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Network Working Group                                          J. Uberti
Internet-Draft                                                    Google
Intended status: Standards Track                             C. Jennings
Expires: December 6, 2012                            Cisco Systems, Inc.
                                                            June 4, 2012


               Javascript Session Establishment Protocol
                       draft-ietf-rtcweb-jsep-01

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.

   This document is an input document for discussion.  It should be
   discussed in the RTCWEB WG list, rtcweb@ietf.org.

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 July 26, 2012.

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



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   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


Table of Contents

   1. Introduction  . . . . . . . . . . . . . . . . . . . . . . . . .  4
     1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . .  5
   2. JSEP Approach . . . . . . . . . . . . . . . . . . . . . . . . .  5
   3. Other Approaches Considered . . . . . . . . . . . . . . . . . .  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. Separation of Signaling and ICE State Machines  . . . . . . 10
     4.5. ICE Candidate Trickling . . . . . . . . . . . . . . . . . . 10
     4.6. ICE Candidate Format  . . . . . . . . . . . . . . . . . . . 11
     4.7. Interactions With Forking . . . . . . . . . . . . . . . . . 11
       4.7.1. Serial Forking  . . . . . . . . . . . . . . . . . . . . 11
       4.7.2. Parallel Forking  . . . . . . . . . . . . . . . . . . . 12
     4.8. Session Rehydration . . . . . . . . . . . . . . . . . . . . 12
   5. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
     5.1. Methods . . . . . . . . . . . . . . . . . . . . . . . . . . 13
       5.1.1. createOffer . . . . . . . . . . . . . . . . . . . . . . 13
       5.1.2. createAnswer  . . . . . . . . . . . . . . . . . . . . . 14
       5.1.3. SessionDescriptionType  . . . . . . . . . . . . . . . . 14
       5.1.4. setLocalDescription . . . . . . . . . . . . . . . . . . 15
       5.1.5. setRemoteDescription  . . . . . . . . . . . . . . . . . 15
       5.1.6. localDescription  . . . . . . . . . . . . . . . . . . . 16
       5.1.7. remoteDescription . . . . . . . . . . . . . . . . . . . 16
       5.1.8. updateIce . . . . . . . . . . . . . . . . . . . . . . . 16
       5.1.9. addIceCandidate . . . . . . . . . . . . . . . . . . . . 17
     5.2. Configurable SDP Parameters . . . . . . . . . . . . . . . . 17
   6. Media Setup Overview  . . . . . . . . . . . . . . . . . . . . . 17
     6.1. Initiating the Session  . . . . . . . . . . . . . . . . . . 18
       6.1.1. Generating An Offer . . . . . . . . . . . . . . . . . . 18
       6.1.2. Applying the Offer  . . . . . . . . . . . . . . . . . . 18
       6.1.3. Handling ICE Callbacks  . . . . . . . . . . . . . . . . 18
       6.1.4. Serializing the Offer and Candidates  . . . . . . . . . 19
     6.2. Receiving the Session . . . . . . . . . . . . . . . . . . . 19
       6.2.1. Receiving the Offer . . . . . . . . . . . . . . . . . . 19
       6.2.2. Handling ICE Messages . . . . . . . . . . . . . . . . . 19
       6.2.3. Generating the Answer . . . . . . . . . . . . . . . . . 20
       6.2.4. Applying the Answer . . . . . . . . . . . . . . . . . . 20
       6.2.5. Serializing the Answer  . . . . . . . . . . . . . . . . 20
     6.3. Completing the Session  . . . . . . . . . . . . . . . . . . 20
       6.3.1. Receiving the Answer  . . . . . . . . . . . . . . . . . 20



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     6.4. Updates to the Session  . . . . . . . . . . . . . . . . . . 20
   7. Security Considerations . . . . . . . . . . . . . . . . . . . . 21
   8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 21
   9. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . 21
   10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 21
     10.1. Normative References . . . . . . . . . . . . . . . . . . . 21
     10.2. Informative References . . . . . . . . . . . . . . . . . . 21
   Appendix A. JSEP Implementation Examples . . . . . . . . . . . . . 22
     A.1. Example API . . . . . . . . . . . . . . . . . . . . . . . . 22
     A.2. Example API Flows . . . . . . . . . . . . . . . . . . . . . 23
       A.2.1. Call using ROAP . . . . . . . . . . . . . . . . . . . . 23
       A.2.2 Call using XMPP  . . . . . . . . . . . . . . . . . . . . 24
       A.2.3. Adding video to a call, using XMPP  . . . . . . . . . . 25
       A.2.4. Simultaneous add of video streams, using XMPP . . . . . 26
       A.2.5. Call using SIP  . . . . . . . . . . . . . . . . . . . . 27
       A.2.6. Handling early media (e.g. 1-800-FEDEX), using SIP  . . 28
     A.3. Full Example Application  . . . . . . . . . . . . . . . . . 28
   Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . . 30
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 30
































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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 original spec for WebRTC attempted to implement this protocol-
   agnostic signaling by providing a mechanism to exchange session
   descriptions in the form of SDP blobs. Upon starting a session, the
   browser would generate a SDP blob, which would be passed to the
   application for transport over its preferred signaling protocol. On
   the remote side, this blob would be passed into the browser from the
   application, and the browser would then generate a blob of its own in
   response. Upon transmission back to the initiator, this blob would be
   plugged into their browser, and the handshake would be complete.

   Experimentation with this mechanism turned up several shortcomings,
   which generally stemmed from there being insufficient context at the
   browser to fully determine the meaning of a SDP blob. For example,
   determining whether a blob is an offer or an answer, or
   differentiating a new offer from a retransmit.

   The ROAP proposal, specified in [I-D.draft-jennings-rtcweb-signaling-
   01], attempted to resolve these issues by providing additional
   structure in the messaging - in essence, to create a generic
   signaling protocol that specifies how the browser signaling state
   machine should operate. However, even though the protocol is
   abstracted, the state machine forces a least-common-denominator
   approach on the signaling interactions. For example, in Jingle, the
   call initiator can provide additional ICE candidates even after the
   initial offer has been sent, which allows the offer to be sent
   immediately for quicker call startup. However, in the browser state
   machine, there is no notion of sending an updated offer before the
   initial offer has been responded to, rendering this functionality
   impossible.

   While specific concerns like this could be addressed by modifying the
   generic protocol, others would likely be discovered later. The main
   reason this mechanism is inflexible is because it embeds a signaling
   state machine within the browser. Since the browser generates the
   session descriptions on its own, and fully controls the possible



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   states and advancement of the signaling state machine, modification
   of the session descriptions or use of alternate state machines
   becomes difficult or impossible.

   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.

   If instead the state machine is run at the browser end, and is
   instantiated within, for example, the PeerConnection object, that
   state machine will be reinitialized when the page is reloaded and the
   JavaScript re-executed. This actually complicates the design of any
   interoperability service, as all cases where an offer or answer has
   already been generated but is now "forgotten" must now be handled by
   trying to move the client state machine forward to the same state it
   had been in previously in order to match what has already been
   delivered to and/or answered by the far side, or handled by ensuring
   that aborts are cleanly handled from every state and the negotiation
   rapidly restarted.

1.1. 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].

2. JSEP Approach

   To resolve the issues mentioned above, this document proposes 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 can do massaging of that offer, if it wants to, and then
   uses it to set up its local config via a 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 a setRemoteDescription() API.



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   When the call is accepted, the callee uses a 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, since only the browser has the necessary knowledge of
   candidates and other transport info. Performing this separation it
   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 easily aggregated and recombined. 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, 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.

3. 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



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   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.

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



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   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
   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 both these methods take a parameter
   to indicate the type of session description being supplied. This
   satisfies the requirements listed above for both the offerer, who
   first calls setLocalDescription("offer", sdp) and then later
   setRemoteDescription("answer", sdp), as well as for the answerer, who
   first calls setRemoteDescription("offer", sdp) and then later
   setLocalDescription("answer", sdp). 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.




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   It also allows for an answer to be treated as provisional by the
   application. Provisional answers provide a way for an answerer to
   communicate session parameters back to the offerer, in order for 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; as a
   result, multiple dissimilar provisional answers can be received and
   applied during call setup.

   As in [RFC3264], an offerer can send an offer, and update it as long
   as it has not been answered. 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



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   In the current 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
   serialized out to SDP. If we were able to agree on a JSON format for
   session descriptions, we could easily enable this object to
   generate/expect JSON.

   Other methods may be added to SessionDescription in the future to
   simplify handling of SessionDescriptions from Javascript.

4.4. Separation of Signaling and ICE State Machines

   JSEP does away with the SDP Agent within the browser, and this
   functionality is now controlled directly by the application, which
   uses the setLocalDescription and setRemoteDescription APIs to tell
   the browser what SDP has been negotiated. The ICE Agent remains in
   the browser, as it still needs to drive the process of gathering
   candidates, connectivity checks, and related ICE functionality.

   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.

4.5. 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. This 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 allocate all
   possible candidates, resulting in faster call startup in many cases.

   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
   onicecandidate callback with a new candidate; applications that do



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   not support this feature can simply wait for the final onicecandidate
   callback that indicates gathering is complete, and create and send
   their offer, with all the candidates, at this time.

   Upon receipt of trickled candidates, the receiving application can
   supply them to its ICE Agent by calling an addIceCandidate method.
   This triggers the ICE Agent to start using this remote candidate for
   connectivity checks. Applications that do not make use of candidate
   tricking can ignore addIceCandidate entirely, and use the
   onicecandidate callback solely to indicate when candidate gathering
   is complete.

4.6. ICE Candidate Format

   As with session descriptions, we choose to provide an IceCandidate
   object that provides some abstraction, but can be easily converted
   to/from SDP a=candidate lines.

   The IceCandidate object has fields to indicate which m= line it
   should be associated with, and a method to convert to a SDP
   representation, ex:

      a=candidate:1 1 UDP 1694498815 66.77.88.99 10000 typ host

   Currently, 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).

4.7. Interactions With Forking

4.7.1. Serial Forking

   Serial 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
   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 send a terminate message to
   any subsequent answers. In SIP parlance, this would be ACK + BYE.



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   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.

4.7.2. Parallel Forking

   Parallel forking involves a call being dispatched to multiple remote
   callees, where each callee can accept the call, and multiple
   simultaneous active sessions can be established as a result. If
   multiple callees send media, this media is mixed and played out at
   the caller side.

   JSEP can handle parallel forking by "cloning" the session when needed
   to create multiple parallel sessions. When the first answer is
   received, the caller can clone the existing session, and then apply
   the answer as a final answer to the original session. Upon receiving
   the next answer, the cloned session is cloned again, and the received
   answer is applied as a final answer to the first clone. This process
   repeats until the caller decides to end the setup flow, and closes
   the final cloned session.

   Cloned sessions inherit the local session description and candidates
   from their parent, and an empty remote description; only sessions
   that have not yet applied an answer can be cloned. Each cloned
   session may discover new peer-reflexive candidates; these candidates
   will be supplied via the onicecandidate callback to that specific
   session. Since the clone uses the same local description as its
   parent, creating a clone will fail if it is not possible to reserve
   the same resources for the clone as have already been reserved by the
   parent.

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

4.8. 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



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   "rehydration".

   With rehydration, the current local session description 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 local session is
   now retrieved, but the previous ICE candidates will no longer be
   valid in this case, so we will need to perform an ICE restart; to do
   so, we simply generate a new ICE ufrag/pwd combo for the local
   description.

   The modified local description is then installed via
   setLocalDescription, and sent off as an offer to the remote side, who
   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.

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. Methods

5.1.1. 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



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   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 does not change state; its use is not required.

5.1.2. 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 |offer|. Like
   createOffer, the returned blob contains descriptions of the local
   MediaStreams attached to this PeerConnection, the codec/RTP/RTCP
   options negotiated for this session, and any candidates that have
   been gathered by the ICE Agent. A constraints parameter may be
   supplied to provide additional control over the generated answer.

   As an answer, the generated SDP will contain a specific configuration
   that specifies how the media plane should be established. For each
   SDP line, the generation of the SDP must follow the appropriate
   process for generating an answer.

   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 does not change state; its use is not required.

5.1.3. SessionDescriptionType

   The strings "offer", "pranswer", and "answer" serve as type arguments
   to setLocalDescription and setRemoteDescription. They 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



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   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
   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).

5.1.4. setLocalDescription

   The setLocalDescription method instructs the PeerConnection to apply
   the supplied SDP blob as its local configuration. The type parameter
   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
   starting of media transmission.

5.1.5. setRemoteDescription



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   The setRemoteDescription method instructs the PeerConnection to apply
   the supplied SDP blob as the desired remote configuration. As in
   setLocalDescription, the |type| parameter 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.

5.1.6. localDescription

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

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

5.1.7. 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.

5.1.8. updateIce

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

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

   This call may result in a change to the state of the ICE Agent, and



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   may result in a change to media state if it results in connectivity
   being established.

5.1.9. 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.

5.2. Configurable SDP Parameters

   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.

    - remove or reorder codecs (m=)
    - change codec attributes (a=fmtp; ptime)
    - enable/disable BUNDLE (a=group)
    - enable/disable RTCP mux (a=rtcp-mux)
    - remove or reorder SRTP crypto-suites (a=crypto)
    - change SRTP parameters or keys (a=crypto)
    - change send resolution or framerate (TBD)
    - change desired recv resolution or framerate (TBD)
    - change total bandwidth (b=)
    - remove desired AVPF mechanisms (a=rtcp-fb)
    - remove RTP header extensions (a=rtphdr-ext)
    - add/change SSRC grouping (e.g. FID, RTX, etc) (a=ssrc-group)
    - add SSRC attributes (a=ssrc)
    - change ICE ufrag/password (a=ice-ufrag/pwd)
    - 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.

6. Media Setup Overview

   The example here shows a typical call setup using the JSEP model,
   indicating the functions that are called and the state changes that
   occur. We assume the following architecture in this example, where UA
   is synonymous with "browser", and JS is synonymous with "web
   application":

   OffererUA <-> OffererJS <-> WebServer <-> AnswererJS <-> AnswererUA



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6.1. Initiating the Session

   The initiator creates a PeerConnection, hooks up to its ICE callback,
   and adds the desired MediaStreams (presumably obtained via
   getUserMedia). The ICE gathering process begins to gather candidates
   for a default number of streams, as the exact number will not be
   known until the local description is applied. The PeerConnection is
   in the NEW state.

   OffererJS->OffererUA: var pc = new PeerConnection(config, null);
   OffererJS->OffererUA: pc.onicecandidate = onIceCandidate;
   OffererJS->OffererUA: pc.addStream(stream);

6.1.1. Generating An Offer

   The initiator then creates a session description to offer to the
   callee. This description includes the codecs and other necessary
   session parameters, as well as information about each of the streams
   that has been added (e.g. SSRC, CNAME, etc.) The created description
   includes all parameters that the offerer's UA supports; if the
   initiator wants to influence the created offer, they can pass in a
   MediaConstraints object to createOffer that allows for customization
   (e.g. if the initiator wants to receive but not send video). The
   initiator can also directly manipulate the created session
   description as well, perhaps if it wants to change the priority of
   the offered codecs.

   OffererJS->OffererUA: var offer = pc.createOffer(null);

6.1.2. Applying the Offer

   The initiator then instructs the PeerConnection to use this offer as
   the local description for this session, i.e. what codecs it will use
   for received media, what SRTP keys it will use for sending media (if
   using SDES), etc. In order that the UA handle the description
   properly, the initiator marks it as an offer when calling
   setLocalDescription; this indicates to the UA that multiple
   capabilities have been offered, but this set may be pared back later,
   when the answer arrives.

   Since the local user agent must be prepared to receive media upon
   applying the offer, this operation will cause local decoder resources
   to be allocated, based on the codecs indicated in the offer.

   OffererJS->OffererUA: pc.setLocalDescription("offer", offer);

6.1.3. Handling ICE Callbacks




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   The initiator starts to receive callbacks on its onicecandidate
   handler. Candidates are provided to the IceCallback as they are
   allocated; when the last allocation completes or times out, this
   callback will be invoked with a null argument.

   OffererUA->OffererJS: onIceCandidate(candidate);

6.1.4. Serializing the Offer and Candidates

   At this point, the offerer is ready to send its offer to the callee
   using its preferred signaling protocol. Depending on the protocol, it
   can either send the initial session description first, and then
   "trickle" the ICE candidates as they are given to the application, or
   it can wait for all the ICE candidates to be collected, and then send
   the offer and list of candidates all at once.

6.2. Receiving the Session

   Through the chosen signaling protocol, the recipient is notified of
   an incoming session request. It creates a PeerConnection, and sets up
   its own ICE callback. The ICE gathering process begins to gather
   candidates for a default number of streams.

   AnswererJS->AnswererUA: var pc = new PeerConnection(config, null);
   AnswererJS->AnswererUA: pc.onicecandidate = onIceCandidate;

6.2.1. Receiving the Offer

   The recipient converts the received offer from its signaling protocol
   into SDP format, and supplies it to its PeerConnection, again marking
   it as an offer. As a remote description, the offer indicates what
   codecs the remote side wants to use for receiving, as well as what
   SRTP keys it will use for sending. The setting of the remote
   description causes callbacks to be issued, informing the application
   of what kinds of streams are present in the offer.

   This step will also cause encoder resources to be allocated, based on
   the codecs specified in |offer|.

   AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
   AnswererUA->AnswererJS: onAddStream(stream);

6.2.2. Handling ICE Messages

   If ICE candidates from the remote site were included in the offer,
   the ICE Agent will automatically start trying to use them. Otherwise,
   if ICE candidates are sent separately, they are passed into the
   PeerConnection when they arrive.



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   AnswererJS->AnswererUA: pc.addIceCandidate(candidate);

6.2.3. Generating the Answer

   Once the recipient has decided to accept the session, it generates an
   answer session description. This process performs the appropriate
   intersection of codecs and other parameters to generate the correct
   answer. As with the offer, MediaConstraints can be provided to
   influence the answer that is generated, and/or the application can
   post-process the answer manually.

   AnswererJS->AnswererUA: pc.createAnswer(offer, null);

6.2.4. Applying the Answer

   The recipient then instructs the PeerConnection to use the answer as
   its local description for this session, i.e. what codecs it will use
   to receive media, etc. It also marks the description as an answer,
   which tells the UA that these parameters are final. This causes the
   PeerConnection to move to the ACTIVE state, and transmission of media
   by the answerer to start (assuming both sides have indicated this in
   their descriptions).

   AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
   AnswererUA->OffererUA:  <media>

6.2.5. Serializing the Answer

   As with the offer, the answer (with or without candidates) is now
   converted to the desired signaling format and sent to the initiator.

6.3. Completing the Session

6.3.1. Receiving the Answer

   The initiator converts the answer from the signaling protocol and
   applies it as the remote description, marking it as an answer. This
   causes the PeerConnection to move to the ACTIVE state, and
   transmission of media by the offerer to start (assuming both sides
   have indicated this in their descriptions).

   OffererJS->OffererUA:  pc.setRemoteDescription("answer", answer);
   OffererUA->AnswererUA: <media>

6.4. Updates to the Session

   Updates to the session are handled with a new offer/answer exchange.
   However, since media will already be flowing at this point, the new



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   offerer needs to support both its old session description as well as
   the new one it has offered, until the change is accepted by the
   remote side.

   Note also that in an update scenario, the roles may be reversed, i.e.
   the update offerer can be different than the original offerer.

7. Security Considerations

   TODO

8. IANA Considerations

   This document requires no actions from IANA.

9. Acknowledgements

   Harald Alvestrand, Dan Burnett, Neil Stratford, Eric Rescorla, Anant
   Narayanan, 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. Richard Ejzak provided the specifics on session cloning.

10. References

10.1. Normative References

   [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

   [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.

   [webrtc-api] Bergkvist, Burnett, Jennings, Narayanan, "WebRTC 1.0:



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   Real-time Communication Between Browsers", May 2011.

   Available at http://dev.w3.org/2012/webrtc/editor/webrtc.html

Appendix A. JSEP Implementation Examples

A.1. Example API

   The interface below shows a basic Javascript API that could be used
   to expose the functionality discussed in this document. This API is
   used for the examples in the following parts of this Appendix.

   // actions, for setLocalDescription/setRemoteDescription
   enum SessionDescriptionType { "offer", "pranswer", "answer" }

   // constraints that can be supplied to the ctor or createXXXX
   enum MediaConstraints {
       "offerConfig",    // controls the kind of offer created;
                         //   "default"    (normal offer)
                         //   "caps"       (all capabilities)
                         //   "new"        (brand new description)
                         //   "iceRestart" (new ICE creds)

       "iceTransports",  // controls ICE candidates; can be
                         //   "none"  (no candidates)
                         //   "relay" (only relay candidates)
                         //   "all"   (all available candidates)
   }

   [Constructor (int index, DOMString id, in DOMString candidateLine)]
   interface IceCandidate {
       // the m= line index for this candidate
       readonly attribute int mLineIndex
       // the mid for the m= line for this candidate
       readonly attribute DOMString mLineId;
       // creates a SDP-ized form of this candidate
       stringifier DOMString ();
   };

   [Constructor (DOMString sdp)]
   interface SessionDescription {
       // adds the specified candidate to the description
       void addCandidate(IceCandidate candidate);
       // serializes the description to SDP
       stringifier DOMString ();
   };

   [Constructor (DOMString configuration,



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                 optional MediaConstraints constraints)]
   interface PeerConnection {
       // creates a blob of SDP to be provided as an offer.
       SessionDescription createOffer (
           SessionDescriptionCallback successCb,
           optional ErrorCallback errorCb,
           optional MediaContraints constraints);
       // creates a blob of SDP to be provided as an answer.
       SessionDescription createAnswer (
           SessionDescription offer,
           SessionDescriptionCallback successCb,
           optional ErrorCallback errorCb,
           optional MediaContraints constraints);

       // sets the local session description
       void setLocalDescription (
           SessionDescriptionType action,
           SessionDescription desc);
       // sets the remote session description
       void setRemoteDescription (
           SessionDescriptionType action,
           SessionDescription desc)
       // returns the current local session description
       readonly attribute SessionDescription localDescription;
       // returns the current remote session description
       readonly attribute SessionDescription remoteDescription;

       // updates the constraints for ICE processing
       void updateIce (
         optional DOMString configuration,
         optional MediaConstraints constraints);
       // starts using a received remote ICE candidate
       void addIceCandidate (
         IceCandidate candidate);
       // notifies the application of a new local ICE candidate
       attribute Function?          onicecandidate;
   };

A.2. 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.2.1. Call using ROAP




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   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

A.2.2 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"/>



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

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

A.2.3. 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.




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   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.2.4. 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




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   // 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.2.5. 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: 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




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

A.2.6. Handling early media (e.g. 1-800-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->OffererJS:   onopen();
   OffererUA->AnswererUA:  Media

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

A.3. Full Example Application

   The following example demonstrates a simple video calling
   application, using both trickle candidates and provisional answers to
   speed up call setup.

   // Usage:
   // Caller calls start(true)
   // Callee calls start(false) to prepare the call/start connecting,
   // and then accept() to start transmitting.

   var signalingChannel = createSignalingChannel();
   var pc = null;
   var localStream = null;
   signalingChannel.onmessage = handleMessage;




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   // Set up the call, get access to local media,
   // and establish connectivity.
   function start(isCaller) {
     // Create a PeerConnection and hook up the IceCallback.
     pc = new webkitPeerConnection(null, null);
     pc.onicecandidate = function(evt) {
       sendMessage("candidate", evt.candidate);
     };

     // Get the local stream and show it in the local video element;
     // if we're the caller, ship off an offer once we get the stream.
     navigator.webkitGetUserMedia(
         {"audio": true, "video": true}, function (stream) {
       selfView.src = webkitURL.createObjectURL(stream);
       localStream = stream;
       if (isCaller) {
         pc.addStream(stream);
         pc.createOffer(function(sdp) {
             setLocalAndSendMessage("offer", sdp);
         });
     });

     // When the remote stream arrives, show it in the remote
     // video element.
     pc.onaddstream = function(evt) {
       remoteView.src = webkitURL.createObjectURL(evt.stream);
     };
   }

   // The callee has accepted the call, attach their media
   // and send a final answer.
   function accept() {
     // The addStream could also be done for the pranswer,
     // although that would delay the pranswer
     // (due to the need for user consent)
     pc.addStream(localStream);  // assumes we have the stream already
     pc.createAnswer(msg.sdp, function(sdp) {
       setLocalAndSendMessage("answer", sdp);
     });
   }

   // -- internal methods --

   // Apply SDP locally and send it to the remote side.
   function setLocalAndSendMessage(type, sdp) {
     pc.setLocalDescription(type, sdp);
     sendMessage(type, sdp);
   }



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   // Send a signaling message to the remote side.
   function sendMessage(type, obj) {
     signalingChannel.send(
          JSON.stringify({ "type": type, "sdp": obj }));
   }

   // Handle incoming signaling messages.
   function handleMessage(str) {
     var msg = JSON.parse(str);
     switch (msg.type) {
       case "offer":
         // create the PeerConnection
         start(false);
         // feed the received offer into the PeerConnection
         pc.setRemoteDescription(msg.type, msg.sdp);
         // create provisional answer to allow ICE/DTLS to start
         pc.createAnswer(msg.sdp, function(sdp) {
           setDirection(sdp, "recvonly");
           setLocalAndSendMessage("pranswer", sdp);
         });
         break;
       case "pranswer":
       case "answer":
         pc.setRemoteDescription(msg.type, msg.sdp);
         break;
       case "candidate":
         pc.addIceCandidate(msg.sdp);
         break;
     }
   }

Appendix B. Change log

   01: Added diagrams for architecture and state machine.
       Added sections on forking and rehydration.
       Clarified meaning of "pranswer" and "answer".
       Reworked how ICE restarts and media directions are controlled.
       Added list of parameters that can be changed in a description.
       Updated suggested API and examples to match latest thinking.
       Suggested API and examples have been moved to an appendix.
   00: Migrated from draft-uberti-rtcweb-jsep-02.

Authors' Addresses

   Justin Uberti
   Google
   5 Cambridge Center
   Cambridge, MA 02142



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   Email: justin@uberti.name

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

   Email:  fluffy@cisco.com










































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