Network Working Group J. Uberti Internet-Draft Google Intended status: Standards Track C. Jennings Expires: August 29, 2013 Cisco February 25, 2013 Javascript Session Establishment Protocol draft-ietf-rtcweb-jsep-03 Abstract This document describes the mechanisms for allowing a Javascript application to fully control the signaling plane of a multimedia session via the interface specified in the W3C RTCPeerConnection API, and discusses how this relates to existing signaling protocols. Status of this 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 August 29, 2013. Copyright Notice Copyright (c) 2013 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Uberti & Jennings Expires August 29, 2013 [Page 1] Internet-Draft JSEP February 2013 Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 1.1. General Design of JSEP . . . . . . . . . . . . . . . . . . 4 1.2. Other Approaches Considered . . . . . . . . . . . . . . . 6 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6 3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . . 7 3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 7 3.2. Session Descriptions and State Machine . . . . . . . . . . 7 3.3. Session Description Format . . . . . . . . . . . . . . . . 10 3.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10 3.4.1. ICE Candidate Trickling . . . . . . . . . . . . . . . 10 3.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . . 11 3.5. Interactions With Forking . . . . . . . . . . . . . . . . 11 3.5.1. Sequential Forking . . . . . . . . . . . . . . . . . . 12 3.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . . 12 3.6. Session Rehydration . . . . . . . . . . . . . . . . . . . 13 4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 14 4.1. SDP Requirements . . . . . . . . . . . . . . . . . . . . . 14 4.2. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 15 4.2.1. createOffer . . . . . . . . . . . . . . . . . . . . . 15 4.2.2. createAnswer . . . . . . . . . . . . . . . . . . . . . 16 4.2.3. SessionDescriptionType . . . . . . . . . . . . . . . . 17 4.2.3.1. Use of Provisional Answers . . . . . . . . . . . . 18 4.2.3.2. Rollback . . . . . . . . . . . . . . . . . . . . . 18 4.2.4. setLocalDescription . . . . . . . . . . . . . . . . . 19 4.2.5. setRemoteDescription . . . . . . . . . . . . . . . . . 19 4.2.6. localDescription . . . . . . . . . . . . . . . . . . . 20 4.2.7. remoteDescription . . . . . . . . . . . . . . . . . . 20 4.2.8. updateIce . . . . . . . . . . . . . . . . . . . . . . 20 4.2.9. addIceCandidate . . . . . . . . . . . . . . . . . . . 21 5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . . 21 5.1. Constructing an Offer . . . . . . . . . . . . . . . . . . 21 5.2. Generating an Answer . . . . . . . . . . . . . . . . . . . 21 5.3. Parsing an Offer . . . . . . . . . . . . . . . . . . . . . 21 5.4. Parsing an Answer . . . . . . . . . . . . . . . . . . . . 21 5.5. Applying a Local Description . . . . . . . . . . . . . . . 21 5.6. Applying a Remote Description . . . . . . . . . . . . . . 21 6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 21 7. Security Considerations . . . . . . . . . . . . . . . . . . . 22 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 23 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23 10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 23 10.1. Normative References . . . . . . . . . . . . . . . . . . . 23 10.2. Informative References . . . . . . . . . . . . . . . . . . 24 Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 25 A.1. Example API Flows . . . . . . . . . . . . . . . . . . . . 25 A.1.1. Call using ROAP . . . . . . . . . . . . . . . . . . . 26 Uberti & Jennings Expires August 29, 2013 [Page 2] Internet-Draft JSEP February 2013 A.1.2. Call using XMPP . . . . . . . . . . . . . . . . . . . 26 A.1.3. Adding video to a call, using XMPP . . . . . . . . . . 28 A.1.4. Simultaneous add of video streams, using XMPP . . . . 28 A.1.5. Call using SIP . . . . . . . . . . . . . . . . . . . . 29 A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using SIP . . . . . . . . . . . . . . . . . . . . . . . . . 30 Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 31 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 32 Uberti & Jennings Expires August 29, 2013 [Page 3] Internet-Draft JSEP February 2013 1. Introduction This document describes how the W3C WEBRTC RTCPeerConnection interface[W3C.WD-webrtc-20111027] is used to control the setup, management and teardown of a multimedia session. 1.1. General Design of JSEP The thinking behind WebRTC call setup has been to fully specify and control the media plane, but to leave the signaling plane up to the application as much as possible. The rationale is that different applications may prefer to use different protocols, such as the existing SIP or Jingle call signaling protocols, or something custom to the particular application, perhaps for a novel use case. In this approach, the key information that needs to be exchanged is the multimedia session description, which specifies the necessary transport and media configuration information necessary to establish the media plane. The browser environment also has its own challenges that pose problems for an embedded signaling state machine. One of these is that the user may reload the web page at any time. If the browser is fully in charge of the signaling state, this will result in the loss of the call when this state is wiped by the reload. However, if the state can be stored at the server, and pushed back down to the new page, the call can be resumed with minimal interruption. With these considerations in mind, this document describes the Javascript Session Establishment Protocol (JSEP) that allows for full control of the signaling state machine from Javascript. This mechanism effectively removes the browser almost completely from the core signaling flow; the only interface needed is a way for the application to pass in the local and remote session descriptions negotiated by whatever signaling mechanism is used, and a way to interact with the ICE state machine. In this document, the use of JSEP is described as if it always occurs between two browsers. Note though in many cases it will actually be between a browser and some kind of server, such as a gateway or MCU. This distinction is invisible to the browser; it just follows the instructions it is given via the API. JSEP's handling of session descriptions is simple and straightforward. Whenever an offer/answer exchange is needed, the initiating side creates an offer by calling a createOffer() API. The application optionally modifies that offer, and then uses it to set up its local config via the setLocalDescription() API. The offer is then sent off to the remote side over its preferred signaling Uberti & Jennings Expires August 29, 2013 [Page 4] Internet-Draft JSEP February 2013 mechanism (e.g., WebSockets); upon receipt of that offer, the remote party installs it using the setRemoteDescription() API. When the call is accepted, the callee uses the createAnswer() API to generate an appropriate answer, applies it using setLocalDescription(), and sends the answer back to the initiator over the signaling channel. When the offerer gets that answer, it installs it using setRemoteDescription(), and initial setup is complete. This process can be repeated for additional offer/answer exchanges. Regarding ICE [RFC5245], JSEP decouples the ICE state machine from the overall signaling state machine, as the ICE state machine must remain in the browser, because only the browser has the necessary knowledge of candidates and other transport info. Performing this separation also provides additional flexibility; in protocols that decouple session descriptions from transport, such as Jingle, the transport information can be sent separately; in protocols that don't, such as SIP, the information can be used in the aggregated form. Sending transport information separately can allow for faster ICE and DTLS startup, since the necessary roundtrips can occur while waiting for the remote side to accept the session. Through its abstraction of signaling, the JSEP approach does require the application to be aware of the signaling process. While the application does not need to understand the contents of session descriptions to set up a call, the application must call the right APIs at the right times, convert the session descriptions and ICE information into the defined messages of its chosen signaling protocol, and perform the reverse conversion on the messages it receives from the other side. One way to mitigate this is to provide a Javascript library that hides this complexity from the developer; said library would implement a given signaling protocol along with its state machine and serialization code, presenting a higher level call-oriented interface to the application developer. For example, this library could easily adapt the JSEP API into the API that was proposed for the ROAP signaling protocol [I-D.jennings-rtcweb-signaling], which would perform a ROAP call setup under the covers, interacting with the application only when it needs a signaling message to be sent. In the same fashion, one could also implement other popular signaling protocols, including SIP or Jingle. This allow JSEP to provide greater control for the experienced developer without forcing any additional complexity on the novice developer. Uberti & Jennings Expires August 29, 2013 [Page 5] Internet-Draft JSEP February 2013 1.2. Other Approaches Considered One approach that was considered instead of JSEP was to include a lightweight signaling protocol. Instead of providing session descriptions to the API, the API would produce and consume messages from this protocol. While providing a more high-level API, this put more control of signaling within the browser, forcing the browser to have to understand and handle concepts like signaling glare. In addition, it prevented the application from driving the state machine to a desired state, as is needed in the page reload case. A second approach that was considered but not chosen was to decouple the management of the media control objects from session descriptions, instead offering APIs that would control each component directly. This was rejected based on a feeling that requiring exposure of this level of complexity to the application programmer would not be beneficial; it would result in an API where even a simple example would require a significant amount of code to orchestrate all the needed interactions, as well as creating a large API surface that needed to be agreed upon and documented. In addition, these API points could be called in any order, resulting in a more complex set of interactions with the media subsystem than the JSEP approach, which specifies how session descriptions are to be evaluated and applied. One variation on JSEP that was considered was to keep the basic session description-oriented API, but to move the mechanism for generating offers and answers out of the browser. Instead of providing createOffer/createAnswer methods within the browser, this approach would instead expose a getCapabilities API which would provide the application with the information it needed in order to generate its own session descriptions. This increases the amount of work that the application needs to do; it needs to know how to generate session descriptions from capabilities, and especially how to generate the correct answer from an arbitrary offer and the supported capabilities. While this could certainly be addressed by using a library like the one mentioned above, it basically forces the use of said library even for a simple example. Providing createOffer/createAnswer avoids this problem, but still allows applications to generate their own offers/answers (to a large extent) if they choose, using the description generated by createOffer as an indication of the browser's capabilities. 2. Terminology The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this Uberti & Jennings Expires August 29, 2013 [Page 6] Internet-Draft JSEP February 2013 document are to be interpreted as described in [RFC2119]. 3. Semantics and Syntax 3.1. Signaling Model JSEP does not specify a particular signaling model or state machine, other than the generic need to exchange SDP media descriptions in the fashion described by [RFC3264] (offer/answer) in order for both sides of the session to know how to conduct the session. JSEP provides mechanisms to create offers and answers, as well as to apply them to a session. However, the browser is totally decoupled from the actual mechanism by which these offers and answers are communicated to the remote side, including addressing, retransmission, forking, and glare handling. These issues are left entirely up to the application; the application has complete control over which offers and answers get handed to the browser, and when. +-----------+ +-----------+ | Web App |<--- App-Specific Signaling -->| Web App | +-----------+ +-----------+ ^ ^ | SDP | SDP V V +-----------+ +-----------+ | Browser |<----------- Media ------------>| Browser | +-----------+ +-----------+ Figure 1: JSEP Signaling Model 3.2. Session Descriptions and State Machine In order to establish the media plane, the user agent needs specific parameters to indicate what to transmit to the remote side, as well as how to handle the media that is received. These parameters are determined by the exchange of session descriptions in offers and answers, and there are certain details to this process that must be handled in the JSEP APIs. Whether a session description applies to the local side or the remote side affects the meaning of that description. For example, the list of codecs sent to a remote party indicates what the local side is willing to receive, which, when intersected with the set of codecs the remote side supports, specifies what the remote side should send. However, not all parameters follow this rule; for example, the SRTP parameters [RFC4568] sent to a remote party indicate what the local side will use to encrypt, and thereby what the remote party should Uberti & Jennings Expires August 29, 2013 [Page 7] Internet-Draft JSEP February 2013 expect to receive; the remote party will have to accept these parameters, with no option to choose a different value. In addition, various RFCs put different conditions on the format of offers versus answers. For example, a offer may propose multiple SRTP configurations, but an answer may only contain a single SRTP configuration. Lastly, while the exact media parameters are only known only after a offer and an answer have been exchanged, it is possible for the offerer to receive media after they have sent an offer and before they have received an answer. To properly process incoming media in this case, the offerer's media handler must be aware of the details of the offer before the answer arrives. Therefore, in order to handle session descriptions properly, the user agent needs: 1. To know if a session description pertains to the local or remote side. 2. To know if a session description is an offer or an answer. 3. To allow the offer to be specified independently of the answer. JSEP addresses this by adding both a setLocalDescription and a setRemoteDescription method and having session description objects contain a type field indicating the type of session description being supplied. This satisfies the requirements listed above for both the offerer, who first calls setLocalDescription(sdp [offer]) and then later setRemoteDescription(sdp [answer]), as well as for the answerer, who first calls setRemoteDescription(sdp [offer]) and then later setLocalDescription(sdp [answer]). JSEP also allows for an answer to be treated as provisional by the application. Provisional answers provide a way for an answerer to communicate initial session parameters back to the offerer, in order to allow the session to begin, while allowing a final answer to be specified later. This concept of a final answer is important to the offer/answer model; when such an answer is received, any extra resources allocated by the caller can be released, now that the exact session configuration is known. These "resources" can include things like extra ICE components, TURN candidates, or video decoders. Provisional answers, on the other hand, do no such deallocation results; as a result, multiple dissimilar provisional answers can be received and applied during call setup. In [RFC3264], the constraint at the signaling level is that only one offer can be outstanding for a given session, but from the media stack level, a new offer can be generated at any point. For example, when using SIP for signaling, if one offer is sent, then cancelled using a SIP CANCEL, another offer can be generated even though no Uberti & Jennings Expires August 29, 2013 [Page 8] Internet-Draft JSEP February 2013 answer was received for the first offer. To support this, the JSEP media layer can provide an offer whenever the Javascript application needs one for the signaling. The answerer can send back zero or more provisional answers, and finally end the offer-answer exchange by sending a final answer. The state machine for this is as follows: setRemote(OFFER) setLocal(PRANSWER) /-----\ /-----\ | | | | v | v | +---------------+ | +---------------+ | | |----/ | |----/ | | setLocal(PRANSWER) | | | Remote-Offer |------------------- >| Local-Pranswer| | | | | | | | | +---------------+ +---------------+ ^ | | | | setLocal(ANSWER) | setRemote(OFFER) | | | | V setLocal(ANSWER) | +---------------+ | | | | | | | | Stable |<---------------------------+ | | | | | | +---------------+ setRemote(ANSWER) | ^ | | | | setLocal(OFFER) | setRemote(ANSWER)| | | | V | +---------------+ +---------------+ | | | | | | setRemote(PRANSWER) | | | Local-Offer |------------------- >|Remote-Pranswer| | | | | | |----\ | |----\ +---------------+ | +---------------+ | ^ | ^ | | | | | \-----/ \-----/ setLocal(OFFER) setRemote(PRANSWER) Figure 2: JSEP State Machine Aside from these state transitions, there is no other difference between the handling of provisional ("pranswer") and final ("answer") Uberti & Jennings Expires August 29, 2013 [Page 9] Internet-Draft JSEP February 2013 answers. 3.3. Session Description Format In the WebRTC specification, session descriptions are formatted as SDP messages. While this format is not optimal for manipulation from Javascript, it is widely accepted, and frequently updated with new features. Any alternate encoding of session descriptions would have to keep pace with the changes to SDP, at least until the time that this new encoding eclipsed SDP in popularity. As a result, JSEP currently uses SDP as the internal representation for its session descriptions. However, to simplify Javascript processing, and provide for future flexibility, the SDP syntax is encapsulated within a SessionDescription object, which can be constructed from SDP, and be serialized out to SDP. If future specifications agree on a JSON format for session descriptions, we could easily enable this object to generate and consume that JSON. Other methods may be added to SessionDescription in the future to simplify handling of SessionDescriptions from Javascript. In the meantime, it would be simple to write a Javascript library to perform these manipulations. 3.4. ICE When a new ICE candidate is available, the ICE Agent will notify the application via a callback; these candidates will automatically be added to the local session description. When all candidates have been gathered, the callback will also be invoked to signal that the gathering process is complete. 3.4.1. ICE Candidate Trickling Candidate trickling is a technique through which a caller may incrementally provide candidates to the callee after the initial offer has been dispatched; the semantics of "Trickle ICE" are defined in [I-D.rescorla-mmusic-ice-trickle]. This process allows the callee to begin acting upon the call and setting up the ICE (and perhaps DTLS) connections immediately, without having to wait for the caller to gather all possible candidates. This results in faster call startup in cases where gathering is not performed prior to initating the call. JSEP supports optional candidate trickling by providing APIs that provide control and feedback on the ICE candidate gathering process. Applications that support candidate trickling can send the initial Uberti & Jennings Expires August 29, 2013 [Page 10] Internet-Draft JSEP February 2013 offer immediately and send individual candidates when they get the notified of a new candidate; applications that do not support this feature can simply wait for the indication that gathering is complete, and then create and send their offer, with all the candidates, at this time. Upon receipt of trickled candidates, the receiving application will supply them to its ICE Agent. This triggers the ICE Agent to start using the new remote candidates for connectivity checks. 3.4.1.1. ICE Candidate Format As with session descriptions, the syntax of the IceCandidate object provides some abstraction, but can be easily converted to and from the SDP candidate lines. The candidate lines are the only SDP information that is contained within IceCandidate, as they represent the only information needed that is not present in the initial offer (i.e. for trickle candidates). This information is carried with the same syntax as the "candidate-attribute" field defined for ICE. For example: candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host The IceCandidate object also contains fields to indicate which m= line it should be associated with. The m line can be identified in one of two ways; either by a m-line index, or a MID. The m-line index is a zero-based index, referring to the Nth m-line in the SDP. The MID uses the "media stream identification", as defined in [RFC3388] , to identify the m-line. WebRTC implementations creating an ICE Candidate object MUST populate both of these fields. Implementations receiving an ICE Candidate object SHOULD use the MID if they implement that functionality, or the m-line index, if not. 3.5. Interactions With Forking Some call signaling systems allow various types of forking where an SDP Offer may be provided to more than one device. For example, SIP [RFC3261] defines both a "Parallel Search" and "Sequential Search". Although these are primarily signaling level issues that are outside the scope of JSEP, they do have some impact on the configuration of the media plane which is relevant. When forking happens at the signaling layer, the Javascript application responsible for the signaling needs to make the decisions about what media should be sent or received at any point of time, as well as which remote endpoint it should communicate with; JSEP is used to make sure the media engine can make the RTP and media perform as required by the application. The basic operations that the applications can have the media engine Uberti & Jennings Expires August 29, 2013 [Page 11] Internet-Draft JSEP February 2013 do are: Start exchanging media to a given remote peer, but keep all the resources reserved in the offer. Start exchanging media with a given remote peer, and free any resources in the offer that are not being used. 3.5.1. Sequential Forking Sequential forking involves a call being dispatched to multiple remote callees, where each callee can accept the call, but only one active session ever exists at a time; no mixing of received media is performed. JSEP handles sequential forking well, allowing the application to easily control the policy for selecting the desired remote endpoint. When an answer arrives from one of the callees, the application can choose to apply it either as a provisional answer, leaving open the possibility of using a different answer in the future, or apply it as a final answer, ending the setup flow. In a "first-one-wins" situation, the first answer will be applied as a final answer, and the application will reject any subsequent answers. In SIP parlance, this would be ACK + BYE. In a "last-one-wins" situation, all answers would be applied as provisional answers, and any previous call leg will be terminated. At some point, the application will end the setup process, perhaps with a timer; at this point, the application could reapply the existing remote description as a final answer. 3.5.2. Parallel Forking Parallel forking involves a call being dispatched to multiple remote callees, where each callee can accept the call, and multiple simultaneous active signaling sessions can be established as a result. If multiple callees send media at the same time, the possibilities for handling this are described in Section 3.1 of [RFC3960]. Most SIP devices today only support exchanging media with a single device at a time, and do not try to mix multiple early media audio sources, as that could result in a confusing situation. For example, consider having a European ringback tone mixed together with the North American ringback tone - the resulting sound would not be like either tone, and would confuse the user. If the signaling application wishes to only exchange media with one of the remote endpoints at a time, then from a media engine point of view, this is exactly like the sequential forking case. In the parallel forking case where the Javascript application wishes Uberti & Jennings Expires August 29, 2013 [Page 12] Internet-Draft JSEP February 2013 to simultaneously exchange media with multiple peers, the flow is slightly more complex, but the Javascript application can follow the strategy that [RFC3960] describes using UPDATE. (It is worth noting that use cases where this is the desired behavior are very unusual.) The UPDATE approach allows the signaling to set up a separate media flow for each peer that it wishes to exchange media with. In JSEP, this offer used in the UPDATE would be formed by simply creating a new PeerConnection and making sure that the same local media streams have been added into this new PeerConnection. Then the new PeerConnection object would produce a SDP offer that could be used by the signaling to perform the UPDATE strategy discussed in [RFC3960] . As a result of sharing the media streams, the application will end up with N parallel PeerConnection sessions, each with a local and remote description and their own local and remote addresses. The media flow from these sessions can be managed by specifying SDP direction attributes in the descriptions, or the application can choose to play out the media from all sessions mixed together. Of course, if the application wants to only keep a single session, it can simply terminate the sessions that it no longer needs. 3.6. Session Rehydration In the event that the local application state is reinitialized, either due to a user reload of the page, or a decision within the application to reload itself (perhaps to update to a new version), it is possible to keep an existing session alive, via a process called "rehydration". The explicit goal of rehydration is to carry out this session resumption with no interaction with the remote side other than normal call signaling messages. With rehydration, the current signaling state is persisted somewhere outside of the page, perhaps on the application server, or in browser local storage. The page is then reloaded, the saved signaling state is retrieved, and a new PeerConnection object is created for the session. The previously obtained MediaStreams are re-acquired, and are given the same IDs as the original session; this ensures the IDs in use by the remote side continue to work. Next, a new offer is generated by the new PeerConnection; this offer will have new ICE and possibly new SDES credentials (since the old ICE and SRTP state has been lost). Finally, this offer is used to re-initiate the session with the existing remote endpoint, who simply sees the new offer as an in-call renegotiation, and replies with an answer that can be supplied to setRemoteDescription. ICE processing proceeds as usual, and as soon as connectivity is established, the session will be back up and running again. [OPEN ISSUE: EKR proposed an alternative rehydration approach where Uberti & Jennings Expires August 29, 2013 [Page 13] Internet-Draft JSEP February 2013 the actual internal PeerConnection object in the browser was kept alive for some time after the web page was killed and provided some way for a new page to acquire the old PeerConnection object.] 4. Interface This section details the basic operations that must be present to implement JSEP functionality. The actual API exposed in the W3C API may have somewhat different syntax, but should map easily to these concepts. 4.1. SDP Requirements Note: The text in this section may not represent working group consensus and is put here so that the working group can discuss it and find out how to change it such that it does have consensus. When generating SDP blobs, either for offers or answers, the generated SDP needs to conform to the following specifications. Similarly, in order to properly process received SDP blobs, implementations need to implement the functionality described in the following specifications. This list is derived from [I-D.ietf-rtcweb-rtp-usage]. R-1 [RFC4566] is the base SDP specification and MUST be implemented. R-2 The [RFC5888] grouping framework MUST be implemented for signaling grouping information, and MUST be used to identify m= lines via the a=mid attribute. R-3 [RFC5124] MUST be supported for signaling RTP/SAVPF RTP profile. R-4 [RFC4585] MUST be implemented to signal RTCP based feedback. R-5 [RFC5245] MUST be implemented for signaling the ICE candidate lines corresponding to each media stream. R-6 [RFC5761] MUST be implemented to signal multiplexing of RTP and RTCP. R-7 The SDP atributes of "sendonly", "recvonly", "inactive", and "sendrecv" from [RFC4566] MUST be implemented to signal information about media direction. R-8 [RFC5576] MUST be implemented to signal RTP SSRC values. [OPEN ISSUE; depends on BUNDLE and how we choose to represent multiple media sources] R-9 [RFC5763] MUST be implemented to signal DTLS certificate fingerprints. Uberti & Jennings Expires August 29, 2013 [Page 14] Internet-Draft JSEP February 2013 R-10 [RFC5506] MAY be implemented to signal Reduced-Size RTCP messages. R-11 [RFC3556] with bandwidth modifiers MAY be supported for specifying RTCP bandwidth as a fraction of the media bandwidth, RTCP fraction allocated to the senders and setting maximum media bit-rate boundaries. R-12 [RFC4568] MAY be implemented to signal SDES SRTP keying information. R-13 A TBD-draft MUST be supported, in order to signal associations between RTP objects and W3C MediaStreams and MediaStreamTracks in a standard way. Though there is not yet WG consensus in this area, this TBD-draft is very likely to be [I-D.alvestrand-mmusic-msid]. R-14 A TBD-draft MUST be supported to signal the use or multiplexing RTP somethings on a single UDP port, in order to avoid excessive use of port number resources. Though there is not yet WG consensus in this area, this TBD-draft is very likely to be [I-D.holmberg-mmusic-sdp-bundle-negotiation]. As required by [RFC4566] Section 5.13 JSEP implementations MUST ignore unknown attributes (a=) lines. Example SDP for RTCWeb call flows can be found in [I-D.nandakumar-rtcweb-sdp]. [TODO: since we are starting to specify how to handle SDP in this document, should these call flows be merged into this document, or this link moved to the examples section?] 4.2. Methods 4.2.1. createOffer The createOffer method generates a blob of SDP that contains a [RFC3264] offer with the supported configurations for the session, including descriptions of the local MediaStreams attached to this PeerConnection, the codec/RTP/RTCP options supported by this implementation, and any candidates that have been gathered by the ICE Agent. A constraints parameters may be supplied to provide additional control over the generated offer. This constraints parameter should allow for the following manipulations to be performed: o To indicate support for a media type even if no MediaStreamTracks of that type have been added to the session (e.g., an audio call that wants to receive video.) o To trigger an ICE restart, for the purpose of reestablishing connectivity. Uberti & Jennings Expires August 29, 2013 [Page 15] Internet-Draft JSEP February 2013 o For re-offer cases, to request an offer that contains the full set of supported capabilities, as opposed to just the currently negotiated parameters. In the initial offer, the generated SDP will contain all desired functionality for the session (certain parts that are supported but not desired by default may be omitted); for each SDP line, the generation of the SDP must follow the process defined for generating an initial offer from the document (listed in Section 4.1) that specifies the given SDP line. In the event createOffer is called after the session is established, createOffer will generate an offer to modify the current session based on any changes that have been made to the session, e.g. adding or removing MediaStreams, or requesting an ICE restart. For each existing stream, the generation of each SDP line must follow the process defined for generating an updated offer from the document that specfies the given SDP line. For each new stream, the generation of the SDP must follow the process of generating an initial offer, as mentioned above. If no changes have been made, or for SDP lines that are unaffected by the requested changes, the offer will only contain the parameters negotiated by the last offer-answer exchange. Session descriptions generated by createOffer must be immediately usable by setLocalDescription; if a system has limited resources (e.g. a finite number of decoders), createOffer should return an offer that reflects the current state of the system, so that setLocalDescription will succeed when it attempts to acquire those resources. Because this method may need to inspect the system state to determine the currently available resources, it may be implemented as an async operation. Calling this method may do things such as generate new ICE credentials, but does not result in candidate gathering, or cause media to start or stop flowing. 4.2.2. createAnswer The createAnswer method generates a blob of SDP that contains a [RFC3264] SDP answer with the supported configuration for the session that is compatible with the parameters supplied in the offer. Like createOffer, the returned blob contains descriptions of the local MediaStreams attached to this PeerConnection, the codec/RTP/RTCP options negotiated for this session, and any candidates that have been gathered by the ICE Agent. A constraints parameter may be supplied to provide additional control over the generated answer. Uberti & Jennings Expires August 29, 2013 [Page 16] Internet-Draft JSEP February 2013 As an answer, the generated SDP will contain a specific configuration that specifies how the media plane should be established; for each SDP line, the generation of the SDP must follow the process defined for generating an answer from the document that specifies the given SDP line. Session descriptions generated by createAnswer must be immediately usable by setLocalDescription; like createOffer, the returned description should reflect the current state of the system. Because this method may need to inspect the system state to determine the currently available resources, it may need to be implemented as an async operation. Calling this method may do things such as generate new ICE credentials, but does not trigger candidate gathering or change media state. 4.2.3. SessionDescriptionType Session description objects (RTCSessionDescription) may be of type "offer", "pranswer", and "answer". These types provide information as to how the description parameter should be parsed, and how the media state should be changed. "offer" indicates that a description should be parsed as an offer; said description may include many possible media configurations. A description used as an "offer" may be applied anytime the PeerConnection is in a stable state, or as an update to a previously supplied but unanswered "offer". "pranswer" indicates that a description should be parsed as an answer, but not a final answer, and so should not result in the freeing of allocated resources. It may result in the start of media transmission, if the answer does not specify an inactive media direction. A description used as a "pranswer" may be applied as a response to an "offer", or an update to a previously sent "answer". "answer" indicates that a description should be parsed as an answer, the offer-answer exchange should be considered complete, and any resources (decoders, candidates) that are no longer needed can be released. A description used as an "answer" may be applied as a response to a "offer", or an update to a previously sent "pranswer". The only difference between a provisional and final answer is that the final answer results in the freeing of any unused resources that were allocated as a result of the offer. As such, the application can use some discretion on whether an answer should be applied as provisional or final, and can change the type of the session Uberti & Jennings Expires August 29, 2013 [Page 17] Internet-Draft JSEP February 2013 description as needed. For example, in a serial forking scenario, an application may receive multiple "final" answers, one from each remote endpoint. The application could choose to accept the initial answers as provisional answers, and only apply an answer as final when it receives one that meets its criteria (e.g. a live user instead of voicemail). 4.2.3.1. Use of Provisional Answers Most web applications will not need to create answers using the "pranswer" type. The preferred handling for a web application would be to create and send an "inactive" answer more or less immediately after receiving the offer, instead of waiting for a human user to physically answer the call. Later, when the human input is received, the application can create a new "sendrecv" offer to update the previous offer/answer pair and start the media flow. This approach is preferred because it minimizes the amount of time that the offer- answer exchange is left open, in addition to avoiding media clipping by ensuring the transport is ready to go by the time the call is phyiscally answered. However, some applications may not be able to do this, particularly ones that are attempting to gateway to other signaling protocols. In these cases, "pranswer" can still allow the application to warm up the transport. Consider a typical web application that will set up a data channel, an audio channel, and a video channel. When an endpoint receives an offer with these channels, it could send an answer accepting the data channel for two-way data, and accepting the audio and video tracks as inactive or receive-only. It could then ask the user to accept the call, acquire the local media streams, and send a new offer to the remote side moving the audio and video to be two-way media. By the time the human has accepted the call and sent the new offer, it is likely that the ICE and DTLS handshaking for all the channels will already be set up. 4.2.3.2. Rollback In certain situations it may be desirable to "undo" a change made to setLocalDescription or setRemoteDescription. Consider a case where a call is ongoing, and one side wants to change some of the session parameters; that side generates an updated offer and then calls setLocalDescription. However, the remote side, either before or after setRemoteDescription, decides it does not want to accept the new parameters, and sends a reject message back to the offerer. Now, the offerer, and possibly the answerer as well, need to return to a stable state and the previous local/remote description. To support this, we introduce the concept of "rollback". Uberti & Jennings Expires August 29, 2013 [Page 18] Internet-Draft JSEP February 2013 A rollback returns the state machine to its previous state, and the local or remote description to its previous value. Any resources or candidates that were allocated by the new local description are discarded; any media that is received will be processed according to the previous session description. A rollback is performed by supplying a session description of type "rollback" to either setLocalDescription or setRemoteDescription, depending on which needs to be rolled back (i.e. if the new offer was supplied to setLocalDescription, the rollback should be done on setLocalDescription as well.) 4.2.4. setLocalDescription The setLocalDescription method instructs the PeerConnection to apply the supplied SDP blob as its local configuration. The type field indicates whether the blob should be processed as an offer, provisional answer, or final answer; offers and answers are checked differently, using the various rules that exist for each SDP line. This API changes the local media state; among other things, it sets up local resources for receiving and decoding media. In order to successfully handle scenarios where the application wants to offer to change from one media format to a different, incompatible format, the PeerConnection must be able to simultaneously support use of both the old and new local descriptions (e.g. support codecs that exist in both descriptions) until a final answer is received, at which point the PeerConnection can fully adopt the new local description, or roll back to the old description if the remote side denied the change. This API indirectly controls the candidate gathering process. When a local description is supplied, and the number of transports currently in use does not match the number of transports needed by the local description, the PeerConnection will create transports as needed and begin gathering candidates for them. If setRemoteDescription was previous called with an offer, and setLocalDescription is called with an answer (provisional or final), and the media directions are compatible, and media are available to send, this will result in the starting of media transmission. 4.2.5. setRemoteDescription The setRemoteDescription method instructs the PeerConnection to apply the supplied SDP blob as the desired remote configuration. As in setLocalDescription, the type field of the indicates how the blob should be processed. Uberti & Jennings Expires August 29, 2013 [Page 19] Internet-Draft JSEP February 2013 This API changes the local media state; among other things, it sets up local resources for sending and encoding media. If setRemoteDescription was previously called with an offer, and setLocalDescription is called with an answer (provisional or final), and the media directions are compatible, and media are available to send, this will result in the starting of media transmission. 4.2.6. localDescription The localDescription method returns a copy of the current local configuration, i.e. what was most recently passed to setLocalDescription, plus any local candidates that have been generated by the ICE Agent. TODO: Do we need to expose accessors for both the current and proposed local description? A null object will be returned if the local description has not yet been established, or if the PeerConnection has been closed. 4.2.7. remoteDescription The remoteDescription method returns a copy of the current remote configuration, i.e. what was most recently passed to setRemoteDescription, plus any remote candidates that have been supplied via processIceMessage. TODO: Do we need to expose accessors for both the current and proposed remote description? A null object will be returned if the remote description has not yet been established, or if the PeerConnection has been closed. 4.2.8. updateIce The updateIce method allows the configuration of the ICE Agent to be changed during the session, primarily for changing which types of local candidates are provided to the application and used for connectivity checks. A callee may initially configure the ICE Agent to use only relay candidates, to avoid leaking location information, but update this configuration to use all candidates once the call is accepted. Regardless of the configuration, the gathering process collects all available candidates, but excluded candidates will not be surfaced in onicecandidate callback or used for connectivity checks. Uberti & Jennings Expires August 29, 2013 [Page 20] Internet-Draft JSEP February 2013 This call may result in a change to the state of the ICE Agent, and may result in a change to media state if it results in connectivity being established. 4.2.9. addIceCandidate The addIceCandidate method provides a remote candidate to the ICE Agent, which, if parsed successfully, will be added to the remote description according to the rules defined for Trickle ICE. Connectivity checks will be sent to the new candidate. This call will result in a change to the state of the ICE Agent, and may result in a change to media state if it results in connectivity being established. 5. SDP Interaction Procedures This section describes the specific procedures to be followed when creating and parsing SDP objects. [Work In Progress] 5.1. Constructing an Offer 5.2. Generating an Answer 5.3. Parsing an Offer 5.4. Parsing an Answer 5.5. Applying a Local Description 5.6. Applying a Remote Description 6. Configurable SDP Parameters Note: This section is still very early and is likely to significantly change as we get a better understanding of a) the use cases for this b) the implications at the protocol level c) feedback from implementors on what they can do. The following elements of the SDP media description MUST NOT be changed between the createOffer and the setLocalDescription, since they reflect transport attributes that are solely under browser control, and the browser MUST NOT honor an attempt to change them: Uberti & Jennings Expires August 29, 2013 [Page 21] Internet-Draft JSEP February 2013 o The number, type and port number of m-lines. o The generated ICE credentials (a=ice-ufrag and a=ice-pwd). o The set of ICE candidates and their parameters (a=candidate). The following modifications, if done by the browser to a description between createOffer/createAnswer and the setLocalDescription, MUST be honored by the browser: o Remove or reorder codecs (m=) The following parameters may be controlled by constraints passed into createOffer/createAnswer. As an open issue, these changes may also be be performed by manipulating the SDP returned from createOffer/ createAnswer, as indicated above, as long as the capabilities of the endpoint are not exceeded (e.g. asking for a resolution greater than what the endpoint can encode): o disable BUNDLE (a=group) o disable RTCP mux (a=rtcp-mux) o change send resolution or framerate o change desired recv resolution or framerate o change maximum total bandwidth (b=) [OPEN ISSUE: need to clarify if this is CT or AS - see section 5.8 of RFC4566] o remove desired AVPF mechanisms (a=rtcp-fb) o remove RTP header extensions (a=extmap) o change media send/recv state (a=sendonly/recvonly/inactive) For example, an application could implement call hold by adding an a=inactive attribute to its local description, and then applying and signaling that description. The application can also modify the SDP to reduce the capabilities in the offer it sends to the far side in any way the application sees fit, as long as it is a valid SDP offer and specifies a subset of what the browser is expecting to do. As always, the application is solely responsible for what it sends to the other party, and all incoming SDP will be processed by the browser to the extent of its capabilities. It is an error to assume that all SDP is well-formed; however, one should be able to assume that any implementation of this specification will be able to process, as a remote offer or answer, unmodified SDP coming from any other implementation of this specification. 7. Security Considerations The intent of the WebRTC protocol suite is to provide an environment Uberti & Jennings Expires August 29, 2013 [Page 22] Internet-Draft JSEP February 2013 that is securable by default: all media is encrypted, keys are exchanged in a secure fashion, and the Javascript API includes functions that can be used to verify the identity of communication partners. 8. IANA Considerations This document requires no actions from IANA. 9. Acknowledgements Significant text incorporated in the draft as well and review was provided by Harald Alvestrand and Suhas Nandakumar. Dan Burnett, Neil Stratford, Eric Rescorla, Anant Narayanan, Andrew Hutton, Richard Ejzak, and Adam Bergkvist all provided valuable feedback on this proposal. Matthew Kaufman provided the observation that keeping state out of the browser allows a call to continue even if the page is reloaded. 10. References 10.1. Normative References [I-D.rescorla-mmusic-ice-trickle] Rescorla, E., Uberti, J., and E. Ivov, "Trickle ICE: Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) Protocol", draft-rescorla-mmusic-ice-trickle-00 (work in progress), October 2012. [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002. [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, June 2002. [RFC3388] Camarillo, G., Eriksson, G., Holler, J., and H. Schulzrinne, "Grouping of Media Lines in the Session Description Protocol (SDP)", RFC 3388, December 2002. Uberti & Jennings Expires August 29, 2013 [Page 23] Internet-Draft JSEP February 2013 [RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)", RFC 3960, December 2004. [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006. [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 2006. [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)", RFC 5124, February 2008. [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols", RFC 5245, April 2010. [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and Control Packets on a Single Port", RFC 5761, April 2010. [RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description Protocol (SDP) Grouping Framework", RFC 5888, June 2010. 10.2. Informative References [I-D.alvestrand-mmusic-msid] Alvestrand, H., "Cross Session Stream Identification in the Session Description Protocol", draft-alvestrand-mmusic-msid-01 (work in progress), October 2012. [I-D.holmberg-mmusic-sdp-bundle-negotiation] Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation Using Session Description Protocol (SDP) Port Numbers", draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in progress), October 2011. [I-D.ietf-rtcweb-rtp-usage] Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time Communication (WebRTC): Media Transport and Use of RTP", draft-ietf-rtcweb-rtp-usage-04 (work in progress), July 2012. [I-D.jennings-rtcweb-signaling] Uberti & Jennings Expires August 29, 2013 [Page 24] Internet-Draft JSEP February 2013 Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/ Answer Protocol (ROAP)", draft-jennings-rtcweb-signaling-01 (work in progress), October 2011. [I-D.nandakumar-rtcweb-sdp] Nandakumar, S. and C. Jennings, "SDP for the WebRTC", draft-nandakumar-rtcweb-sdp-00 (work in progress), October 2012. [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 3556, July 2003. [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session Description Protocol (SDP) Security Descriptions for Media Streams", RFC 4568, July 2006. [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences", RFC 5506, April 2009. [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific Media Attributes in the Session Description Protocol (SDP)", RFC 5576, June 2009. [RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework for Establishing a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)", RFC 5763, May 2010. [W3C.WD-webrtc-20111027] Bergkvist, A., Burnett, D., Narayanan, A., and C. Jennings, "WebRTC 1.0: Real-time Communication Between Browsers", World Wide Web Consortium WD WD-webrtc- 20111027, October 2011, . Appendix A. JSEP Implementation Examples A.1. Example API Flows Below are several sample flows for the new PeerConnection and library APIs, demonstrating when the various APIs are called in different situations and with various transport protocols. For clarity and simplicity, the createOffer/createAnswer calls are assumed to be synchronous in these examples, whereas the actual APIs are async. Uberti & Jennings Expires August 29, 2013 [Page 25] Internet-Draft JSEP February 2013 A.1.1. Call using ROAP This example demonstrates a ROAP call, without the use of trickle candidates. // Call is initiated toward Answerer OffererJS->OffererUA: pc = new PeerConnection(); OffererJS->OffererUA: pc.addStream(localStream, null); OffererUA->OffererJS: iceCallback(candidate); OffererJS->OffererUA: offer = pc.createOffer(null); OffererJS->OffererUA: pc.setLocalDescription("offer", offer); OffererJS->AnswererJS: {"type":"OFFER", "sdp":offer } // OFFER arrives at Answerer AnswererJS->AnswererUA: pc = new PeerConnection(); AnswererJS->AnswererUA: pc.setRemoteDescription("offer", msg.sdp); AnswererUA->AnswererJS: onaddstream(remoteStream); AnswererUA->OffererUA: iceCallback(candidate); // Answerer accepts call AnswererJS->AnswererUA: pc.addStream(localStream, null); AnswererJS->AnswererUA: answer = pc.createAnswer(msg.sdp, null); AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer); AnswererJS->OffererJS: {"type":"ANSWER","sdp":answer } // ANSWER arrives at Offerer OffererJS->OffererUA: pc.setRemoteDescription("answer", answer); OffererUA->OffererJS: onaddstream(remoteStream); // ICE Completes (at Answerer) AnswererUA->OffererUA: Media // ICE Completes (at Offerer) OffererJS->AnswererJS: {"type":"OK" } OffererUA->AnswererUA: Media A.1.2. Call using XMPP This example demonstrates an XMPP call, making use of trickle candidates. Uberti & Jennings Expires August 29, 2013 [Page 26] Internet-Draft JSEP February 2013 // 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: OffererJS->OffererUA: pc.startIce(); OffererUA->OffererJS: onicecandidate(cand); OffererJS: createTransportInfo(cand); OffererJS->AnswererJS: // 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: // transport-infos arrive at Offerer OffererJS->OffererUA: candidates = parseTransportInfo(xmpp); OffererJS->OffererUA: pc.addIceCandidate(candidates); // Answerer accepts call AnswererJS->AnswererUA: pc.addStream(localStream, null); AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null); AnswererJS: xmpp = createSessionAccept(answer); AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer); AnswererJS->OffererJS: // session-accept arrives at Offerer OffererJS: answer = parseSessionAccept(xmpp); OffererJS->OffererUA: pc.setRemoteDescription("answer", answer); OffererUA->OffererJS: onaddstream(remoteStream); // ICE Completes (at Answerer) AnswererUA->OffererUA: Media // ICE Completes (at Offerer) OffererUA->AnswererUA: Media Uberti & Jennings Expires August 29, 2013 [Page 27] Internet-Draft JSEP February 2013 A.1.3. Adding video to a call, using XMPP This example demonstrates an XMPP call, where the XMPP content-add mechanism is used to add video media to an existing session. For simplicity, candidate exchange is not shown. Note that the offerer for the change to the session may be different than the original call offerer. // Offerer adds video stream OffererJS->OffererUA: pc.addStream(videoStream) OffererJS->OffererUA: offer = pc.createOffer(null); OffererJS: xmpp = createContentAdd(offer); OffererJS->OffererUA: pc.setLocalDescription("offer", offer); OffererJS->AnswererJS: // 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: // content-accept arrives at Offerer OffererJS: answer = parseContentAccept(xmpp); OffererJS->OffererUA: pc.setRemoteDescription("answer", answer); A.1.4. Simultaneous add of video streams, using XMPP This example demonstrates an XMPP call, where new video sources are added at the same time to a call that already has video; since adding these sources only affects one side of the call, there is no conflict. The XMPP description-info mechanism is used to indicate the new sources to the remote side. Uberti & Jennings Expires August 29, 2013 [Page 28] Internet-Draft JSEP February 2013 // 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: AnswererJS->AnswererUA: pc.addStream(answererVideoStream2) AnswererJS->AnswererUA: offer = pc.createOffer(null); AnswererJS: xmpp = createDescriptionInfo(offer); AnswererJS->AnswererUA: pc.setLocalDescription("offer", offer); AnswererJS->OffererJS: // description-info arrives at "Answerer", and is acked AnswererJS: offer = parseDescriptionInfo(xmpp); AnswererJS->OffererJS: // ack // description-info arrives at Offerer, and is acked OffererJS: offer = parseDescriptionInfo(xmpp); OffererJS->AnswererJS: // ack // ack arrives at Offerer; remote offer is used as an answer OffererJS->OffererUA: pc.setRemoteDescription("answer", offer); // ack arrives at "Answerer"; remote offer is used as an answer AnswererJS->AnswererUA: pc.setRemoteDescription("answer", offer); A.1.5. Call using SIP This example demonstrates a simple SIP call (e.g. where the client talks to a SIP proxy over WebSockets). Uberti & Jennings Expires August 29, 2013 [Page 29] Internet-Draft JSEP February 2013 // Call is initiated toward Answerer OffererJS->OffererUA: pc = new PeerConnection(); OffererJS->OffererUA: pc.addStream(localStream, null); OffererUA->OffererJS: onicecandidate(candidate); OffererJS->OffererUA: offer = pc.createOffer(null); OffererJS->OffererUA: pc.setLocalDescription("offer", offer); OffererJS: sip = createInvite(offer); OffererJS->AnswererJS: SIP INVITE w/ SDP // INVITE arrives at Answerer AnswererJS->AnswererUA: pc = new PeerConnection(); AnswererJS: offer = parseInvite(sip); AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer); AnswererUA->AnswererJS: onaddstream(remoteStream); AnswererUA->OffererUA: onicecandidate(candidate); // Answerer accepts call AnswererJS->AnswererUA: pc.addStream(localStream, null); AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null); AnswererJS: sip = createResponse(200, answer); AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer); AnswererJS->OffererJS: 200 OK w/ SDP // 200 OK arrives at Offerer OffererJS: answer = parseResponse(sip); OffererJS->OffererUA: pc.setRemoteDescription("answer", answer); OffererUA->OffererJS: onaddstream(remoteStream); OffererJS->AnswererJS: ACK // ICE Completes (at Answerer) AnswererUA->OffererUA: Media // ICE Completes (at Offerer) OffererUA->AnswererUA: Media A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using SIP This example demonstrates how early media could be handled; for simplicity, only the offerer side of the call is shown. Uberti & Jennings Expires August 29, 2013 [Page 30] Internet-Draft JSEP February 2013 // Call is initiated toward Answerer OffererJS->OffererUA: pc = new PeerConnection(); OffererJS->OffererUA: pc.addStream(localStream, null); OffererUA->OffererJS: onicecandidate(candidate); OffererJS->OffererUA: offer = pc.createOffer(null); OffererJS->OffererUA: pc.setLocalDescription("offer", offer); OffererJS: sip = createInvite(offer); OffererJS->AnswererJS: SIP INVITE w/ SDP // 180 Ringing is received by offerer, w/ SDP OffererJS: answer = parseResponse(sip); OffererJS->OffererUA: pc.setRemoteDescription("pranswer", answer); OffererUA->OffererJS: onaddstream(remoteStream); // ICE Completes (at Offerer) OffererUA->AnswererUA: Media // 200 OK arrives at Offerer OffererJS: answer = parseResponse(sip); OffererJS->OffererUA: pc.setRemoteDescription("answer", answer); OffererJS->AnswererJS: ACK Appendix B. Change log Changes in draft-03: o Added text describing relationship to W3C specification Changes in draft -02: o Converted from nroff o Removed comparisons to old approaches abandoned by the working group o Removed stuff that has moved to W3C specificaiton o Align SDP handling with W3C draft o Clarified section on forking. Changes in draft -01: o Added diagrams for architecture and state machine. o Added sections on forking and rehydration. o Clarified meaning of "pranswer" and "answer". o Reworked how ICE restarts and media directions are controlled. o Added list of parameters that can be changed in a description. o Updated suggested API and examples to match latest thinking. o Suggested API and examples have been moved to an appendix. Changes in draft -00: Uberti & Jennings Expires August 29, 2013 [Page 31] Internet-Draft JSEP February 2013 o Migrated from draft-uberti-rtcweb-jsep-02. Authors' Addresses Justin Uberti Google 747 6th Ave S Kirkland, WA 98033 USA Email: justin@uberti.name Cullen Jennings Cisco 170 West Tasman Drive San Jose, CA 95134 USA Email: fluffy@iii.ca Uberti & Jennings Expires August 29, 2013 [Page 32]