tone, and would confuse the user.  If the signaling
   application wishes to only exchange media with one of the remote
   endpoints at a time, then from a media engine point of view, this is
   exactly like the sequential forking case.

   In the parallel forking case where the Javascript application wishes
   to simultaneously exchange media with multiple peers, the flow is
   slightly more complex, but the Javascript application can follow the
   strategy that RFC 3960 describes using UPDATE.  (It is worth noting
   that use cases where this is the desired behavior are very unusual.)
   The UPDATE approach allows the signaling to set up a separate media
   flow for each peer that it wishes to exchange media with.  In JSEP,
   this offer used in the UPDATE would be formed by simply creating a
   new PeerConnection and making sure that the same local media streams
   have been added into this new PeerConnection.  Then the new
   PeerConnection object would produce a SDP offer that could be used by
   the signaling to perform the UPDATE strategy discussed in RFC 3690.

   As a result of sharing the media streams, the application will end up
   with N parallel PeerConnection sessions, each with a local and remote



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   description and their own local and remote addresses.  The media flow
   from these sessions can be managed by specifying SDP direction
   attributes in the descriptions, or the application can choose to play
   out the media from all sessions mixed together.  Of course, if the
   application wants to only keep a single session, it can simply
   terminate the sessions that it no longer needs.

4.6.  Session Rehydration

   In the event that the local application state is reinitialized,
   either due to a user reload of the page, or a decision within the
   application to reload itself (perhaps to update to a new version), it
   is possible to keep an existing session alive via a process called
   "rehydration".

   With rehydration, the current signaling state is persisted somewhere
   outside of the page, perhaps on the application server, or in browser
   local storage.  The page is then reloaded, and a new session object
   is created in Javascript.  The saved signaling state is now
   retrieved, and a new PeerConnection object is created for the
   session.  At this point a new offer can be generated by the new
   PeerConnection, with new ICE and SDES credentials.  This can then be
   used to re-initiate the session with the existing remote endpoint,
   who simply sees the new offer as an in-call renegotiation, and will
   reply with an answer that can be supplied to setRemoteDescription.
   ICE processing proceeds as usual, and as soon as connectivity is
   established, the session will be back up and running again.

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



















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5.  Interface

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

5.1.  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
   and find out how to change it such that it does have consensus.

   When generating SDP blobs, either for offers or answers, the
   generated SDP needs to conform to the following specifications.
   Similarly, in order to properly process received SDP blobs,
   implementations need to implement the functionality described in the
   following specifications.  This list is derived from
   [I-D.ietf-rtcweb-rtp-usage].

      RFC4566 is the base SDP specification and MUST be implemented.

      RFC5124 MUST be supported for signaling RTP/SAVPF RTP profile.

      RFC5104 MUST be implemented to signal RTCP based feedback.

      RFC5761 MUST be implemented to signal multiplexing of RTP and
      RTCP.

      RFC5245 MUST be implemented for signaling the ICE candidate lines
      corresponding to each media stream.

      RFC3264 MUST be implemented to signal information about media
      direction.

      The RFC5888 grouping framework MUST be implemented for signaling
      the grouping information.

      RFC5506 MAY be implemented to signal Reduced-Size RTCP messages.

      RFC5576 MAY be implemented to signal RTP SSRC values.

      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.

   As required by RFC 4566 Section 5.13 JSEP implementations MUST ignore



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   unknown attributes (a=) lines.

   Example SDP for RTCWeb call flows can be found in
   [I-D.nandakumar-rtcweb-sdp].

5.2.  Methods

5.2.1.  createOffer

   The createOffer method generates a blob of SDP that contains a RFC
   3264 offer with the supported configurations for the session,
   including descriptions of the local MediaStreams attached to this
   PeerConnection, the codec/RTP/RTCP options supported by this
   implementation, and any candidates that have been gathered by the ICE
   Agent.  A constraints parameters may be supplied to provide
   additional control over the generated offer, e.g. to get a full set
   of session capabilities, or to request a new set of ICE credentials.

   In the initial offer, the generated SDP will contain all desired
   functionality for the session (certain parts that are supported but
   not desired by default may be omitted); for each SDP line, the
   generation of the SDP must follow the appropriate process for
   generating an offer.  In the event createOffer is called after the
   session is established, createOffer will generate an offer that is
   compatible with the current session, incorporating any changes that
   have been made to the session since the last complete offer-answer
   exchange, such as addition or removal of streams.  If no changes have
   been made, the offer will be identical to the current local
   description.

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

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

5.2.2.  createAnswer

   The createAnswer method generates a blob of SDP that contains a RFC
   3264 SDP answer with the supported configuration for the session that
   is compatible with the parameters supplied in the offer.  Like
   createOffer, the returned blob contains descriptions of the local



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   MediaStreams attached to this PeerConnection, the codec/RTP/RTCP
   options negotiated for this session, and any candidates that have
   been gathered by the ICE Agent.  A constraints parameter may be
   supplied to provide additional control over the generated answer.

   As an answer, the generated SDP will contain a specific configuration
   that specifies how the media plane should be established.

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

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

5.2.3.  SessionDescriptionType

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

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

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

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

   The application can use some discretion on whether an answer should
   be applied as provisional or final.  For example, in a serial forking
   scenario, an application may receive multiple "final" answers, one
   from each remote endpoint.  The application could accept the initial



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   answers as provisional answers, and only apply an answer as final
   when it receives one that meets its criteria (e.g. a live user
   instead of voicemail).

5.2.3.1.  Creating Answers

   Most web applications will not need to create answers using the
   "pranswer" type.  The general recommendation for a web application
   would be to create an answer more or less immediately after receiving
   the offer, instead of waiting for a human user to provide input.
   Later when the human input is received, the applications can create a
   new offer to update the previous offer/answer pair.  Some
   applications may not be able to do this, particularly ones that Some
   application may not be able to do this, particular ones that are
   attempting to gateway to other signaling protocols.

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

5.2.4.  setLocalDescription

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

   This API changes the local media state; among other things, it sets
   up local resources for receiving and decoding media.  In order to
   successfully handle scenarios where the application wants to offer to
   change from one media format to a different, incompatible format, the
   PeerConnection must be able to simultaneously support use of both the
   old and new local descriptions (e.g. support codecs that exist in
   both descriptions) until a final answer is received, at which point
   the PeerConnection can fully adopt the new local description, or roll
   back to the old description if the remote side denied the change.

   If setRemoteDescription was previous called with an offer, and
   setLocalDescription is called with an answer (provisional or final),
   and the media directions are compatible, this will result in the



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   starting of media transmission.

5.2.5.  setRemoteDescription

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

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

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

5.2.6.  localDescription

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

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

5.2.7.  remoteDescription

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

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

5.2.8.  updateIce

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

   Regardless of the configuration, the gathering process collects all



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   available candidates, but excluded candidates will not be surfaced in
   onicecallback or used for connectivity checks.

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

5.2.9.  addIceCandidate

   The addIceCandidate method provides a remote candidate to the ICE
   Agent, which will be added to the remote description.  Connectivity
   checks will be sent to the new candidate.

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



































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6.  Configurable SDP Parameters

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

   The following is a partial list of SDP parameters that an application
   may want to control, in either local or remote descriptions, using
   this API.

   o  remove or reorder codecs (m=)

   o  change codec attributes (a=fmtp; ptime)

   o  enable/disable BUNDLE (a=group)

   o  enable/disable RTCP mux (a=rtcp-mux)

   o  change send resolution or framerate (TBD)

   o  change desired recv resolution or framerate (TBD)

   o  change total bandwidth (b=)

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

   o  remove RTP header extensions (a=rtphdr-ext)

   o  add/change SSRC grouping (e.g.  FID, RTX, etc) (a=ssrc-group)

   o  add SSRC attributes (a=ssrc)

   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.













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7.  Security Considerations

   TODO
















































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8.  IANA Considerations

   This document requires no actions from IANA.
















































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9.  Acknowledgements

   Harald Alvestrand, Dan Burnett, Neil Stratford, Eric Rescorla, Anant
   Narayanan, and Adam Bergkvist all provided valuable feedback on this
   proposal.  Suhas Nandakumar provided text and input for SDP
   requirements.  Matthew Kaufman provided the observation that keeping
   state out of the browser allows a call to continue even if the page
   is reloaded.











































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10.  References

10.1.  Normative References

   [I-D.rescorla-mmusic-ice-trickle]
              Rescorla, E., Uberti, J., and E. Ivov, "Trickle ICE:
              Incremental Provisioning of Candidates for the Interactive
              Connectivity Establishment (ICE) Protocol",
              draft-rescorla-mmusic-ice-trickle-00 (work in progress),
              October 2012.

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

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

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

10.2.  Informative References

   [I-D.ietf-rtcweb-rtp-usage]
              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
              Communication (WebRTC): Media Transport and Use of RTP",
              draft-ietf-rtcweb-rtp-usage-04 (work in progress),
              July 2012.

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

   [I-D.nandakumar-rtcweb-sdp]
              Nandakumar, S. and C. Jennings, "SDP for the WebRTC",
              draft-nandakumar-rtcweb-sdp-00 (work in progress),
              October 2012.

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

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



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













































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

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

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

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




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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: peer.addStream(localStream, null);
   AnswererJS->AnswererUA: answer = peer.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:   peer.setRemoteDescription("answer", answer);
   OffererUA->OffererJS:   onaddstream(remoteStream);

   // ICE Completes (at Answerer)



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   AnswererUA->AnswererJS: onopen();
   AnswererUA->OffererUA:  Media

   // ICE Completes (at Offerer)
   OffererUA->OffererJS:   onopen();
   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.









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   // 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).




















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

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

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

   // ICE Completes (at Offerer)
   OffererUA->OffererJS:   onopen();
   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.










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   // 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->OffererJS:   onopen();
   OffererUA->AnswererUA:  Media

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





























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Appendix B.  Change log

   Changes in draft -02:

   o  Converted from nroff

   o  Removed comparisons to old approaches abandoned by the working
      group

   o  Removed stuff that has moved to W3C specificaiton

   o  Align SDP handling with W3C draft

   o  Clarified section on forking.

   Changes in draft -01:

   o  Added diagrams for architecture and state machine.

   o  Added sections on forking and rehydration.

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

   o  Reworked how ICE restarts and media directions are controlled.

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

   o  Updated suggested API and examples to match latest thinking.

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

   Changes in draft -00:

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

















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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

































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