--- 1/draft-ietf-rtcweb-jsep-01.txt 2012-10-23 02:14:25.233342966 +0200 +++ 2/draft-ietf-rtcweb-jsep-02.txt 2012-10-23 02:14:25.281342098 +0200 @@ -1,597 +1,605 @@ Network Working Group J. Uberti Internet-Draft Google Intended status: Standards Track C. Jennings -Expires: December 6, 2012 Cisco Systems, Inc. - June 4, 2012 +Expires: April 25, 2013 Cisco + October 22, 2012 Javascript Session Establishment Protocol - draft-ietf-rtcweb-jsep-01 + draft-ietf-rtcweb-jsep-02 Abstract This document proposes a mechanism for allowing a Javascript application to fully control the signaling plane of a multimedia session, and discusses how this would work with existing signaling protocols. - This document is an input document for discussion. It should be - discussed in the RTCWEB WG list, rtcweb@ietf.org. - Status of this Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." - This Internet-Draft will expire on July 26, 2012. + This Internet-Draft will expire on April 25, 2013. Copyright Notice Copyright (c) 2012 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Table of Contents - 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 - 1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . . 5 - 2. JSEP Approach . . . . . . . . . . . . . . . . . . . . . . . . . 5 - 3. Other Approaches Considered . . . . . . . . . . . . . . . . . . 6 + 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 + 2. Other Approaches Considered . . . . . . . . . . . . . . . . . 5 + 3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6 4. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . . 7 - 4.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . . 7 + 4.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 7 4.2. Session Descriptions and State Machine . . . . . . . . . . 7 4.3. Session Description Format . . . . . . . . . . . . . . . . 9 - 4.4. Separation of Signaling and ICE State Machines . . . . . . 10 - 4.5. ICE Candidate Trickling . . . . . . . . . . . . . . . . . . 10 - 4.6. ICE Candidate Format . . . . . . . . . . . . . . . . . . . 11 - 4.7. Interactions With Forking . . . . . . . . . . . . . . . . . 11 - 4.7.1. Serial Forking . . . . . . . . . . . . . . . . . . . . 11 - 4.7.2. Parallel Forking . . . . . . . . . . . . . . . . . . . 12 - 4.8. Session Rehydration . . . . . . . . . . . . . . . . . . . . 12 - 5. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . 13 - 5.1. Methods . . . . . . . . . . . . . . . . . . . . . . . . . . 13 - 5.1.1. createOffer . . . . . . . . . . . . . . . . . . . . . . 13 - 5.1.2. createAnswer . . . . . . . . . . . . . . . . . . . . . 14 - 5.1.3. SessionDescriptionType . . . . . . . . . . . . . . . . 14 - 5.1.4. setLocalDescription . . . . . . . . . . . . . . . . . . 15 - 5.1.5. setRemoteDescription . . . . . . . . . . . . . . . . . 15 - 5.1.6. localDescription . . . . . . . . . . . . . . . . . . . 16 - 5.1.7. remoteDescription . . . . . . . . . . . . . . . . . . . 16 - 5.1.8. updateIce . . . . . . . . . . . . . . . . . . . . . . . 16 - 5.1.9. addIceCandidate . . . . . . . . . . . . . . . . . . . . 17 - 5.2. Configurable SDP Parameters . . . . . . . . . . . . . . . . 17 - 6. Media Setup Overview . . . . . . . . . . . . . . . . . . . . . 17 - 6.1. Initiating the Session . . . . . . . . . . . . . . . . . . 18 - 6.1.1. Generating An Offer . . . . . . . . . . . . . . . . . . 18 - 6.1.2. Applying the Offer . . . . . . . . . . . . . . . . . . 18 - 6.1.3. Handling ICE Callbacks . . . . . . . . . . . . . . . . 18 - 6.1.4. Serializing the Offer and Candidates . . . . . . . . . 19 - 6.2. Receiving the Session . . . . . . . . . . . . . . . . . . . 19 - 6.2.1. Receiving the Offer . . . . . . . . . . . . . . . . . . 19 - 6.2.2. Handling ICE Messages . . . . . . . . . . . . . . . . . 19 - 6.2.3. Generating the Answer . . . . . . . . . . . . . . . . . 20 - 6.2.4. Applying the Answer . . . . . . . . . . . . . . . . . . 20 - 6.2.5. Serializing the Answer . . . . . . . . . . . . . . . . 20 - 6.3. Completing the Session . . . . . . . . . . . . . . . . . . 20 - 6.3.1. Receiving the Answer . . . . . . . . . . . . . . . . . 20 - - 6.4. Updates to the Session . . . . . . . . . . . . . . . . . . 20 - 7. Security Considerations . . . . . . . . . . . . . . . . . . . . 21 - 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 21 - 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 21 - 10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 21 - 10.1. Normative References . . . . . . . . . . . . . . . . . . . 21 - 10.2. Informative References . . . . . . . . . . . . . . . . . . 21 - Appendix A. JSEP Implementation Examples . . . . . . . . . . . . . 22 - A.1. Example API . . . . . . . . . . . . . . . . . . . . . . . . 22 - A.2. Example API Flows . . . . . . . . . . . . . . . . . . . . . 23 - A.2.1. Call using ROAP . . . . . . . . . . . . . . . . . . . . 23 - A.2.2 Call using XMPP . . . . . . . . . . . . . . . . . . . . 24 - A.2.3. Adding video to a call, using XMPP . . . . . . . . . . 25 - A.2.4. Simultaneous add of video streams, using XMPP . . . . . 26 - A.2.5. Call using SIP . . . . . . . . . . . . . . . . . . . . 27 - A.2.6. Handling early media (e.g. 1-800-FEDEX), using SIP . . 28 - A.3. Full Example Application . . . . . . . . . . . . . . . . . 28 - Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . . 30 - Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 30 + 4.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10 + 4.4.1. ICE Candidate Trickling . . . . . . . . . . . . . . . 10 + 4.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . . 10 + 4.5. Interactions With Forking . . . . . . . . . . . . . . . . 11 + 4.5.1. Sequential Forking . . . . . . . . . . . . . . . . . . 11 + 4.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . . 12 + 4.6. Session Rehydration . . . . . . . . . . . . . . . . . . . 13 + 5. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 14 + 5.1. SDP Requirements . . . . . . . . . . . . . . . . . . . . . 14 + 5.2. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 15 + 5.2.1. createOffer . . . . . . . . . . . . . . . . . . . . . 15 + 5.2.2. createAnswer . . . . . . . . . . . . . . . . . . . . . 15 + 5.2.3. SessionDescriptionType . . . . . . . . . . . . . . . . 16 + 5.2.3.1. Creating Answers . . . . . . . . . . . . . . . . . 17 + 5.2.4. setLocalDescription . . . . . . . . . . . . . . . . . 17 + 5.2.5. setRemoteDescription . . . . . . . . . . . . . . . . . 18 + 5.2.6. localDescription . . . . . . . . . . . . . . . . . . . 18 + 5.2.7. remoteDescription . . . . . . . . . . . . . . . . . . 18 + 5.2.8. updateIce . . . . . . . . . . . . . . . . . . . . . . 18 + 5.2.9. addIceCandidate . . . . . . . . . . . . . . . . . . . 19 + 6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 20 + 7. Security Considerations . . . . . . . . . . . . . . . . . . . 21 + 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 22 + 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23 + 10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 24 + 10.1. Normative References . . . . . . . . . . . . . . . . . . . 24 + 10.2. Informative References . . . . . . . . . . . . . . . . . . 24 + Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 26 + A.1. Example API Flows . . . . . . . . . . . . . . . . . . . . 26 + A.1.1. Call using ROAP . . . . . . . . . . . . . . . . . . . 26 + A.1.2. Call using XMPP . . . . . . . . . . . . . . . . . . . 27 + A.1.3. Adding video to a call, using XMPP . . . . . . . . . . 28 + A.1.4. Simultaneous add of video streams, using XMPP . . . . 28 + A.1.5. Call using SIP . . . . . . . . . . . . . . . . . . . . 29 + A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using + SIP . . . . . . . . . . . . . . . . . . . . . . . . . 30 + Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 32 + Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 33 1. Introduction The thinking behind WebRTC call setup has been to fully specify and control the media plane, but to leave the signaling plane up to the application as much as possible. The rationale is that different applications may prefer to use different protocols, such as the existing SIP or Jingle call signaling protocols, or something custom to the particular application, perhaps for a novel use case. In this approach, the key information that needs to be exchanged is the multimedia session description, which specifies the necessary transport and media configuration information necessary to establish the media plane. - The original spec for WebRTC attempted to implement this protocol- - agnostic signaling by providing a mechanism to exchange session - descriptions in the form of SDP blobs. Upon starting a session, the - browser would generate a SDP blob, which would be passed to the - application for transport over its preferred signaling protocol. On - the remote side, this blob would be passed into the browser from the - application, and the browser would then generate a blob of its own in - response. Upon transmission back to the initiator, this blob would be - plugged into their browser, and the handshake would be complete. - - Experimentation with this mechanism turned up several shortcomings, - which generally stemmed from there being insufficient context at the - browser to fully determine the meaning of a SDP blob. For example, - determining whether a blob is an offer or an answer, or - differentiating a new offer from a retransmit. - - The ROAP proposal, specified in [I-D.draft-jennings-rtcweb-signaling- - 01], attempted to resolve these issues by providing additional - structure in the messaging - in essence, to create a generic - signaling protocol that specifies how the browser signaling state - machine should operate. However, even though the protocol is - abstracted, the state machine forces a least-common-denominator - approach on the signaling interactions. For example, in Jingle, the - call initiator can provide additional ICE candidates even after the - initial offer has been sent, which allows the offer to be sent - immediately for quicker call startup. However, in the browser state - machine, there is no notion of sending an updated offer before the - initial offer has been responded to, rendering this functionality - impossible. - - While specific concerns like this could be addressed by modifying the - generic protocol, others would likely be discovered later. The main - reason this mechanism is inflexible is because it embeds a signaling - state machine within the browser. Since the browser generates the - session descriptions on its own, and fully controls the possible - states and advancement of the signaling state machine, modification - of the session descriptions or use of alternate state machines - becomes difficult or impossible. - The browser environment also has its own challenges that cause problems for an embedded signaling state machine. One of these is that the user may reload the web page at any time. If this happens, and the state machine is being run at a server, the server can simply push the current state back down to the page and resume the call where it left off. - If instead the state machine is run at the browser end, and is - instantiated within, for example, the PeerConnection object, that - state machine will be reinitialized when the page is reloaded and the - JavaScript re-executed. This actually complicates the design of any - interoperability service, as all cases where an offer or answer has - already been generated but is now "forgotten" must now be handled by - trying to move the client state machine forward to the same state it - had been in previously in order to match what has already been - delivered to and/or answered by the far side, or handled by ensuring - that aborts are cleanly handled from every state and the negotiation - rapidly restarted. - -1.1. Terminology - - The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", - "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this - document are to be interpreted as described in RFC 2119 [RFC2119]. - -2. JSEP Approach - - To resolve the issues mentioned above, this document proposes the - Javascript Session Establishment Protocol (JSEP) that pulls the - signaling state machine out of the browser and into Javascript. This - mechanism effectively removes the browser almost completely from the - core signaling flow; the only interface needed is a way for the - application to pass in the local and remote session descriptions - negotiated by whatever signaling mechanism is used, and a way to - interact with the ICE state machine. + This document describes the Javascript Session Establishment Protocol + (JSEP) that pulls the signaling state machine out of the browser and + into Javascript. This mechanism effectively removes the browser + almost completely from the core signaling flow; the only interface + needed is a way for the application to pass in the local and remote + session descriptions negotiated by whatever signaling mechanism is + used, and a way to interact with the ICE state machine. JSEP's handling of session descriptions is simple and straightforward. Whenever an offer/answer exchange is needed, the initiating side creates an offer by calling a createOffer() API. The - application 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. + application optionally modifies that offer, and then uses it to set + up its local config via the setLocalDescription() API. The offer is + then sent off to the remote side over its preferred signaling + mechanism (e.g., WebSockets); upon receipt of that offer, the remote + party installs it using the setRemoteDescription() API. - When the call is accepted, the callee uses a createAnswer() API to + When the call is accepted, the callee uses the createAnswer() API to generate an appropriate answer, applies it using setLocalDescription(), and sends the answer back to the initiator over the signaling channel. When the offerer gets that answer, it installs it using setRemoteDescription(), and initial setup is complete. This process can be repeated for additional offer/answer exchanges. Regarding ICE, JSEP decouples the ICE state machine from the overall signaling state machine, as the ICE state machine must remain in the - browser, since only the browser has the necessary knowledge of - candidates and other transport info. Performing this separation it + 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 easily aggregated and recombined. Sending + SIP, the information can be used in the aggregated form. Sending transport information separately can allow for faster ICE and DTLS startup, since the necessary roundtrips can occur while waiting for the remote side to accept the session. - The JSEP approach does come with a minor downside. As the application - now is responsible for driving the signaling state machine, slightly - more application code is necessary to perform call setup; the - application must call the right APIs at the right times, and convert - the session descriptions and ICE information into the defined - messages of its chosen signaling protocol, instead of simply + The JSEP approach does come with a minor downside. As the + application now is responsible for driving the signaling state + machine, slightly more application code is necessary to perform call + setup; the application must call the right APIs at the right times, + and convert the session descriptions and ICE information into the + defined messages of its chosen signaling protocol, instead of simply forwarding the messages emitted from the browser. One way to mitigate this is to provide a Javascript library that hides this complexity from the developer, which would implement the state machine and serialization of the desired signaling protocol. For example, this library could convert easily adapt the JSEP API - into the exact ROAP API, thereby implementing the ROAP signaling - protocol. Such a library could of course also implement other popular - signaling protocols, including SIP or Jingle. In this fashion we can - enable greater control for the experienced developer without forcing - any additional complexity on the novice developer. + into the exact ROAP API [I-D.jennings-rtcweb-signaling], thereby + implementing the ROAP signaling protocol. Such a library could of + course also implement other popular signaling protocols, including + SIP or Jingle. In this fashion we can enable greater control for the + experienced developer without forcing any additional complexity on + the novice developer. -3. Other Approaches Considered +2. Other Approaches Considered Another approach that was considered for JSEP was to move the mechanism for generating offers and answers out of the browser as well. Instead of providing createOffer/createAnswer methods within the browser, this approach would instead expose a getCapabilities API which would provide the application with the information it needed in order to generate its own session descriptions. This increases the amount of work that the application needs to do; it needs to know how to generate session descriptions from capabilities, and especially how to generate the correct answer from an arbitrary offer and the supported capabilities. While this could certainly be addressed by using a library like the one mentioned above, it basically forces the - use of said library even for a simple example. Exposing - createOffer/createAnswer avoids that problem, but still allows - applications to generate their own offers/answers if they choose, - using the description generated by createOffer as an indication of - the browser's capabilities. + use of said library even for a simple example. Exposing createOffer/ + createAnswer avoids that problem, but still allows applications to + generate their own offers/answers if they choose, using the + description generated by createOffer as an indication of the + browser's capabilities. Note also that while JSEP transfers more control to Javascript, it is not intended to be an example of a "low-level" API. The general argument against a low-level API is that there are too many necessary API points, and they can be called in any order, leading to something that is hard to specify and test. In the approach proposed here, control is performed via session descriptions; this requires only a few APIs to handle these descriptions, and they are evaluated in a specific fashion, which reduces the number of possible states and interactions. +3. Terminology + + The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", + "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this + document are to be interpreted as described in RFC 2119 [RFC2119]. + 4. Semantics and Syntax 4.1. Signaling Model JSEP does not specify a particular signaling model or state machine, other than the generic need to exchange RFC 3264 offers and answers in order for both sides of the session to know how to conduct the session. JSEP provides mechanisms to create offers and answers, as well as to apply them to a session. However, the actual mechanism by which these offers and answers are communicated to the remote side, including addressing, retransmission, forking, and glare handling, is left entirely up to the application. +-----------+ +-----------+ - | Web App |<--- App-Specific Signaling --->| Web App | + | Web App |<--- App-Specific Signaling -->| Web App | +-----------+ +-----------+ - | | + ^ ^ | SDP | SDP V V +-----------+ +-----------+ | Browser |<----------- Media ------------>| Browser | +-----------+ +-----------+ Figure 1: JSEP Signaling Model 4.2. Session Descriptions and State Machine In order to establish the media plane, the user agent needs specific parameters to indicate what to transmit to the remote side, as well as how to handle the media that is received. These parameters are determined by the exchange of session descriptions in offers and answers, and there are certain details to this process that must be handled in the JSEP APIs. Whether a session description was sent or received affects the meaning of that description. For example, the list of codecs sent to a remote party indicates what the local side is willing to decode, - and what the remote party should send. Not all parameters follow this - rule; for example, the SRTP parameters [RFC4568] sent to a remote - party indicate what the local side will use to encrypt, and thereby - how the remote party should expect to receive. + and what the remote party should send. Not all parameters follow + this rule; for example, the SRTP parameters [RFC4568] sent to a + remote party indicate what the local side will use to encrypt, and + thereby how the remote party should expect to receive. In addition, various RFCs put different conditions on the format of - offers versus answers. For example, a offer may propose multiple SRTP - configurations, but an answer may only contain a single SRTP + offers versus answers. For example, a offer may propose multiple + SRTP configurations, but an answer may only contain a single SRTP configuration. Lastly, while the exact media parameters are only known only after a offer and an answer have been exchanged, it is possible for the offerer to receive media after they have sent an offer and before they have received an answer. To properly process incoming media in this case, the offerer's media handler must be aware of the details of the offerer before the answer arrives. Therefore, in order to handle session descriptions properly, the user agent needs: - 1. To know if a session description pertains to the local or - remote side. + 1. To know if a session description pertains to the local or remote + side. 2. To know if a session description is an offer or an answer. 3. To allow the offer to be specified independently of the answer. JSEP addresses this by adding both a setLocalDescription and a - setRemoteDescription method, and both these methods take a parameter - to indicate the type of session description being supplied. This - satisfies the requirements listed above for both the offerer, who - first calls setLocalDescription("offer", sdp) and then later - setRemoteDescription("answer", sdp), as well as for the answerer, who - first calls setRemoteDescription("offer", sdp) and then later - setLocalDescription("answer", sdp). While it could be possible to - implicitly determine the value of the offer/answer argument, + setRemoteDescription method and having session description objects + contain a type field indicating the type of session description being + supplied. This satisfies the requirements listed above for both the + offerer, who first calls setLocalDescription(sdp [offer]) and then + later setRemoteDescription(sdp [answer]), as well as for the + answerer, who first calls setRemoteDescription(sdp [offer]) and then + later setLocalDescription(sdp [answer]). While it could be possible + to implicitly determine the value of the offer/answer argument, requiring it to be specified explicitly is more robust, allowing invalid combinations (i.e. an answer before an offer) to generate an appropriate error. - It also allows for an answer to be treated as provisional by the + JSEP also allows for an answer to be treated as provisional by the application. Provisional answers provide a way for an answerer to - communicate session parameters back to the offerer, in order for the - session to begin, while allowing a final answer to be specified - later. This concept of a final answer is important to the + 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; as a - result, multiple dissimilar provisional answers can be received and - applied during call setup. + 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. - As in [RFC3264], an offerer can send an offer, and update it as long - as it has not been answered. The answerer can send back zero or more + In [RFC3264], the constraints at the signaling level is that only one + offer can be outstanding for a given session but from the media stack + level, a new offer can be generated at any point. For example, when + using SIP for signaling, if one offer is sent, then cancelled using a + SIP CANCEL, another offer can be generated even though no answer was + received for the first offer. To support this, the JSEP media layer + can provide an offer whenever the Javascript application needs one + for the signaling. The answerer can send back zero or more provisional answers, and finally end the offer-answer exchange by sending a final answer. The state machine for this is as follows: +-----------+ | | | | | Stable |<---------------\ | | | | | | +-----------+ | ^ | | | | OFFER | ANSWER | | | ANSWER | V | +-----------+ +-----------+ | | | | | | PRANSWER | | - | Offer |--------->| Pranswer | + | Offer |-------- >| Pranswer | | | | | | |----\ | |----\ +-----------+ | +-----------+ | ^ | ^ | | | | | \-----/ \-----/ OFFER PRANSWER Figure 2: JSEP State Machine Aside from these state transitions, there is no other difference between the handling of provisional ("pranswer") and final ("answer") answers. 4.3. Session Description Format - In the 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. + + In the WebRTC specification, session descriptions are formatted as + SDP messages. While this format is not optimal for manipulation from + Javascript, it is widely accepted, and frequently updated with new + features. Any alternate encoding of session descriptions would have + to keep pace with the changes to SDP, at least until the time that + this new encoding eclipsed SDP in popularity. As a result, JSEP + continues to use SDP as the internal representation for its session + descriptions. However, to simplify Javascript processing, and provide for future flexibility, the SDP syntax is encapsulated within a SessionDescription object, which can be constructed from SDP, and be - 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. + 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. - -4.4. Separation of Signaling and ICE State Machines + simplify handling of SessionDescriptions from Javascript. Though it + is unclear exactly what manipulations developer will commonly want to + do to SDP, it would be simple to write a Javascript library to + perform these manipulations. - JSEP does away with the SDP Agent within the browser, and this - functionality is now controlled directly by the application, which - uses the setLocalDescription and setRemoteDescription APIs to tell - the browser what SDP has been negotiated. The ICE Agent remains in - the browser, as it still needs to drive the process of gathering - candidates, connectivity checks, and related ICE functionality. +4.4. ICE When a new ICE candidate is available, the ICE Agent will notify the application via a callback; these candidates will automatically be - added to the local session description. When all candidates have been - gathered, the callback will also be invoked to signal that the + added to the local session description. When all candidates have + been gathered, the callback will also be invoked to signal that the gathering process is complete. -4.5. ICE Candidate Trickling +4.4.1. ICE Candidate Trickling Candidate trickling is a technique through which a caller may incrementally provide candidates to the callee after the initial - offer has been dispatched. 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. + offer has been dispatched; the semantics of "Trickle ICE" are defined + in [I-D.rescorla-mmusic-ice-trickle]. This process allows the callee + to begin acting upon the call and setting up the ICE (and perhaps + DTLS) connections immediately, without having to wait for the caller + to gather all possible candidates. This results in faster call + startup in cases where gathering is not performed prior to initating + the call. JSEP supports optional candidate trickling by providing APIs that provide control and feedback on the ICE candidate gathering process. Applications that support candidate trickling can send the initial offer immediately and send individual candidates when they get the - onicecandidate callback with a new candidate; applications that do - not support this feature can simply wait for the final onicecandidate - callback that indicates gathering is complete, and create and send - their offer, with all the candidates, at this time. + 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 can - supply them to its ICE Agent by calling an addIceCandidate method. - This triggers the ICE Agent to start using this remote candidate for - connectivity checks. Applications that do not make use of candidate - tricking can ignore addIceCandidate entirely, and use the - onicecandidate callback solely to indicate when candidate gathering - is complete. + Upon receipt of trickled candidates, the receiving application will + supply them to its ICE Agent. This triggers the ICE Agent to start + using the new remote candidates for connectivity checks. -4.6. ICE Candidate Format +4.4.1.1. 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. + As with session descriptions, the syntax of the IceCandidate object + provides some abstraction, but can be easily converted to and from + the SDP a=candidate lines. - The IceCandidate object has fields to indicate which m= line it - should be associated with, and a method to convert to a SDP - representation, ex: + The a=candidate lines are the only SDP information that is contained + within IceCandidate, as they represent the only information needed + that is not present in the initial offer (i.e. for trickle + candidates). This information is carried with the same syntax as the + "a=candidate" line in SDP. For example: - a=candidate:1 1 UDP 1694498815 66.77.88.99 10000 typ host + a=candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host - Currently, a=candidate lines are the only SDP information that is - contained within IceCandidate, as they represent the only information - needed that is not present in the initial offer (i.e. for trickle - candidates). + The IceCandidate object also contains fields to indicate which m= + line it should be associated with. The m line can be identified in + one of two ways; either by a m-line index, or a MID. The m-line + index is a zero-based index, referring to the Nth m-line in the SDP. + The MID uses the "media stream identification", as defined in [RFC + 3388], to identify the m-line. WebRTC implementations creating an + ICE Candidate object MUST populate both of these fields. + Implementations receiving an ICE Candidate object SHOULD use the MID + if they implement that functionality, or the m-line index, if not. -4.7. Interactions With Forking +4.5. Interactions With Forking -4.7.1. Serial Forking + Some call signaling systems allow various types of forking where an + SDP Offer may be provided to more than one device. For example, SIP + RFC 3261 defines both a "Parallel Search" and "Sequential Search". + Although these are primarily signaling level issues that are outside + the scope of JSEP, they do have some impact on the configuration of + the media plane, which is relevant. When forking is happening at the + signaling layer, the Javascript application responsible for the + signaling needs to make the decisions about what media should be sent + or received at any point of time and which remote endpoint it should + communicate with. JSEP is used to make sure the media engine can + make the RTP and media perform as required by the application. The + basic operations that the applications can have the media engine do + are: - Serial forking involves a call being dispatched to multiple remote - callees, where each callee can accept the call, but only one active - session ever exists at a time; no mixing of received media is + Start exchanging media to a given remote peer but keep all the + resources reserved in the offer. + + Start exchanging media with a given remote peer and free any + resources in the offer that are not being used. + +4.5.1. Sequential Forking + + Sequential forking involves a call being dispatched to multiple + remote callees, where each callee can accept the call, but only one + active session ever exists at a time; no mixing of received media is performed. JSEP handles serial forking well, allowing the application to easily - 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. + control the policy for selecting the desired remote endpoint. When + an answer arrives from one of the callees, the application can choose + to apply it either as a provisional answer, leaving open the + possibility of using a different answer in the future, or apply it as + a final answer, ending the setup flow. In a "first-one-wins" situation, the first answer will be applied as - a final answer, and the application will send a terminate message to - any subsequent answers. In SIP parlance, this would be ACK + BYE. + 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. + provisional answers, and any previous call leg will be terminated. + At some point, the application will end the setup process, perhaps + with a timer; at this point, the application could reapply the + existing remote description as a final answer. -4.7.2. Parallel Forking +4.5.2. Parallel Forking Parallel forking involves a call being dispatched to multiple remote callees, where each callee can accept the call, and multiple - simultaneous active sessions can be established as a result. If - multiple callees send media, this media is mixed and played out at - the caller side. - - JSEP can handle parallel forking by "cloning" the session when needed - to create multiple parallel sessions. When the first answer is - received, the caller can clone the existing session, and then apply - the answer as a final answer to the original session. Upon receiving - the next answer, the cloned session is cloned again, and the received - answer is applied as a final answer to the first clone. This process - repeats until the caller decides to end the setup flow, and closes - the final cloned session. + simultaneous active signaling sessions can be established as a + result. If multiple callees send media at the same time, the + possibilities for handling this are described in Section 3.1 of RFC + 3960. Most SIP devices today only support exchanging media with a + single device at a time, and do not try to mix multiple early media + audio sources, as that could result in a confusing situation. For + example. consider having a European ringback tone mixed together with + the North American ringback tone - the resulting sound would not be + like either tone, and would confuse the user. If the signaling + application wishes to only exchange media with one of the remote + endpoints at a time, then from a media engine point of view, this is + exactly like the sequential forking case. - Cloned sessions inherit the local session description and candidates - from their parent, and an empty remote description; only sessions - that have not yet applied an answer can be cloned. Each cloned - session may discover new peer-reflexive candidates; these candidates - will be supplied via the onicecandidate callback to that specific - session. Since the clone uses the same local description as its - parent, creating a clone will fail if it is not possible to reserve - the same resources for the clone as have already been reserved by the - parent. + In the parallel forking case where the Javascript application wishes + to simultaneously exchange media with multiple peers, the flow is + slightly more complex, but the Javascript application can follow the + strategy that RFC 3960 describes using UPDATE. (It is worth noting + that use cases where this is the desired behavior are very unusual.) + The UPDATE approach allows the signaling to set up a separate media + flow for each peer that it wishes to exchange media with. In JSEP, + this offer used in the UPDATE would be formed by simply creating a + new PeerConnection and making sure that the same local media streams + have been added into this new PeerConnection. Then the new + PeerConnection object would produce a SDP offer that could be used by + the signaling to perform the UPDATE strategy discussed in RFC 3690. - As a result of this cloning, the application will end up with N - parallel sessions, each with a local and remote description and their - own local and remote addresses. The media flow from these sessions - can be managed by specifying SDP direction attributes in the - descriptions, or the application can choose to play out the media - from all sessions mixed together. Of course, if the application wants - to only keep a single session, it can simply terminate the sessions - that it no longer needs. + 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. -4.8. Session Rehydration +4.6. Session Rehydration In the event that the local application state is reinitialized, either due to a user reload of the page, or a decision within the application to reload itself (perhaps to update to a new version), it is possible to keep an existing session alive via a process called "rehydration". - With rehydration, the current local session description is persisted - somewhere outside of the page, perhaps on the application server, or - in browser local storage. The page is then reloaded, and a new - session object is created in Javascript. The saved local session is - now retrieved, but the previous ICE candidates will no longer be - valid in this case, so we will need to perform an ICE restart; to do - so, we simply generate a new ICE ufrag/pwd combo for the local - description. + With rehydration, the current signaling state is persisted somewhere + outside of the page, perhaps on the application server, or in browser + local storage. The page is then reloaded, and a new session object + is created in Javascript. The saved signaling state is now + retrieved, and a new PeerConnection object is created for the + session. At this point a new offer can be generated by the new + PeerConnection, with new ICE and SDES credentials. This can then be + used to re-initiate the session with the existing remote endpoint, + who simply sees the new offer as an in-call renegotiation, and will + reply with an answer that can be supplied to setRemoteDescription. + ICE processing proceeds as usual, and as soon as connectivity is + established, the session will be back up and running again. - The modified local description is then installed via - setLocalDescription, and sent off as an offer to the remote side, who - will reply with an answer that can be supplied to - setRemoteDescription. ICE processing proceeds as usual, and as soon - as connectivity is established, the session will be back up and - running again. + Open Issue: EKR proposed an alternative rehydration approach where + the actual internal PeerConnection object in the browser was kept + alive for some time after the web page was killed and provided some + way for a new page to acquire the old PeerConnection object. 5. Interface This section details the basic operations that must be present to implement JSEP functionality. The actual API exposed in the W3C API may have somewhat different syntax, but should map easily to these concepts. -5.1. Methods +5.1. SDP Requirements -5.1.1. createOffer + Note: The text in this section may not represent working group + consensus and is put here so that the working group can discuss it + and find out how to change it such that it does have consensus. + + When generating SDP blobs, either for offers or answers, the + generated SDP needs to conform to the following specifications. + Similarly, in order to properly process received SDP blobs, + implementations need to implement the functionality described in the + following specifications. This list is derived from + [I-D.ietf-rtcweb-rtp-usage]. + + RFC4566 is the base SDP specification and MUST be implemented. + + RFC5124 MUST be supported for signaling RTP/SAVPF RTP profile. + + RFC5104 MUST be implemented to signal RTCP based feedback. + + RFC5761 MUST be implemented to signal multiplexing of RTP and + RTCP. + + RFC5245 MUST be implemented for signaling the ICE candidate lines + corresponding to each media stream. + + RFC3264 MUST be implemented to signal information about media + direction. + + The RFC5888 grouping framework MUST be implemented for signaling + the grouping information. + + RFC5506 MAY be implemented to signal Reduced-Size RTCP messages. + + RFC5576 MAY be implemented to signal RTP SSRC values. + + RFC3556 with bandwidth modifiers MAY be supported for specifying + RTCP bandwidth as a fraction of the media bandwidth, RTCP fraction + allocated to the senders and setting maximum media bit-rate + boundaries. + + As required by RFC 4566 Section 5.13 JSEP implementations MUST ignore + unknown attributes (a=) lines. + + Example SDP for RTCWeb call flows can be found in + [I-D.nandakumar-rtcweb-sdp]. + +5.2. Methods + +5.2.1. createOffer The createOffer method generates a blob of SDP that contains a RFC 3264 offer with the supported configurations for the session, including descriptions of the local MediaStreams attached to this PeerConnection, the codec/RTP/RTCP options supported by this implementation, and any candidates that have been gathered by the ICE - Agent. A constraints parameters may be supplied to provide additional - control over the generated offer, e.g. to get a full set of session - capabilities, or to request a new set of ICE credentials. + Agent. A constraints parameters may be supplied to provide + additional control over the generated offer, e.g. to get a full set + of session capabilities, or to request a new set of ICE credentials. In the initial offer, the generated SDP will contain all desired functionality for the session (certain parts that are supported but not desired by default may be omitted); for each SDP line, the generation of the SDP must follow the appropriate process for generating an offer. In the event createOffer is called after the session is established, createOffer will generate an offer that is compatible with the current session, incorporating any changes that have been made to the session since the last complete offer-answer exchange, such as addition or removal of streams. If no changes have @@ -600,53 +608,53 @@ Session descriptions generated by createOffer must be immediately usable by setLocalDescription; if a system has limited resources (e.g. a finite number of decoders), createOffer should return an offer that reflects the current state of the system, so that setLocalDescription will succeed when it attempts to acquire those resources. Because this method may need to inspect the system state to determine the currently available resources, it may be implemented as an async operation. - Calling this method does not change state; its use is not required. + Calling this method may do things such as generate new ICE + credentials, but does not change media state. -5.1.2. createAnswer +5.2.2. createAnswer The createAnswer method generates a blob of SDP that contains a RFC 3264 SDP answer with the supported configuration for the session that - is compatible with the parameters supplied in |offer|. Like + 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. 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. + that specifies how the media plane should be established. Session descriptions generated by createAnswer must be immediately usable by setLocalDescription; like createOffer, the returned description should reflect the current state of the system. Because this method may need to inspect the system state to determine the currently available resources, it may need to be implemented as an async operation. - Calling this method does not change state; its use is not required. + Calling this method may do things such as generate new ICE + credentials, but does not change media state. -5.1.3. SessionDescriptionType +5.2.3. SessionDescriptionType - The strings "offer", "pranswer", and "answer" serve as type arguments - to setLocalDescription and setRemoteDescription. They provide - information as to how the description parameter should be parsed, and - how the media state should be changed. + Session description objects (RTCSessionDescription) may be of type + "offer", "pranswer", and "answer". These types provide information + as to how the description parameter should be parsed, and how the + media state should be changed. "offer" indicates that a description should be parsed as an offer; said description may include many possible media configurations. A description used as an "offer" may be applied anytime the PeerConnection is in a stable state, or as an update to a previously sent but unanswered "offer". "pranswer" indicates that a description should be parsed as an answer, but not a final answer, and so should not result in the freeing of allocated resources. It may result in the start of media @@ -661,428 +669,253 @@ response to a "offer", or an update to a previously sent "pranswer". The application can use some discretion on whether an answer should be applied as provisional or final. For example, in a serial forking scenario, an application may receive multiple "final" answers, one from each remote endpoint. The application could accept the initial answers as provisional answers, and only apply an answer as final when it receives one that meets its criteria (e.g. a live user instead of voicemail). -5.1.4. setLocalDescription +5.2.3.1. Creating Answers + + Most web applications will not need to create answers using the + "pranswer" type. The general recommendation for a web application + would be to create an answer more or less immediately after receiving + the offer, instead of waiting for a human user to provide input. + Later when the human input is received, the applications can create a + new offer to update the previous offer/answer pair. Some + applications may not be able to do this, particularly ones that Some + application may not be able to do this, particular ones that are + attempting to gateway to other signaling protocols. + + Consider a typical web application that will set up a data channel, + an audio channel, and a video channel. When an endpoint receives an + offer with these channels, it could send an answer accepting the data + channel for two-way data, and accepting the audio and video tracks as + receive-only. It could then ask the user if they wanted to transmit + audio and video to the far end, acquire the local media streams, and + send a new offer to the remote side moving the audio and video to be + two-way media. By the time the human has authorized sending media, + it is likely that the ICE and DTLS handshaking with the remote side + will already be set up. + +5.2.4. setLocalDescription The setLocalDescription method instructs the PeerConnection to apply - the supplied SDP blob as its local configuration. The type parameter + the supplied SDP blob as its local configuration. The type field indicates whether the blob should be processed as an offer, provisional answer, or final answer; offers and answers are checked differently, using the various rules that exist for each SDP line. This API changes the local media state; among other things, it sets up local resources for receiving and decoding media. In order to successfully handle scenarios where the application wants to offer to change from one media format to a different, incompatible format, the PeerConnection must be able to simultaneously support use of both the old and new local descriptions (e.g. support codecs that exist in both descriptions) until a final answer is received, at which point the PeerConnection can fully adopt the new local description, or roll back to the old description if the remote side denied the change. If setRemoteDescription was previous called with an offer, and setLocalDescription is called with an answer (provisional or final), and the media directions are compatible, this will result in the starting of media transmission. -5.1.5. setRemoteDescription +5.2.5. setRemoteDescription + The setRemoteDescription method instructs the PeerConnection to apply the supplied SDP blob as the desired remote configuration. As in - setLocalDescription, the |type| parameter indicates how the blob + setLocalDescription, the type field of the indicates how the blob should be processed. This API changes the local media state; among other things, it sets up local resources for sending and encoding media. If setRemoteDescription was previous called with an offer, and setLocalDescription is called with an answer (provisional or final), and the media directions are compatible, this will result in the starting of media transmission. -5.1.6. localDescription +5.2.6. localDescription The localDescription method returns a copy of the current local configuration, i.e. what was most recently passed to setLocalDescription, plus any local candidates that have been generated by the ICE Agent. A null object will be returned if the local description has not yet been established. -5.1.7. remoteDescription +5.2.7. remoteDescription The remoteDescription method returns a copy of the current remote configuration, i.e. what was most recently passed to setRemoteDescription, plus any remote candidates that have been supplied via processIceMessage. A null object will be returned if the remote description has not yet been established. -5.1.8. updateIce +5.2.8. updateIce The updateIce method allows the configuration of the ICE Agent to be changed during the session, primarily for changing which types of local candidates are provided to the application and used for connectivity checks. A callee may initially configure the ICE Agent to use only relay candidates, to avoid leaking location information, but update this configuration to use all candidates once the call is accepted. Regardless of the configuration, the gathering process collects all available candidates, but excluded candidates will not be surfaced in onicecallback or used for connectivity checks. This call may result in a change to the state of the ICE Agent, and may result in a change to media state if it results in connectivity being established. -5.1.9. addIceCandidate +5.2.9. addIceCandidate The addIceCandidate method provides a remote candidate to the ICE Agent, which will be added to the remote description. Connectivity checks will be sent to the new candidate. This call will result in a change to the state of the ICE Agent, and may result in a change to media state if it results in connectivity being established. -5.2. Configurable SDP Parameters +6. Configurable SDP Parameters + + Note: This section is still very early and is likely to + significantly change as we get a better understanding of the a) the + use cases for this b) the implications at the protocol level c) + feedback from implementors on what they can do. The following is a partial list of SDP parameters that an application may want to control, in either local or remote descriptions, using this API. - - remove or reorder codecs (m=) - - change codec attributes (a=fmtp; ptime) - - enable/disable BUNDLE (a=group) - - enable/disable RTCP mux (a=rtcp-mux) - - remove or reorder SRTP crypto-suites (a=crypto) - - change SRTP parameters or keys (a=crypto) - - change send resolution or framerate (TBD) - - change desired recv resolution or framerate (TBD) - - change total bandwidth (b=) - - remove desired AVPF mechanisms (a=rtcp-fb) - - remove RTP header extensions (a=rtphdr-ext) - - add/change SSRC grouping (e.g. FID, RTX, etc) (a=ssrc-group) - - add SSRC attributes (a=ssrc) - - change ICE ufrag/password (a=ice-ufrag/pwd) - - change media send/recv state (a=sendonly/recvonly/inactive) - - For example, an application could implement call hold by adding an - a=inactive attribute to its local description, and then applying and - signaling that description. - -6. Media Setup Overview - - The example here shows a typical call setup using the JSEP model, - indicating the functions that are called and the state changes that - occur. We assume the following architecture in this example, where UA - is synonymous with "browser", and JS is synonymous with "web - application": - - OffererUA <-> OffererJS <-> WebServer <-> AnswererJS <-> AnswererUA - -6.1. Initiating the Session - - The initiator creates a PeerConnection, hooks up to its ICE callback, - and adds the desired MediaStreams (presumably obtained via - getUserMedia). The ICE gathering process begins to gather candidates - for a default number of streams, as the exact number will not be - known until the local description is applied. The PeerConnection is - in the NEW state. - - OffererJS->OffererUA: var pc = new PeerConnection(config, null); - OffererJS->OffererUA: pc.onicecandidate = onIceCandidate; - OffererJS->OffererUA: pc.addStream(stream); - -6.1.1. Generating An Offer - - The initiator then creates a session description to offer to the - callee. This description includes the codecs and other necessary - session parameters, as well as information about each of the streams - that has been added (e.g. SSRC, CNAME, etc.) The created description - includes all parameters that the offerer's UA supports; if the - initiator wants to influence the created offer, they can pass in a - MediaConstraints object to createOffer that allows for customization - (e.g. if the initiator wants to receive but not send video). The - initiator can also directly manipulate the created session - description as well, perhaps if it wants to change the priority of - the offered codecs. - - OffererJS->OffererUA: var offer = pc.createOffer(null); - -6.1.2. Applying the Offer - - The initiator then instructs the PeerConnection to use this offer as - the local description for this session, i.e. what codecs it will use - for received media, what SRTP keys it will use for sending media (if - using SDES), etc. In order that the UA handle the description - properly, the initiator marks it as an offer when calling - setLocalDescription; this indicates to the UA that multiple - capabilities have been offered, but this set may be pared back later, - when the answer arrives. - - Since the local user agent must be prepared to receive media upon - applying the offer, this operation will cause local decoder resources - to be allocated, based on the codecs indicated in the offer. - - OffererJS->OffererUA: pc.setLocalDescription("offer", offer); - -6.1.3. Handling ICE Callbacks - The initiator starts to receive callbacks on its onicecandidate - handler. Candidates are provided to the IceCallback as they are - allocated; when the last allocation completes or times out, this - callback will be invoked with a null argument. - - OffererUA->OffererJS: onIceCandidate(candidate); - -6.1.4. Serializing the Offer and Candidates - - At this point, the offerer is ready to send its offer to the callee - using its preferred signaling protocol. Depending on the protocol, it - can either send the initial session description first, and then - "trickle" the ICE candidates as they are given to the application, or - it can wait for all the ICE candidates to be collected, and then send - the offer and list of candidates all at once. - -6.2. Receiving the Session - - Through the chosen signaling protocol, the recipient is notified of - an incoming session request. It creates a PeerConnection, and sets up - its own ICE callback. The ICE gathering process begins to gather - candidates for a default number of streams. - - AnswererJS->AnswererUA: var pc = new PeerConnection(config, null); - AnswererJS->AnswererUA: pc.onicecandidate = onIceCandidate; - -6.2.1. Receiving the Offer - - The recipient converts the received offer from its signaling protocol - into SDP format, and supplies it to its PeerConnection, again marking - it as an offer. As a remote description, the offer indicates what - codecs the remote side wants to use for receiving, as well as what - SRTP keys it will use for sending. The setting of the remote - description causes callbacks to be issued, informing the application - of what kinds of streams are present in the offer. - - This step will also cause encoder resources to be allocated, based on - the codecs specified in |offer|. - - AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer); - AnswererUA->AnswererJS: onAddStream(stream); - -6.2.2. Handling ICE Messages - - If ICE candidates from the remote site were included in the offer, - the ICE Agent will automatically start trying to use them. Otherwise, - if ICE candidates are sent separately, they are passed into the - PeerConnection when they arrive. - - AnswererJS->AnswererUA: pc.addIceCandidate(candidate); - -6.2.3. Generating the Answer - - Once the recipient has decided to accept the session, it generates an - answer session description. This process performs the appropriate - intersection of codecs and other parameters to generate the correct - answer. As with the offer, MediaConstraints can be provided to - influence the answer that is generated, and/or the application can - post-process the answer manually. - - AnswererJS->AnswererUA: pc.createAnswer(offer, null); + o remove or reorder codecs (m=) -6.2.4. Applying the Answer + o change codec attributes (a=fmtp; ptime) - The recipient then instructs the PeerConnection to use the answer as - its local description for this session, i.e. what codecs it will use - to receive media, etc. It also marks the description as an answer, - which tells the UA that these parameters are final. This causes the - PeerConnection to move to the ACTIVE state, and transmission of media - by the answerer to start (assuming both sides have indicated this in - their descriptions). + o enable/disable BUNDLE (a=group) - AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer); - AnswererUA->OffererUA: + o enable/disable RTCP mux (a=rtcp-mux) -6.2.5. Serializing the Answer + o change send resolution or framerate (TBD) - As with the offer, the answer (with or without candidates) is now - converted to the desired signaling format and sent to the initiator. + o change desired recv resolution or framerate (TBD) -6.3. Completing the Session + o change total bandwidth (b=) -6.3.1. Receiving the Answer + o remove desired AVPF mechanisms (a=rtcp-fb) - The initiator converts the answer from the signaling protocol and - applies it as the remote description, marking it as an answer. This - causes the PeerConnection to move to the ACTIVE state, and - transmission of media by the offerer to start (assuming both sides - have indicated this in their descriptions). + o remove RTP header extensions (a=rtphdr-ext) - OffererJS->OffererUA: pc.setRemoteDescription("answer", answer); - OffererUA->AnswererUA: + o add/change SSRC grouping (e.g. FID, RTX, etc) (a=ssrc-group) -6.4. Updates to the Session + o add SSRC attributes (a=ssrc) - 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. + o change media send/recv state (a=sendonly/recvonly/inactive) - Note also that in an update scenario, the roles may be reversed, i.e. - the update offerer can be different than the original offerer. + 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. 7. Security Considerations TODO 8. IANA Considerations This document requires no actions from IANA. 9. Acknowledgements Harald Alvestrand, Dan Burnett, Neil Stratford, Eric Rescorla, Anant Narayanan, and Adam Bergkvist all provided valuable feedback on this - proposal. Matthew Kaufman provided the observation that keeping state - out of the browser allows a call to continue even if the page is - reloaded. Richard Ejzak provided the specifics on session cloning. + proposal. Suhas Nandakumar provided text and input for SDP + requirements. Matthew Kaufman provided the observation that keeping + state out of the browser allows a call to continue even if the page + is reloaded. 10. References 10.1. Normative References + [I-D.rescorla-mmusic-ice-trickle] + Rescorla, E., Uberti, J., and E. Ivov, "Trickle ICE: + Incremental Provisioning of Candidates for the Interactive + Connectivity Establishment (ICE) Protocol", + draft-rescorla-mmusic-ice-trickle-00 (work in progress), + October 2012. + [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model - with Session Description Protocol (SDP)", RFC 3264, June 2002. + with Session Description Protocol (SDP)", RFC 3264, + June 2002. [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006. 10.2. Informative References - [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session - Description Protocol (SDP) Security Descriptions for Media Streams", - RFC 4568, July 2006. - - [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment - (ICE): A Protocol for Network Address Translator (NAT) Traversal for - Offer/Answer Protocols", RFC 5245, April 2010. - - [webrtc-api] Bergkvist, Burnett, Jennings, Narayanan, "WebRTC 1.0: - - Real-time Communication Between Browsers", May 2011. - - Available at http://dev.w3.org/2012/webrtc/editor/webrtc.html - -Appendix A. JSEP Implementation Examples - -A.1. Example API - - The interface below shows a basic Javascript API that could be used - to expose the functionality discussed in this document. This API is - used for the examples in the following parts of this Appendix. - - // actions, for setLocalDescription/setRemoteDescription - enum SessionDescriptionType { "offer", "pranswer", "answer" } - - // constraints that can be supplied to the ctor or createXXXX - enum MediaConstraints { - "offerConfig", // controls the kind of offer created; - // "default" (normal offer) - // "caps" (all capabilities) - // "new" (brand new description) - // "iceRestart" (new ICE creds) + [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. - "iceTransports", // controls ICE candidates; can be - // "none" (no candidates) - // "relay" (only relay candidates) - // "all" (all available candidates) - } + [I-D.jennings-rtcweb-signaling] + Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/ + Answer Protocol (ROAP)", + draft-jennings-rtcweb-signaling-01 (work in progress), + October 2011. - [Constructor (int index, DOMString id, in DOMString candidateLine)] - interface IceCandidate { - // the m= line index for this candidate - readonly attribute int mLineIndex - // the mid for the m= line for this candidate - readonly attribute DOMString mLineId; - // creates a SDP-ized form of this candidate - stringifier DOMString (); - }; + [I-D.nandakumar-rtcweb-sdp] + Nandakumar, S. and C. Jennings, "SDP for the WebRTC", + draft-nandakumar-rtcweb-sdp-00 (work in progress), + October 2012. - [Constructor (DOMString sdp)] - interface SessionDescription { - // adds the specified candidate to the description - void addCandidate(IceCandidate candidate); - // serializes the description to SDP - stringifier DOMString (); - }; + [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session + Description Protocol (SDP) Security Descriptions for Media + Streams", RFC 4568, July 2006. - [Constructor (DOMString configuration, - optional MediaConstraints constraints)] - interface PeerConnection { - // creates a blob of SDP to be provided as an offer. - SessionDescription createOffer ( - SessionDescriptionCallback successCb, - optional ErrorCallback errorCb, - optional MediaContraints constraints); - // creates a blob of SDP to be provided as an answer. - SessionDescription createAnswer ( - SessionDescription offer, - SessionDescriptionCallback successCb, - optional ErrorCallback errorCb, - optional MediaContraints constraints); + [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment + (ICE): A Protocol for Network Address Translator (NAT) + Traversal for Offer/Answer Protocols", RFC 5245, + April 2010. - // sets the local session description - void setLocalDescription ( - SessionDescriptionType action, - SessionDescription desc); - // sets the remote session description - void setRemoteDescription ( - SessionDescriptionType action, - SessionDescription desc) - // returns the current local session description - readonly attribute SessionDescription localDescription; - // returns the current remote session description - readonly attribute SessionDescription remoteDescription; + [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, + . - // updates the constraints for ICE processing - void updateIce ( - optional DOMString configuration, - optional MediaConstraints constraints); - // starts using a received remote ICE candidate - void addIceCandidate ( - IceCandidate candidate); - // notifies the application of a new local ICE candidate - attribute Function? onicecandidate; - }; +Appendix A. JSEP Implementation Examples -A.2. Example API Flows +A.1. Example API Flows Below are several sample flows for the new PeerConnection and library APIs, demonstrating when the various APIs are called in different situations and with various transport protocols. For clarity and simplicity, the createOffer/createAnswer calls are assumed to be synchronous in these examples, whereas the actual APIs are async. -A.2.1. Call using ROAP +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 } @@ -1105,21 +938,21 @@ // ICE Completes (at Answerer) AnswererUA->AnswererJS: onopen(); AnswererUA->OffererUA: Media // ICE Completes (at Offerer) OffererUA->OffererJS: onopen(); OffererJS->AnswererJS: {"type":"OK" } OffererUA->AnswererUA: Media -A.2.2 Call using XMPP +A.1.2. Call using XMPP This example demonstrates an XMPP call, making use of trickle candidates. // Call is initiated toward Answerer OffererJS->OffererUA: pc = new PeerConnection(); OffererJS->OffererUA: pc.addStream(localStream, null); OffererJS->OffererUA: offer = pc.createOffer(null); OffererJS->OffererUA: pc.setLocalDescription("offer", offer); OffererJS: xmpp = createSessionInitiate(offer); @@ -1159,21 +993,21 @@ OffererUA->OffererJS: onaddstream(remoteStream); // ICE Completes (at Answerer) AnswererUA->AnswererJS: onopen(); AnswererUA->OffererUA: Media // ICE Completes (at Offerer) OffererUA->OffererJS: onopen(); OffererUA->AnswererUA: Media -A.2.3. Adding video to a call, using XMPP +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) @@ -1187,55 +1021,56 @@ 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.2.4. Simultaneous add of video streams, using XMPP +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. + conflict. The XMPP description-info mechanism is used to indicate + the new sources to the remote side. // Offerer and "Answerer" add video streams at the same time OffererJS->OffererUA: pc.addStream(offererVideoStream2) OffererJS->OffererUA: offer = pc.createOffer(null); OffererJS: xmpp = createDescriptionInfo(offer); OffererJS->OffererUA: pc.setLocalDescription("offer", offer); OffererJS->AnswererJS: 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.2.5. Call using SIP +A.1.5. Call using SIP This example demonstrates a simple SIP call (e.g. where the client talks to a SIP proxy over WebSockets). // Call is initiated toward Answerer OffererJS->OffererUA: pc = new PeerConnection(); OffererJS->OffererUA: pc.addStream(localStream, null); OffererUA->OffererJS: onicecandidate(candidate); OffererJS->OffererUA: offer = pc.createOffer(null); OffererJS->OffererUA: pc.setLocalDescription("offer", offer); @@ -1258,25 +1093,26 @@ // 200 OK arrives at Offerer OffererJS: answer = parseResponse(sip); OffererJS->OffererUA: peer.setRemoteDescription("answer", answer); OffererUA->OffererJS: onaddstream(remoteStream); OffererJS->AnswererJS: ACK // ICE Completes (at Answerer) AnswererUA->AnswererJS: onopen(); AnswererUA->OffererUA: Media + // ICE Completes (at Offerer) OffererUA->OffererJS: onopen(); OffererUA->AnswererUA: Media -A.2.6. Handling early media (e.g. 1-800-FEDEX), using SIP +A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using SIP This example demonstrates how early media could be handled; for simplicity, only the offerer side of the call is shown. // Call is initiated toward Answerer OffererJS->OffererUA: pc = new PeerConnection(); OffererJS->OffererUA: pc.addStream(localStream, null); OffererUA->OffererJS: onicecandidate(candidate); OffererJS->OffererUA: offer = pc.createOffer(null); OffererJS->OffererUA: pc.setLocalDescription("offer", offer); @@ -1290,130 +1126,62 @@ // ICE Completes (at Offerer) OffererUA->OffererJS: onopen(); OffererUA->AnswererUA: Media // 200 OK arrives at Offerer OffererJS: answer = parseResponse(sip); OffererJS->OffererUA: pc.setRemoteDescription("answer", answer); OffererJS->AnswererJS: ACK -A.3. Full Example Application +Appendix B. Change log - The following example demonstrates a simple video calling - application, using both trickle candidates and provisional answers to - speed up call setup. + Changes in draft -02: - // Usage: - // Caller calls start(true) - // Callee calls start(false) to prepare the call/start connecting, - // and then accept() to start transmitting. + o Converted from nroff - var signalingChannel = createSignalingChannel(); - var pc = null; - var localStream = null; - signalingChannel.onmessage = handleMessage; - // Set up the call, get access to local media, - // and establish connectivity. - function start(isCaller) { - // Create a PeerConnection and hook up the IceCallback. - pc = new webkitPeerConnection(null, null); - pc.onicecandidate = function(evt) { - sendMessage("candidate", evt.candidate); - }; + o Removed comparisons to old approaches abandoned by the working + group - // Get the local stream and show it in the local video element; - // if we're the caller, ship off an offer once we get the stream. - navigator.webkitGetUserMedia( - {"audio": true, "video": true}, function (stream) { - selfView.src = webkitURL.createObjectURL(stream); - localStream = stream; - if (isCaller) { - pc.addStream(stream); - pc.createOffer(function(sdp) { - setLocalAndSendMessage("offer", sdp); - }); - }); + o Removed stuff that has moved to W3C specificaiton - // When the remote stream arrives, show it in the remote - // video element. - pc.onaddstream = function(evt) { - remoteView.src = webkitURL.createObjectURL(evt.stream); - }; - } + o Align SDP handling with W3C draft - // The callee has accepted the call, attach their media - // and send a final answer. - function accept() { - // The addStream could also be done for the pranswer, - // although that would delay the pranswer - // (due to the need for user consent) - pc.addStream(localStream); // assumes we have the stream already - pc.createAnswer(msg.sdp, function(sdp) { - setLocalAndSendMessage("answer", sdp); - }); - } + o Clarified section on forking. - // -- internal methods -- + Changes in draft -01: - // Apply SDP locally and send it to the remote side. - function setLocalAndSendMessage(type, sdp) { - pc.setLocalDescription(type, sdp); - sendMessage(type, sdp); - } - // Send a signaling message to the remote side. - function sendMessage(type, obj) { - signalingChannel.send( - JSON.stringify({ "type": type, "sdp": obj })); - } + o Added diagrams for architecture and state machine. - // Handle incoming signaling messages. - function handleMessage(str) { - var msg = JSON.parse(str); - switch (msg.type) { - case "offer": - // create the PeerConnection - start(false); - // feed the received offer into the PeerConnection - pc.setRemoteDescription(msg.type, msg.sdp); - // create provisional answer to allow ICE/DTLS to start - pc.createAnswer(msg.sdp, function(sdp) { - setDirection(sdp, "recvonly"); - setLocalAndSendMessage("pranswer", sdp); - }); - break; - case "pranswer": - case "answer": - pc.setRemoteDescription(msg.type, msg.sdp); - break; - case "candidate": - pc.addIceCandidate(msg.sdp); - break; - } - } + o Added sections on forking and rehydration. -Appendix B. Change log + o Clarified meaning of "pranswer" and "answer". - 01: Added diagrams for architecture and state machine. - Added sections on forking and rehydration. - Clarified meaning of "pranswer" and "answer". - Reworked how ICE restarts and media directions are controlled. - Added list of parameters that can be changed in a description. - Updated suggested API and examples to match latest thinking. - Suggested API and examples have been moved to an appendix. - 00: Migrated from draft-uberti-rtcweb-jsep-02. + o Reworked how ICE restarts and media directions are controlled. + + o Added list of parameters that can be changed in a description. + + o Updated suggested API and examples to match latest thinking. + + o Suggested API and examples have been moved to an appendix. + + Changes in draft -00: + + o Migrated from draft-uberti-rtcweb-jsep-02. Authors' Addresses Justin Uberti Google - 5 Cambridge Center - Cambridge, MA 02142 + 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@cisco.com + Email: fluffy@iii.ca