[Docs] [txt|pdf] [Tracker] [Email] [Diff1] [Diff2] [Nits]

Versions: 00 01 02 draft-ietf-rtcweb-jsep

Network Working Group                                          J. Uberti
Internet-Draft                                                    Google
Network Working Group                                        C. Jennings
Internet-Draft                                       Cisco Systems, Inc.
Intended status: Standards Track                       February 16, 2012
Expires: August 19, 2012


               Javascript Session Establishment Protocol
                      draft-uberti-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.

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

Requirements Language

   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].

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.



Uberti                  Expires August 19, 2012                 [Page 1]

Internet-Draft                    JSEP                 February 16, 2012


   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.


Table of Contents

   1. Introduction  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   2. JSEP Approach . . . . . . . . . . . . . . . . . . . . . . . . .  5
   3. Other Approaches Considered . . . . . . . . . . . . . . . . . .  6
   4. Semantics and Syntax  . . . . . . . . . . . . . . . . . . . . .  7
     4.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . .  7
     4.2. Session Descriptions  . . . . . . . . . . . . . . . . . . .  7
     4.3. Session Description Format  . . . . . . . . . . . . . . . .  8
     4.4. Separation of Signaling and ICE State Machines  . . . . . .  9
     4.5. ICE Candidate Trickling . . . . . . . . . . . . . . . . . .  9
     4.6. ICE Candidate Format  . . . . . . . . . . . . . . . . . . . 10
   5. Media Setup Overview  . . . . . . . . . . . . . . . . . . . . . 10
     5.1. Initiating the Session  . . . . . . . . . . . . . . . . . . 10
       5.1.1. Generating An Offer . . . . . . . . . . . . . . . . . . 10
       5.1.2. Applying the Offer  . . . . . . . . . . . . . . . . . . 11
       5.1.3. Initiating ICE  . . . . . . . . . . . . . . . . . . . . 11
       5.1.4. Serializing the Offer and Candidates  . . . . . . . . . 11
     5.2. Receiving the Session . . . . . . . . . . . . . . . . . . . 12
       5.2.1. Receiving the Offer . . . . . . . . . . . . . . . . . . 12
       5.2.2. Initiating ICE  . . . . . . . . . . . . . . . . . . . . 12
       5.2.3. Handling ICE Messages . . . . . . . . . . . . . . . . . 12
       5.2.4. Generating the Answer . . . . . . . . . . . . . . . . . 12
       5.2.5. Applying the Answer . . . . . . . . . . . . . . . . . . 13
       5.2.6. Serializing the Answer  . . . . . . . . . . . . . . . . 13
     5.3. Completing the Session  . . . . . . . . . . . . . . . . . . 13
       5.3.1. Receiving the Answer  . . . . . . . . . . . . . . . . . 13
     5.4. Updates to the Session  . . . . . . . . . . . . . . . . . . 13
   6. Proposed WebRTC API changes . . . . . . . . . . . . . . . . . . 13
     6.1. PeerConnection API  . . . . . . . . . . . . . . . . . . . . 13
       6.1.1 MediaHints . . . . . . . . . . . . . . . . . . . . . . . 15
       6.1.2 createOffer  . . . . . . . . . . . . . . . . . . . . . . 15
       6.1.3 createAnswer . . . . . . . . . . . . . . . . . . . . . . 16
       6.1.4 SDP_OFFER, SDP_PRANSWER, and SDP_ANSWER  . . . . . . . . 17
       6.1.5 setLocalDescription  . . . . . . . . . . . . . . . . . . 17
       6.1.6 setRemoteDescription . . . . . . . . . . . . . . . . . . 18
       6.1.7 localDescription . . . . . . . . . . . . . . . . . . . . 18



Uberti                  Expires August 19, 2012                 [Page 2]

Internet-Draft                    JSEP                 February 16, 2012


       6.1.8 remoteDescription  . . . . . . . . . . . . . . . . . . . 19
       6.1.9 IceOptions . . . . . . . . . . . . . . . . . . . . . . . 19
       6.1.10 startIce  . . . . . . . . . . . . . . . . . . . . . . . 19
       6.1.11 processIceMessage . . . . . . . . . . . . . . . . . . . 19
   7. Example API Flows . . . . . . . . . . . . . . . . . . . . . . . 20
     7.1. Call using ROAP . . . . . . . . . . . . . . . . . . . . . . 20
     7.2. Call using XMPP . . . . . . . . . . . . . . . . . . . . . . 21
     7.3. Adding video to a call, using XMPP  . . . . . . . . . . . . 22
     7.4. Simultaneous add of video streams, using XMPP . . . . . . . 22
     7.5. Call using SIP  . . . . . . . . . . . . . . . . . . . . . . 23
     7.6. Handling early media (e.g. 1-800-FEDEX), using SIP  . . . . 24
   8. Example Application . . . . . . . . . . . . . . . . . . . . . . 25
   9. Security Considerations . . . . . . . . . . . . . . . . . . . . 26
   10. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 26
   11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 26
   12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 26
     12.1. Normative References . . . . . . . . . . . . . . . . . . . 26
     12.2. Informative References . . . . . . . . . . . . . . . . . . 27
   Appendix A. Open Issues  . . . . . . . . . . . . . . . . . . . . . 27
   Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . . 27
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 27






























Uberti                  Expires August 19, 2012                 [Page 3]

Internet-Draft                    JSEP                 February 16, 2012


1. Introduction

   The general 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 http://tools.ietf.org/html/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



Uberti                  Expires August 19, 2012                 [Page 4]

Internet-Draft                    JSEP                 February 16, 2012


   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.


2. JSEP Approach

   To resolve these issues, 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 on
   PeerConnection. 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.

   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



Uberti                  Expires August 19, 2012                 [Page 5]

Internet-Draft                    JSEP                 February 16, 2012


   complete. This process can be repeated for additional offer/answer
   exchanges.

   Regarding ICE, in this approach we decouple the ICE state machine
   from the overall signaling state machine; the ICE state machine must
   remain in the browser, given that only the browser has the necessary
   knowledge of candidates and other transport info. While transport has
   typically been lumped in with session descriptions, 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 desciptions 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. This approach would add a getCapabilities API which would
   provide the application with the information it needed in order to
   generate 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 available capabilities.
   While this could certainly be addressed by using a library like the
   one mentioned above, some experimentation also indicates that coming
   up with a sufficiently complete getCapabilities API is a nontrivial
   undertaking. Nevertheless, if we wanted to go down this road, JSEP



Uberti                  Expires August 19, 2012                 [Page 6]

Internet-Draft                    JSEP                 February 16, 2012


   makes it significantly easier; if a getCapabilities API is added in
   the future, the application can generate session descriptions
   accordingly and pass those to the
   setLocalDescription/setRemoteDescription APIs added by JSEP. (Even
   with JSEP, an application could still perform its own browser
   fingerprinting and generate approximate session descriptions as a
   result.)

   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 PeerConnection. 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.

4.2. Session Descriptions

   In order to establish the media plane, PeerConnection 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; 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.




Uberti                  Expires August 19, 2012                 [Page 7]

Internet-Draft                    JSEP                 February 16, 2012


   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,
   PeerConnection 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 as a first
   parameter either the value SDP_OFFER, SDP_PRANSWER (for a non-final
   answer) or SDP_ANSWER (for a final answer). This satisfies the
   requirements listed above for both the offerer, who first calls
   setLocalDescription(SDP_OFFER, sdp) and then later
   setRemoteDescription(SDP_ANSWER, sdp), as well as for the answerer,
   who first calls setRemoteDescription(SDP_OFFER, sdp) and then later
   setLocalDescription(SDP_ANSWER, sdp).

   While it could be possible to implicitly determine the value of the
   offer/answer argument inside of PeerConnection, requiring it to be
   specified explicitly seems substantially more robust, allowing
   invalid combinations (i.e. an answer before an offer) to generate an
   appropriate error.

4.3. Session Description Format

   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.



Uberti                  Expires August 19, 2012                 [Page 8]

Internet-Draft                    JSEP                 February 16, 2012


   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

   Previously, PeerConnection operated two state machines, referred to
   in the spec as an "ICE Agent", which handles the establishment of
   peer-to-peer connectivity, and an "SDP Agent", which handles the
   state of the offer-answer signaling. The states of these state
   machines were exposed through the iceState and sdpState attributes on
   PeerConnection, with an additional readyState attribute that
   reflected the high-level state of the PeerConnection.

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

   The net effect of this is that sdpState goes away, and
   processSignalingMessage becomes processIceMessage, which now
   specifically handles incoming ICE candidates. To allow the
   application to control exactly when it wants to start ICE negotiation
   (e.g. either on receipt of the call, or only after accepting the
   call), a startIce method has been added.

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 a
   callback with a new candidate; applications that do not support this



Uberti                  Expires August 19, 2012                 [Page 9]

Internet-Draft                    JSEP                 February 16, 2012


   feature can simply wait for the callback that indicates gathering is
   complete, and simply create and send their offer, with all the
   candidates, at this time.

   To be clear, aplications that do not make use of candidate tricking
   can ignore processIceMessage entirely, and use IceCallback 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 a field 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 thing that are contained
   within IceCandidate, as this is the only information that is needed
   that is not present in the initial offer (i.e. for trickle
   candidates).

5. Media Setup Overview

   The example here shows a typical call setup using the JSEP model. 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

5.1. Initiating the Session

   The initiator creates a PeerConnection, installs its IceCallback, and
   adds the desired MediaStreams (presumably obtained via getUserMedia).
   The PeerConnection is in the NEW state.

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

5.1.1. 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



Uberti                  Expires August 19, 2012                [Page 10]

Internet-Draft                    JSEP                 February 16, 2012


   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
   MediaHints 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 offerered codecs.

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

5.1.2. 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(SDP_OFFER, offer);

5.1.3. Initiating ICE

   The initiator can now start the ICE process of candidate generation
   and connectivity checking. This results in callbacks to the
   application's IceCallback. Candidates are provided to the IceCallback
   as they are allocated, with the |moreToFollow| argument set to true
   if there are still allocations pending; when the last allocation
   completes or times out, this callback will be invoked with
   |moreToFollow| set to false.

   OffererJS->OffererUA: pc.startIce();
   OffererUA->OffererJS: iceCallback(candidate, ...);

5.1.4. 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.



Uberti                  Expires August 19, 2012                [Page 11]

Internet-Draft                    JSEP                 February 16, 2012


5.2. Receiving the Session

   Through the chosen signaling protocol, the recipient is notified of
   an incoming session request. It creates a PeerConnection, and
   installs its own IceCallback.

   AnswererJS->AnswererUA: var pc = new PeerConnection(config, iceCb);

5.2.1. 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(SDP_OFFER, offer);
   AnswererUA->AnswererJS: onAddStream(stream);

5.2.2. Initiating ICE

   The recipient then starts its own ICE state machine, to allow
   connectivity to be established as quickly as possible.

   AnswererJS->AnswererUA: pc.startIce();
   AnswererUA->AnswererJS: iceCallback(candidate, ...);

5.2.3. 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.

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

5.2.4. 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, MediaHints can be provided to influence
   the answer that is generated, and/or the application can post-process



Uberti                  Expires August 19, 2012                [Page 12]

Internet-Draft                    JSEP                 February 16, 2012


   the answer manually.

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

5.2.5. 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.

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

5.2.6. 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.

5.3. Completing the Session

5.3.1. 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.

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

5.4. 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
   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.

6. Proposed WebRTC API changes

6.1. PeerConnection API




Uberti                  Expires August 19, 2012                [Page 13]

Internet-Draft                    JSEP                 February 16, 2012


   The text below indicates the recommended changes to the
   PeerConnection API to implement the JSEP functionality. Methods
   marked with a [+] are new/proposed; methods marked with a [-] have
   been removed in this proposal.

   [Constructor (in DOMString configuration, in IceCallback iceCb)]
   interface PeerConnection {
       // creates a blob of SDP to be provided as an offer.
   [+] SessionDescription createOffer (MediaHints hints);
       // creates a blob of SDP to be provided as an answer.
   [+] SessionDescription createAnswer (DOMString offer,
                                        MediaHints hints);
       // actions, for setLocalDescription/setRemoteDescription
   [+] const unsigned short SDP_OFFER = 0x100;
   [+] const unsigned short SDP_PRANSWER = 0x200;
   [+] const unsigned short SDP_ANSWER = 0x300;
       // sets the local session description
   [+] void setLocalDescription (unsigned short action,
                                 SessionDescription desc);
       // sets the remote session description
   [+] void setRemoteDescription (unsigned short action,
                                  SessionDescription desc);
       // returns the current local session description
   [+] readonly SessionDescription localDescription;
       // returns the current remote session description
   [+] readonly SessionDescription remoteDescription;
   [-] void processSignalingMessage (DOMString message);
       const unsigned short NEW = 0;     // initial state
   [+] const unsigned short OPENING = 1; // local or remote desc set
       const unsigned short ACTIVE = 2;  // local and remote desc set
       const unsigned short CLOSED = 3;  // ended state
       readonly attribute unsigned short readyState;
       // starts ICE connection/handshaking
   [+] void startIce (optional IceOptions options);
       // processes received ICE information
   [+] void processIceMessage (IceCandidate candidate);
       const unsigned short ICE_GATHERING = 0x100;
       const unsigned short ICE_WAITING = 0x200;
       const unsigned short ICE_CHECKING = 0x300;
       const unsigned short ICE_CONNECTED = 0x400;
       const unsigned short ICE_COMPLETED = 0x500;
       const unsigned short ICE_FAILED = 0x600;
       const unsigned short ICE_CLOSED = 0x700;
       readonly attribute unsigned short iceState;
   [-] const unsigned short SDP_IDLE = 0x1000;
   [-] const unsigned short SDP_WAITING = 0x2000;
   [-] const unsigned short SDP_GLARE = 0x3000;
   [-] readonly attribute unsigned short sdpState;



Uberti                  Expires August 19, 2012                [Page 14]

Internet-Draft                    JSEP                 February 16, 2012


       void addStream (MediaStream stream, MediaStreamHints hints);
       void removeStream (MediaStream stream);
       readonly attribute MediaStream[]  localStreams;
       readonly attribute MediaStream[]  remoteStreams;
       void close ();
       [ rest of interface omitted ]
   };

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

   [Constructor (in DOMString label, in DOMString candidateLine)]
   interface IceCandidate {
     // the m= line this candidate is associated with
     readonly DOMString label;
     // creates a SDP-ized form of this candidate
     DOMString toSdp();
   };

6.1.1 MediaHints

   MediaHints is an object that can be passed into createOffer or
   createAnswer to affect the type of offer/answer that is generated.

   The following properties can be set on MediaHints:

      has_audio: boolean

      Indicates whether we want to receive audio; defaults to true if we
      have audio streams, else false

      has_video: boolean

      Indicates whether we want to receive video; defaults to true if we
      have video streams, else false

   As an example, MediaHints could be used to create a session that
   transmits only audio, but is able to receive video from the remote
   side, by forcing the inclusion of a m=video line even when no video
   sources are provided.

6.1.2 createOffer




Uberti                  Expires August 19, 2012                [Page 15]

Internet-Draft                    JSEP                 February 16, 2012


   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. The |hints| parameter may be supplied to provide additional
   control over the generated offer.

   As an offer, the generated SDP will contain the full set of
   capabilities supported by the session (as opposed to an answer, which
   will include only a specific negotiated subset to use); 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.

   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.

   Calling this method does not change the state of the PeerConnection;
   its use is not required.

   A TBD exception is thrown if the |hints| parameter is malformed.

6.1.3 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. The |hints| parameter may be supplied



Uberti                  Expires August 19, 2012                [Page 16]

Internet-Draft                    JSEP                 February 16, 2012


   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 setRemoteDescription; like createOffer, the returned
   description should reflect the current state of the system.

   Session descriptions generated by createAnswer must be immediately
   usable by setRemoteDescription; like createOffer, the returned
   description should reflect the current state of the system.

   Calling this method does not change the state of the PeerConnection;
   its use is not required.

   A TBD exception is thrown if the |hints| parameter is malformed, or
   the |offer| parameter is missing or malformed.

6.1.4 SDP_OFFER, SDP_PRANSWER, and SDP_ANSWER

   The SDP_XXXX enums serve as 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.

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

   SDP_PRANSWER indicates that a description should be parsed as an
   answer, but not a final answer, and so should not result in the
   starting of media transmission. A description used as a SDP_PRANSWER
   may be applied as a response to a SDP_OFFER, or an update to a
   previously sent SDP_PRANSWER.

   SDP_ANSWER indicates that a description should be parsed as an
   answer, and the offer-answer exchange should be considered complete.
   A description used as a SDP_ANSWER may be applied as a response to a
   SDP_OFFER, or an update to a previously send SDP_PRANSWER.

6.1.5 setLocalDescription

   The setLocalDescription method instructs the PeerConnection to apply



Uberti                  Expires August 19, 2012                [Page 17]

Internet-Draft                    JSEP                 February 16, 2012


   the supplied SDP blob as its local configuration. The |type|
   parameter indicates whether the blob should be processed as an offer
   (SDP_OFFER), provisional answer (SDP_PRANSWER), or final answer
   (SDP_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.

   Changes to the state of media transmission will only occur when a
   final answer is successfully applied.

   A TBD exception is thrown if |description| is invalid. A TBD
   exception is thrown if there are insufficient local resources to
   apply |description|.

6.1.6 setRemoteDescription

   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.

   Changes to the state of media transmission will only occur when a
   final answer is successfully applied.

   A TBD exception is thrown if |description| is invalid. A TBD
   exception is thrown if there are insufficient local resources to
   apply |description|.

6.1.7 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



Uberti                  Expires August 19, 2012                [Page 18]

Internet-Draft                    JSEP                 February 16, 2012


   been established.

6.1.8 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.

6.1.9 IceOptions

   IceOptions is an object that can be passed into startIce to restrict
   the candidates that are provided to the application and used for
   connectivity checks. This can be useful if the application wants to
   only use TURN candidates for privacy reasons, or only local + STUN
   candidates for cost reasons.

   The following properties can be set on IceOptions:

      use_candidates: "all", "no_relay", "only_relay"

      Indicates what types of local candidates should be used; defaults
      to "all"

6.1.10 startIce

   The startIce method starts or updates the ICE Agent process of
   gathering local candidates and pinging remote candidates. The
   |options| argument can be used to restrict which types of local
   candidates are provided to the application and used for pinging; this
   can be used to limit the use of TURN candidates by a callee to avoid
   leaking location information prior to the call being accepted.

   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.

   A TBD exception will be thrown if |options| is malformed.

6.1.11 processIceMessage

   The processIceMessage method provides a remote candidate to the ICE
   Agent, which will be added to the remote description. If startIce has
   been called, connectivity checks will be sent to the new candidates.




Uberti                  Expires August 19, 2012                [Page 19]

Internet-Draft                    JSEP                 February 16, 2012


   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.

   A TBD exception will be thrown if |candidate| is missing or
   malformed.

7. 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.

7.1. 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);
   OffererJS->OffererUA:   pc.startIce();
   OffererUA->OffererJS:   iceCallback(candidate, false);
   OffererJS->OffererUA:   offer = pc.createOffer(null);
   OffererJS->OffererUA:   pc.setLocalDescription(SDP_OFFER, offer.toSdp());
   OffererJS->AnswererJS:  {"type":"OFFER", "sdp":"<offer>"}

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

   // Answerer accepts call
   AnswererJS->AnswererUA: peer.addStream(localStream, null);
   AnswererJS->AnswererUA: answer = peer.createAnswer(msg.offer, null);
   AnswererJS->AnswererUA: peer.setLocalDescription(SDP_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




Uberti                  Expires August 19, 2012                [Page 20]

Internet-Draft                    JSEP                 February 16, 2012


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

7.2. 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(SDP_OFFER, offer);
   OffererJS:              xmpp = createSessionInitiate(offer);
   OffererJS->AnswererJS:  <jingle action="session-initiate"/>

   OffererJS->OffererUA:   pc.startIce();
   OffererUA->OffererJS:   iceCallback(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(SDP_OFFER, offer);
   AnswererUA->AnswererJS: onaddstream(remoteStream);

   // transport-infos arrive at Answerer
   AnswererJS->AnswererUA: candidates = parseTransportInfo(xmpp);
   AnswererJS->AnswererUA: pc.processIceMessage(candidates);
   AnswererJS->AnswererUA: pc.startIce();
   AnswererUA->AnswererJS: iceCallback(cand, ...)
   AnswererJS:             createTransportInfo(cand);
   AnswererJS->OffererJS:  <jingle action="transport-info"/>

   // transport-infos arrive at Offerer
   OffererJS->OffererUA:  candidates = parseTransportInfo(xmpp);
   OffererJS->OffererUA:  pc.processIceMessage(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(SDP_ANSWER, answer);
   AnswererJS->OffererJS:  <jingle action="session-accept"/>




Uberti                  Expires August 19, 2012                [Page 21]

Internet-Draft                    JSEP                 February 16, 2012


   // 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

7.3. 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(SDP_OFFER, offer);
   OffererJS->AnswererJS:  <jingle action="content-add"/>

   // content-add arrives at Answerer
   AnswererJS:             offer = parseContentAdd(xmpp);
   AnswererJS->AnswererUA: pc.setRemoteDescription(SDP_OFFER, offer);
   AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
   AnswererJS->AnswererUA: pc.setLocalDescription(SDP_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(SDP_ANSWER, answer);


7.4. 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



Uberti                  Expires August 19, 2012                [Page 22]

Internet-Draft                    JSEP                 February 16, 2012


   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(SDP_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(SDP_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(SDP_ANSWER, offer);

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

7.5. 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);
   OffererJS->OffererUA:   pc.startIce();
   OffererUA->OffererJS:   iceCallback(candidate, false);
   OffererJS->OffererUA:   offer = pc.createOffer(null);
   OffererJS->OffererUA:   pc.setLocalDescription(SDP_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(SDP_OFFER, offer);



Uberti                  Expires August 19, 2012                [Page 23]

Internet-Draft                    JSEP                 February 16, 2012


   AnswererUA->AnswererJS: onaddstream(remoteStream);
   AnswererJS->AnswererUA: pc.startIce();
   AnswererUA->OffererUA:  iceCallback(candidate, false);

   // 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(SDP_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

7.6. 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);
   OffererJS->OffererUA:   pc.startIce();
   OffererUA->OffererJS:   iceCallback(candidate, false);
   OffererJS->OffererUA:   offer = pc.createOffer(null);
   OffererJS->OffererUA:   pc.setLocalDescription(SDP_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(SDP_PRANSWER, answer);
   OffererUA->OffererJS:   onaddstream(remoteStream);

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



Uberti                  Expires August 19, 2012                [Page 24]

Internet-Draft                    JSEP                 February 16, 2012


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

8. Example Application

   The following example demonstrates a simple video calling
   application, roughly corresponding to the flow in Example 7.1.

   var signalingChannel = createSignalingChannel();
   var pc = null;
   var hasCandidates = false;

   function start(isCaller) {
     // create a PeerConnection and hook up the IceCallback
     pc = new webkitPeerConnection(
            "", function (candidate, moreToFollow) {
       if (!moreToFollow) {
         hasCandidates = true;
         maybeSignal(isCaller);
       }
     });

     // get the local stream and show it in the local video element
     navigator.webkitGetUserMedia(
           {"audio": true, "video": true}, function (localStream) {
       selfView.src = webkitURL.createObjectURL(localStream);
       pc.addStream(localStream);
       maybeSignal(isCaller);
     }

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

     // if we're the caller, create and install our offer,
     // and start candidate generation
     if (isCaller) {
       offer = pc.createOffer(null);
       pc.setLocalDescription(SDP_OFFER, offer);
       pc.startIce();
     }
   }

   function maybeSignal(isCaller) {
     // only signal once we have a local stream and local candidates



Uberti                  Expires August 19, 2012                [Page 25]

Internet-Draft                    JSEP                 February 16, 2012


     if (localStreams.size() == 0 || !hasCandidates) return;
     if (isCaller) {
       offer = pc.localDescription;
       signalingChannel.send(
           JSON.stringify({ "type": "offer", "sdp": offer }));
     } else {
       // if we're the callee, generate, apply, and send the answer
       answer = pc.createAnswer(pc.remoteDescription, null);
       pc.setLocalDescription(SDP_ANSWER, answer);
       signalingChannel.send(
           JSON.stringify({ "type": "answer", "sdp": answer }));
     }
   }

   signalingChannel.onmessage = function(evt) {
     var msg = JSON.parse(evt.data);
     if (msg.type == "offer") {
       // create the PeerConnection
       start(false);
       // feed the received offer into the PeerConnection and
       // start candidate generation
       pc.setRemoteDescription(PeerConnection.SDP_OFFER, msg.sdp);
       pc.startIce();
     } else if (msg.type == "answer") {
       // feed the answer into the PeerConnection to complete setup
       pc.setRemoteDescription(PeerConnection.SDP_ANSWER, msg.sdp);
     }

9. Security Considerations

   TODO

10. IANA Considerations

   This document requires no actions from IANA.

11. Acknowledgements

   Harald Alvestrand, Dan Burnett, Neil Stratford, Eric Rescorla, and
   Anant Narayanan 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.
   Adam Bergvist provided a code example that served as the basis for
   the example in Section 8.

12. References

12.1. Normative References



Uberti                  Expires August 19, 2012                [Page 26]

Internet-Draft                    JSEP                 February 16, 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.


12.2. 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:
   Real-time Communication Between Browsers", October 2011.

   Available at http://dev.w3.org/2011/webrtc/editor/webrtc.html

Appendix A. Open Issues

   - Determine list of exceptions that can be thrown by each method.
   Leaning toward something like a PCException, a la
   https://developer.mozilla.org/en/IndexedDB/IDBDatabaseException

   - Need callback to indicate that the transport is down, e.g.
   ICE_DISCONNECTED or ondisconnected().

Appendix B. Change log

   02: Updates based on additional feedback: clarified handling of
       createOffer/Answer and setLocal/RemoteDescription; fixed bug in
       sample app.
   01: Updates based on IETF 82.5 feedback: simpler handing of SDP and
       candidates, connect()->startIce(), added more specifics on APIs,
       more examples, full sample application.
   00: Initial version; includes some improvements from W3C mailing list
       feedback.

Authors' Addresses

   Justin Uberti



Uberti                  Expires August 19, 2012                [Page 27]

Internet-Draft                    JSEP                 February 16, 2012


   Google
   5 Cambridge Center
   Cambridge, MA 02142

   Email: justin@uberti.name

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

   Email:  fluffy@cisco.com






































Uberti                  Expires August 19, 2012                [Page 28]


Html markup produced by rfcmarkup 1.109, available from https://tools.ietf.org/tools/rfcmarkup/