--- 1/draft-ietf-rtcweb-jsep-03.txt 2013-09-17 16:14:25.478800573 -0700 +++ 2/draft-ietf-rtcweb-jsep-04.txt 2013-09-17 16:14:25.566802765 -0700 @@ -1,115 +1,128 @@ Network Working Group J. Uberti Internet-Draft Google Intended status: Standards Track C. Jennings -Expires: August 29, 2013 Cisco - February 25, 2013 +Expires: March 22, 2014 Cisco + September 18, 2013 Javascript Session Establishment Protocol - draft-ietf-rtcweb-jsep-03 + draft-ietf-rtcweb-jsep-04 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. + application to 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 +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. + This Internet-Draft will expire on March 22, 2014. 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. Table of Contents - 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 - 1.1. General Design of JSEP . . . . . . . . . . . . . . . . . . 4 - 1.2. Other Approaches Considered . . . . . . . . . . . . . . . 6 + 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 + 1.1. General Design of JSEP . . . . . . . . . . . . . . . . . 3 + 1.2. Other Approaches Considered . . . . . . . . . . . . . . . 5 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. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . 6 + 3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 6 + 3.2. Session Descriptions and State Machine . . . . . . . . . 7 + 3.3. Session Description Format . . . . . . . . . . . . . . . 9 3.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10 3.4.1. ICE Candidate Trickling . . . . . . . . . . . . . . . 10 - 3.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . . 11 + 3.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . 10 3.5. Interactions With Forking . . . . . . . . . . . . . . . . 11 - 3.5.1. Sequential Forking . . . . . . . . . . . . . . . . . . 12 - 3.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . . 12 + 3.5.1. Sequential Forking . . . . . . . . . . . . . . . . . 11 + 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 - 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 + 4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 13 + 4.1. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 14 + 4.1.1. createOffer . . . . . . . . . . . . . . . . . . . . . 14 + 4.1.2. createAnswer . . . . . . . . . . . . . . . . . . . . 15 + 4.1.3. SessionDescriptionType . . . . . . . . . . . . . . . 15 + 4.1.3.1. Use of Provisional Answers . . . . . . . . . . . 16 + 4.1.3.2. Rollback . . . . . . . . . . . . . . . . . . . . 17 + 4.1.4. setLocalDescription . . . . . . . . . . . . . . . . . 17 + 4.1.5. setRemoteDescription . . . . . . . . . . . . . . . . 18 + 4.1.6. localDescription . . . . . . . . . . . . . . . . . . 18 + 4.1.7. remoteDescription . . . . . . . . . . . . . . . . . . 18 + 4.1.8. updateIce . . . . . . . . . . . . . . . . . . . . . . 19 + 4.1.9. addIceCandidate . . . . . . . . . . . . . . . . . . . 19 + 5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 19 + 5.1. SDP Requirements Overview . . . . . . . . . . . . . . . . 19 + 5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 21 + 5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 21 + 5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 25 + 5.2.3. Constraints Handling . . . . . . . . . . . . . . . . 26 + 5.2.3.1. OfferToReceiveAudio . . . . . . . . . . . . . . . 26 + 5.2.3.2. OfferToReceiveVideo . . . . . . . . . . . . . . . 27 + 5.2.3.3. VoiceActivityDetection . . . . . . . . . . . . . 27 + 5.2.3.4. IceRestart . . . . . . . . . . . . . . . . . . . 27 + 5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 27 + 5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 27 + 5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 31 + 5.3.3. Constraints Handling . . . . . . . . . . . . . . . . 31 + 5.4. Parsing an Offer . . . . . . . . . . . . . . . . . . . . 31 + 5.5. Parsing an Answer . . . . . . . . . . . . . . . . . . . . 31 + 5.6. Applying a Local Description . . . . . . . . . . . . . . 31 + 5.7. Applying a Remote Description . . . . . . . . . . . . . . 31 + + 6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 31 + 7. Security Considerations . . . . . . . . . . . . . . . . . . . 33 + 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33 + 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 33 + 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 33 + 10.1. Normative References . . . . . . . . . . . . . . . . . . 33 + 10.2. Informative References . . . . . . . . . . . . . . . . . 35 + Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 36 + A.1. Example API Flows . . . . . . . . . . . . . . . . . . . . 36 + A.1.1. Call using ROAP . . . . . . . . . . . . . . . . . . . 36 + A.1.2. Call using XMPP . . . . . . . . . . . . . . . . . . . 37 + A.1.3. Adding video to a call, using XMPP . . . . . . . . . 38 + A.1.4. Simultaneous add of video streams, using XMPP . . . . 39 + A.1.5. Call using SIP . . . . . . . . . . . . . . . . . . . 40 + A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using SIP 40 + A.2. Example Session Descriptions . . . . . . . . . . . . . . 41 + A.2.1. createOffer . . . . . . . . . . . . . . . . . . . . . 41 + A.2.2. createAnswer . . . . . . . . . . . . . . . . . . . . 43 + Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 44 + Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 45 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 @@ -228,25 +241,25 @@ 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. + 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 document are to be interpreted as described in [RFC2119]. 3. Semantics and Syntax 3.1. Signaling Model @@ -352,32 +369,32 @@ v | v | +---------------+ | +---------------+ | | |----/ | |----/ | | setLocal(PRANSWER) | | | Remote-Offer |------------------- >| Local-Pranswer| | | | | | | | | +---------------+ +---------------+ ^ | | | | setLocal(ANSWER) | -setRemote(OFFER) | | | + setRemote(OFFER) | | | V setLocal(ANSWER) | +---------------+ | | | | | | | | Stable |<---------------------------+ | | | | | | +---------------+ setRemote(ANSWER) | ^ | | | | setLocal(OFFER) | -setRemote(ANSWER)| | | + setRemote(ANSWER) | | | V | +---------------+ +---------------+ | | | | | | setRemote(PRANSWER) | | | Local-Offer |------------------- >|Remote-Pranswer| | | | | | |----\ | |----\ +---------------+ | +---------------+ | ^ | ^ | | | | | @@ -403,42 +420,50 @@ 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. + meantime, Javascript libraries can be used to perform these + manipulations. + + Note that most applications should be able to treat the + SessionDescriptions produced and consumed by these various API calls + as opaque blobs; that is, the application will not need to read or + change them. The W3C API will provide appropriate APIs to allow the + application to control various session parameters, which will provide + the necessary information to the browser about what sort of + SessionDescription to produce. 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. + in [I-D.ivov-mmusic-trickle-ice]. 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 initiating the + call. JSEP supports optional candidate trickling by providing APIs that provide control and feedback on the ICE candidate gathering process. Applications that support candidate trickling can send the initial offer immediately and send individual candidates when they get the notified of a new candidate; applications that do not support this feature can simply wait for the indication that gathering is complete, and then create and send their offer, with all the candidates, at this time. @@ -458,21 +483,21 @@ 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 + [RFC5888] , 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 @@ -564,104 +591,43 @@ 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. + possibly new DTLS-SRTP certificate fingerprints (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 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. - - 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 +4.1. Methods -4.2.1. createOffer +4.1.1. 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: @@ -671,79 +639,82 @@ o To trigger an ICE restart, for the purpose of reestablishing connectivity. 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. + generation of the SDP will follow the process defined for generating + an initial offer from the document that specifies the given SDP line. + The exact handling of initial offer generation is detailed in + Section 5.2.1. below. 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 + that specifies 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. + exchange. The exact handling of subsequent offer generation is + detailed in Section 5.2.2. below. 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 +4.1.2. 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. 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. + SDP line. The exact handling of answer generation is detailed in + Section 5.3. below. 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 +4.1.3. 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 @@ -767,49 +738,49 @@ 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 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 +4.1.3.1. 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 + physically 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 +4.1.3.2. 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 @@ -820,21 +791,21 @@ 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 +4.1.4. 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 @@ -849,121 +820,665 @@ 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 +4.1.5. setRemoteDescription The setRemoteDescription method instructs the PeerConnection to apply the supplied SDP blob as the desired remote configuration. As in setLocalDescription, the type field of the indicates how the blob should be processed. This API changes the local media state; among other things, it sets up local resources for sending and encoding media. If setRemoteDescription was 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 +4.1.6. localDescription The localDescription method returns a copy of the current local configuration, i.e. what was most recently passed to setLocalDescription, plus any local candidates that have been generated by the ICE Agent. 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 - +4.1.7. remoteDescription The remoteDescription method returns a copy of the current remote configuration, i.e. what was most recently passed to setRemoteDescription, plus any remote candidates that have been supplied via processIceMessage. 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 +4.1.8. updateIce The updateIce method allows the configuration of the ICE Agent to be changed during the session, primarily for changing which types of local candidates are provided to the application and used for connectivity checks. A callee may initially configure the ICE Agent to use only relay candidates, to avoid leaking location information, but update this configuration to use all candidates once the call is accepted. Regardless of the configuration, the gathering process collects all available candidates, but excluded candidates will not be surfaced in onicecandidate callback 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. -4.2.9. addIceCandidate +4.1.9. 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] + creating and parsing SDP objects. -5.1. Constructing an Offer +5.1. SDP Requirements Overview + The key specifications that govern creation and processing of offers + and answers are listed below. This list is derived from + [I-D.ietf-rtcweb-rtp-usage]. -5.2. Generating an Answer + R-1 [RFC4566] is the base SDP specification and MUST be + implemented. -5.3. Parsing an Offer + 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. -5.4. Parsing an Answer + R-3 [RFC5124] MUST be supported for signaling RTP/SAVPF RTP + profile. -5.5. Applying a Local Description + R-4 [RFC4585] MUST be implemented to signal RTCP based feedback. -5.6. Applying a Remote Description + 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. + + R-9 [RFC5763] MUST be implemented to signal DTLS certificate + fingerprints. + + 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] MUST NOT be implemented to signal SDES SRTP keying + information. + + R-13 A [I-D.ietf-mmusic-msid] MUST be supported, in order to signal + associations between RTP objects and W3C MediaStreams and + MediaStreamTracks in a standard way. + + R-14 The bundle mechanism in + [I-D.ietf-mmusic-sdp-bundle-negotiation] 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. + + 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?] + +5.2. Constructing an Offer + + When createOffer is called, a new SDP description must be created + that includes the functionality specified in + [I-D.ietf-rtcweb-rtp-usage]. The exact details of this process are + explained below. + +5.2.1. Initial Offers + + When createOffer is called for the first time, the result is known as + the initial offer. + + The first step in generating an initial offer is to generate session- + level attributes, as specified in [RFC4566], Section 5. + Specifically: + + o The first SDP line MUST be "v=0", as specified in [RFC4566], + Section 5.1 + + o The second SDP line MUST be an "o=" line, as specified in + [RFC4566], Section 5.2. The value of the field SHOULD + be "-". The value of the field SHOULD be a + cryptographically random number. To ensure uniqueness, this + number SHOULD be at least 64 bits long. The value of the field SHOULD be zero. The value of the + tuple SHOULD be set to a non- + meaningful address, such as IN IP4 0.0.0.0, to prevent leaking the + local address in this field. As mentioned in [RFC4566], the + entire o= line needs to be unique, but selecting a random number + for is sufficient to accomplish this. + + o The third SDP line MUST be a "s=" line, as specified in [RFC4566], + Section 5.3; a single space SHOULD be used as the session name, + e.g. "s= " + + o Session Information ("i="), URI ("u="), Email Address ("e="), + Phone Number ("p="), Bandwidth ("b="), Repeat Times ("r="), and + Time Zones ("z=") lines are not useful in this context and SHOULD + NOT be included. + + o Encryption Keys ("k=") lines do not provide sufficient security + and MUST NOT be included. + + o A "t=" line MUST be added, as specified in [RFC4566], Section 5.9; + both and SHOULD be set to zero, e.g. "t=0 + 0". + + The next step is to generate m= sections for each MediaStreamTrack + that has been added to the PeerConnection via the addStream method. + Note that this method takes a MediaStream, which can contain multiple + MediaStreamTracks, and therefore multiple m= sections can be + generated even if addStream is only called once. + + Each m= section should be generated as specified in [RFC4566], + Section 5.14. The field MUST be set to "RTP/SAVPF". If a m= + section is not being bundled into another m= section, it MUST + generate a unique set of ICE credentials and gather its own set of + candidates. Otherwise, it MUST use the same ICE credentials and + candidates that were used in the m= section that it is being bundled + into. For DTLS, all m= sections MUST use the same certificate [OPEN + ISSUE: how this is configured] and will therefore have the same + fingerprint values. + + Each m= section MUST include the following: + + o An "a=mid" line, as specified in [RFC5888], Section 4. + + o An "a=msid" line, as specified in [I-D.ietf-mmusic-msid], + Section 2. + + o [OPEN ISSUE: Use of App Token versus stream-correlator ] + + o An "a=sendrecv" line, as specified in [RFC3264], Section 5.1. + + o For each supported codec, "a=rtpmap" and "a=fmtp" lines, as + specified in [RFC4566], Section 6. For audio, the codecs + specified in [I-D.ietf-rtcweb-audio], Section 3, MUST be be + supported. + + o For each primary codec where RTP retransmission should be used, a + corresponding "a=rtpmap" line indicating "rtx" with the clock rate + of the primary codec and an "a=fmtp" line that references the + payload type fo the primary codec, as specified in [RFC4588], + Section 8.1. + + o For each supported FEC mechanism, a corresponding "a=rtpmap" line + indicating the desired FEC codec. + + o "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245], + Section 15.4. + + o An "a=ice-options" line, with the "trickle" option, as specified + in [I-D.ivov-mmusic-trickle-ice], Section 4. + + o For each candidate that has been gathered during the most recent + gathering phase, an "a=candidate" line, as specified in [RFC5245], + Section 4.3., paragraph 3. + + o For the current default candidate, a "c=" line, as specific in + [RFC5245], Section 4.3., paragraph 6. [OPEN ISSUE, pending + resolution in mmusic: If no candidates have yet been gathered yet, + the default candidate should be set to the null value defined in + [I-D.ivov-mmusic-trickle-ice], Section 5.1.] + + o An "a=fingerprint" line, as specified in [RFC4572], Section 5. + Use of the SHA-256 algorithm for the fingerprint is REQUIRED; if + the browser also supports stronger hashes, additional + "a=fingerprint" lines with these hashes MAY also be added. + + o An "a=setup" line, as specified in [RFC4145], Section 4, and + clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5. + The role value in the offer MUST be "actpass". + + o An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.1. + + o An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5. + + o For each supported RTP header extension, an "a=extmap" line, as + specified in [RFC5285], Section 5. The list of header extensions + that SHOULD/MUST be supported is specified in + [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header extensions + that require encryption MUST be specified as indicated in + [RFC6904], Section 4. + + o For each supported RTCP feedback mechanism, an "a=rtcp-fb" + mechanism, as specified in [RFC4585], Section 4.2. The list of + RTCP feedback mechanisms that SHOULD/MUST be supported is + specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.1. + + o An "a=ssrc" line, as specified in [RFC5576], Section 4.1, + indicating the SSRC to be used for sending media. + + o If RTX is supported for this media type, another "a=ssrc" line + with the RTX SSRC, and an "a=ssrc-group" line, as specified in + [RFC5576], section 4.2, with semantics set to "FID" and including + the primary and RTX SSRCs. + + o If FEC is supported for this media type, another "a=ssrc" line + with the FEC SSRC, and an "a=ssrc-group" line, as specified in + [RFC5576], section 4.2, with semantics set to "FEC" and including + the primary and FEC SSRCs. + + o [OPEN ISSUE: Handling of a=imageattr] + + o [TODO: bundle-only] + + Lastly, if a data channel has been created, a m= section MUST be + generated for data. The field MUST be set to "application" + and the field MUST be set to "DTLS/SCTP", as specified in + [I-D.ietf-mmusic-sctp-sdp], Section 3. The "a=mid", "a=ice-ufrag", + "a=ice-passwd", "a=ice-options", "a=candidate", "a=fingerprint", and + "a=setup" lines MUST be included as mentioned above. [OPEN ISSUE: + additional SCTP-specific stuff to be included, as indicated in + [I-D.jesup-rtcweb-data-protocol] (currently none)] + + Once all m= sections have been generated, a session-level "a=group" + attribute MUST be added as specified in [RFC5888]. This attribute + MUST have semantics "BUNDLE", and identify the m= sections to be + bundled. [OPEN ISSUE: Need to determine exactly how this decision is + made.] + + Attributes that are common between all m= sections MAY be moved to + session-level, if desired. + + Attributes other than the ones specified above MAY be included, + except for the following attributes which are specifically + incompatible with the requirements of [I-D.ietf-rtcweb-rtp-usage], + and MUST NOT be included: + + o "a=crypto" + + o "a=key-mgmt" + + o "a=ice-lite" + + Note that when BUNDLE is used, any additional attributes that are + added MUST follow the advice in + [I-D.nandakumar-mmusic-sdp-mux-attributes] on how those attributes + interact with BUNDLE. + +5.2.2. Subsequent Offers + + When createOffer is called a second (or later) time, the processing + is different, depending on the current signaling state. + + If the initial offer was not applied using setLocalDescription, + meaning the PeerConnection is still in the "stable" state, the steps + for generating an initial offer should be followed, with this + exception: + + o The "o=" line MUST stay the same. + + If the initial offer was applied using setLocalDescription, but an + answer from the remote side has not yet been applied, meaning the + PeerConnection is still in the "local-offer" state, the steps for + generating an initial offer should be followed, with these + exceptions: + + o The "o=" line MUST stay the same, except for the + field, which MUST increase by 1 from the previously applied local + description. + + o The "s=" and "t=" lines MUST stay the same. + + o Each "a=mid" line MUST stay the same. + + o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same. + + o For MediaStreamTracks that are still present, the "a=msid", + "a=ssrc", and "a=ssrc-group" lines MUST stay the same. + + o If any MediaStreamTracks have been removed, either through the + removeStream method or by removing them from an added MediaStream, + their m= sections MUST be marked as recvonly by changing the value + of the [RFC3264] directional attribute to "a=recvonly". The + "a=msid", "a=ssrc", and "a=ssrc-group" lines MUST be removed from + the associated m= sections. + + If the initial offer was applied using setLocalDescription, and an + answer from the remote side has been applied using + setRemoteDescription, meaning the PeerConnection is in the "remote- + pranswer" or "stable" states, an offer is generated based on the + negotiated session descriptions by following the steps mentioned for + the "local-offer" state above, along with these exceptions: [OPEN + ISSUE: should this be permitted in the remote-pranswer state?] + + o If a m= section was rejected, i.e. has had its port set to zero in + either the local or remote description, it MUST remain rejected + and have a zero port in the new offer, as indicated in RFC3264, + Section 5.1. + + o If a m= section exists in the current local description, but has + its state set to inactive or recvonly, and a new MediaStreamTrack + is added, the previously existing m= section MUST be recycled + instead of creating a new m= section. [OPEN ISSUE: Nail down + exactly what this means. Should the codecs remain the same? + (No.) Should ICE restart? (No.) Can the "a=mid" attribute be + changed? (Yes?)] + + o If a m= section exists in the current local description, but does + not have an associated MediaStreamTrack (i.e. it is inactive or + recvonly), a corresponding m= section MUST be generated in the new + offer, but without "a=msid", "a=ssrc", or "a=ssrc-group" + attributes, and the appropriate directional attribute must be + specified. + + In addition, for each previously existing, non-rejected m= section in + the new offer, the following adjustments are made based on the + contents of the corresponding m= section in the current remote + description: + + o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST + only include codecs present in the remote description. + + o The RTP header extensions MUST only include those that are present + in the remote description. + + o The RTCP feedback extensions MUST only include those that are + present in the remote description. + + o The "a=rtcp-mux" line MUST only be added if present in the remote + description. + + o The "a=rtcp-rsize" line MUST only be added if present in the + remote description. + +5.2.3. Constraints Handling + + The createOffer method takes as a parameter a MediaConstraints + object. Special processing is performed when generating a SDP + description if the following constraints are present. + +5.2.3.1. OfferToReceiveAudio + + If the "OfferToReceiveAudio" constraint is specified, with a value of + "true", the offer MUST include a non-rejected m= section with media + type "audio", even if no audio MediaStreamTrack has been added to the + PeerConnection. This allows the offerer to receive audio even when + not sending it; accordingly, the directional attribute on the audio + m= section MUST be set to recvonly. If this constraint is specified + when an audio MediaStreamTrack has already been added to the + PeerConnection, or a non-rejected m= section with media type "audio" + previously existed, it has no effect. + +5.2.3.2. OfferToReceiveVideo + + If the "OfferToReceiveAudio" constraint is specified, with a value of + "true", the offer MUST include a m= section with media type "video", + even if no video MediaStreamTrack has been added to the + PeerConnection. This allows the offerer to receive video even when + not sending it; accordingly, the directional attribute on the video + m= section MUST be set to recvonly. If this constraint is specified + when an video MediaStreamTrack has already been added to the + PeerConnection, or a non-rejected m= section with media type "video" + previously existed, it has no effect. + +5.2.3.3. VoiceActivityDetection + + If the "VoiceActivityDetection" constraint is specified, with a value + of "true", the offer MUST indicate support for silence suppression by + including comfort noise ("CN") codecs for each supported clock rate, + as specified in [RFC3389], Section 5.1. [OPEN issue: should this do + anything in signaling, or should it just control built-in DTX modes + in audio codecs? Opus has built-in DTX, but G.711 does not.] + +5.2.3.4. IceRestart + + If the "IceRestart" constraint is specified, with a value of "true", + the offer MUST indicate an ICE restart by generating new ICE ufrag + and pwd attributes, as specified in RFC5245, Section 9.1.1.1. If + this constraint is specified on an initial offer, it has no effect + (since a new ICE ufrag and pwd are already generated). + +5.3. Generating an Answer + + When createAnswer is called, a new SDP description must be created + that is compatible with the supplied remote description as well as + the requirements specified in [I-D.ietf-rtcweb-rtp-usage]. The exact + details of this process are explained below. + +5.3.1. Initial Answers + + When createAnswer is called for the first time after a remote + description has been provided, the result is known as the initial + answer. If no remote description has been installed, an answer + cannot be generated, and an error MUST be returned. + + Note that the remote description SDP may not have been created by a + WebRTC endpoint and may not conform to all the requirements listed in + Section 5.2. For many cases, this is not a problem. However, if any + mandatory SDP attributes are missing, or functionality listed as + mandatory-to-use is not present (e.g. ICE, DTLS) [TODO: find + reference for this], this MUST be treated as an error. [OPEN ISSUE: + Should this cause setRemoteDescription to fail, or should this cause + createAnswer to reject those particular m= sections?] + + The first step in generating an initial answer is to generate + session-level attributes. The process here is identical to that + indicated in the Initial Offers section above, with the addition that + + The next step is to generate m= sections for each m= section that is + present in the remote offer, as specified in [RFC3264], Section 6. + For the purposes of this discussion, any session-level attributes in + the offer that are also valid as media-level attributes SHALL be + considered to be present in each m= section. + + If any of the offered m= sections have been rejected, by stopping the + associated remote MediaStreamTrack, the corresponding m= section in + the answer MUST be marked as rejected by setting the port in the m= + line to zero, as indicated in [RFC3264], Section 6., and processing + continues with the next m= section. + + For each non-rejected m= section of a given media type, if there is a + local MediaStreamTrack of the specified type which has been added to + the PeerConnection via addStream and not yet associated with a m= + section, the MediaStreamTrack is associated with the m= section at + this time. If there are more m= sections of a certain type than + MediaStreamTracks, some m= sections will not have an associated + MediaStreamTrack. If there are more MediaStreamTracks of a certain + type than m= sections, only the first N MediaStreamTracks will be + able to be associated in the constructed answer. The remainder will + need to be associated in a subsequent offer. + + Each m= section should then generated as specified in [RFC3264], + Section 6.1. The field MUST be set to "RTP/SAVPF". If the + offer supports BUNDLE, all m= sections to be BUNDLEd must use the + same ICE credentials and candidates; all m= sections not being + BUNDLEd must use unique ICE credentials and candidates. Each m= + section MUST include the following: + + o If present in the offer, an "a=mid" line, as specified in + [RFC5888], Section 9.1. The "mid" value MUST match that specified + in the offer. + + o If a local MediaStreamTrack has been associated, an "a=msid" line, + as specified in [I-D.ietf-mmusic-msid], Section 2. + + o [OPEN ISSUE: Use of App Token versus stream-correlator ] + + o If a local MediaStreamTrack has been associated, an "a=sendrecv" + line, as specified in [RFC3264], Section 6.1. If no local + MediaStreamTrack has been associated, an "a=recvonly" line. + [TODO: handle non-sendrecv offered m= sections] + + o For each supported codec that is present in the offer, "a=rtpmap" + and "a=fmtp" lines, as specified in [RFC4566], Section 6, and + [RFC3264], Section 6.1. For audio, the codecs specified in + [I-D.ietf-rtcweb-audio], Section 3, MUST be be supported. Note + that for simplicity, the answerer MAY use different payload types + for codecs than the offerer, as it is not prohibited by + Section 6.1. + + o If "rtx" is present in the offer, for each primary codec where RTP + retransmission should be used, a corresponding "a=rtpmap" line + indicating "rtx" with the clock rate of the primary codec and an + "a=fmtp" line that references the payload type fo the primary + codec, as specified in [RFC4588], Section 8.1. + + o For each supported FEC mechanism that is present in the offer, a + corresponding "a=rtpmap" line indicating the desired FEC codec. + + o "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245], + Section 15.4. + + o If the "trickle" ICE option is present in the offer, an "a=ice- + options" line, with the "trickle" option, as specified in + [I-D.ivov-mmusic-trickle-ice], Section 4. + + o For each candidate that has been gathered during the most recent + gathering phase, an "a=candidate" line, as specified in [RFC5245], + Section 4.3., paragraph 3. + + o For the current default candidate, a "c=" line, as specific in + [RFC5245], Section 4.3., paragraph 6. [OPEN ISSUE, pending + resolution in mmusic: If no candidates have yet been gathered yet, + the default candidate should be set to the null value defined in + [I-D.ivov-mmusic-trickle-ice], Section 5.1.] + + o An "a=fingerprint" line, as specified in [RFC4572], Section 5. + Use of the SHA-256 algorithm for the fingerprint is REQUIRED; if + the browser also supports stronger hashes, additional + "a=fingerprint" lines with these hashes MAY also be added. + + o An "a=setup" line, as specified in [RFC4145], Section 4, and + clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5. + The role value in the answer MUST be "active" or "passive"; the + "active" role is RECOMMENDED. + + o If present in the offer, an "a=rtcp-mux" line, as specified in + [RFC5761], Section 5.1.1. + + o If present in the offer, an "a=rtcp-rsize" line, as specified in + [RFC5506], Section 5. + + o For each supported RTP header extension that is present in the + offer, an "a=extmap" line, as specified in [RFC5285], Section 5. + The list of header extensions that SHOULD/MUST be supported is + specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header + extensions that require encryption MUST be specified as indicated + in [RFC6904], Section 4. + + o For each supported RTCP feedback mechanism that is present in the + offer, an "a=rtcp-fb" mechanism, as specified in [RFC4585], + Section 4.2. The list of RTCP feedback mechanisms that SHOULD/ + MUST be supported is specified in [I-D.ietf-rtcweb-rtp-usage], + Section 5.1. + + o If a local MediaStreamTrack has been associated, an "a=ssrc" line, + as specified in [RFC5576], Section 4.1, indicating the SSRC to be + used for sending media. + + o If a local MediaStreamTrack has been associated, and RTX has been + negotiated for this m= section, another "a=ssrc" line with the RTX + SSRC, and an "a=ssrc-group" line, as specified in [RFC5576], + section 4.2, with semantics set to "FID" and including the primary + and RTX SSRCs. + + o If a local MediaStreamTrack has been associated, and FEC has been + negotiated for this m= section, another "a=ssrc" line with the FEC + SSRC, and an "a=ssrc-group" line, as specified in [RFC5576], + section 4.2, with semantics set to "FEC" and including the primary + and FEC SSRCs. + + o [OPEN ISSUE: Handling of a=imageattr] + + o [TODO: bundle-only] + If a data channel m= section has been offered, a m= section MUST also + be generated for data. The field MUST be set to + "application" and the field MUST be set to "DTLS/SCTP", as + specified in [I-D.ietf-mmusic-sctp-sdp], Section 3. The "a=mid", "a + =ice-ufrag", "a=ice-passwd", "a=ice-options", "a=candidate", + "a=fingerprint", and "a=setup" lines MUST be included as mentioned + above. [OPEN ISSUE: additional SCTP-specific stuff to be included, + as indicated in [I-D.jesup-rtcweb-data-protocol] (currently none)] + + [TODO: processing of BUNDLE group] + + Attributes that are common between all m= sections MAY be moved to + session-level, if desired. + + The attributes prohibited in creation of offers are also prohibited + in the creation of answers. + +5.3.2. Subsequent Answers + +5.3.3. Constraints Handling + +5.4. Parsing an Offer + +5.5. Parsing an Answer + +5.6. Applying a Local Description + +5.7. 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. + 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: 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 @@ -966,27 +1481,34 @@ 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] + 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. @@ -1018,127 +1540,163 @@ 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. + [I-D.ietf-mmusic-msid] + Alvestrand, H., "Cross Session Stream Identification in + the Session Description Protocol", draft-ietf-mmusic- + msid-01 (work in progress), August 2013. + + [I-D.ietf-mmusic-sctp-sdp] + Loreto, S. and G. Camarillo, "Stream Control Transmission + Protocol (SCTP)-Based Media Transport in the Session + Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-04 + (work in progress), June 2013. + + [I-D.ietf-mmusic-sdp-bundle-negotiation] + Holmberg, C., Alvestrand, H., and C. Jennings, + "Multiplexing Negotiation Using Session Description + Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp- + bundle-negotiation-04 (work in progress), June 2013. + + [I-D.ietf-rtcweb-audio] + Valin, J. and C. Bran, "WebRTC Audio Codec and Processing + Requirements", draft-ietf-rtcweb-audio-02 (work in + progress), August 2013. + + [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-09 (work in progress), + September 2013. + + [I-D.nandakumar-mmusic-sdp-mux-attributes] + Nandakumar, S., "A Framework for SDP Attributes when + Multiplexing", draft-nandakumar-mmusic-sdp-mux- + attributes-03 (work in progress), July 2013. [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. + with Session Description Protocol (SDP)", RFC 3264, June + 2002. - [RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing - Tone Generation in the Session Initiation Protocol (SIP)", - RFC 3960, December 2004. + [RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in + the Session Description Protocol (SDP)", RFC 4145, + September 2005. [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006. + [RFC4572] Lennox, J., "Connection-Oriented Media Transport over the + Transport Layer Security (TLS) Protocol in the Session + Description Protocol (SDP)", RFC 4572, 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. + 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. + Traversal for Offer/Answer Protocols", RFC 5245, April + 2010. + + [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP + Header Extensions", RFC 5285, July 2008. [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. + [RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure + Real-time Transport Protocol (SRTP)", RFC 6904, April + 2013. - [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. +10.2. Informative References - [I-D.ietf-rtcweb-rtp-usage] - Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time - Communication (WebRTC): Media Transport and Use of RTP", - draft-ietf-rtcweb-rtp-usage-04 (work in progress), - July 2012. + [I-D.ivov-mmusic-trickle-ice] + Ivov, E., Rescorla, E., and J. Uberti, "Trickle ICE: + Incremental Provisioning of Candidates for the Interactive + Connectivity Establishment (ICE) Protocol", draft-ivov- + mmusic-trickle-ice-01 (work in progress), March 2013. [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. + Answer Protocol (ROAP)", draft-jennings-rtcweb- + signaling-01 (work in progress), October 2011. + + [I-D.jesup-rtcweb-data-protocol] + Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel + Protocol", draft-jesup-rtcweb-data-protocol-04 (work in + progress), February 2013. [I-D.nandakumar-rtcweb-sdp] Nandakumar, S. and C. Jennings, "SDP for the WebRTC", - draft-nandakumar-rtcweb-sdp-00 (work in progress), - October 2012. + draft-nandakumar-rtcweb-sdp-02 (work in progress), July + 2013. + + [RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for + Comfort Noise (CN)", RFC 3389, September 2002. [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth - Modifiers for RTP Control Protocol (RTCP) Bandwidth", - RFC 3556, July 2003. + Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC + 3556, July 2003. + + [RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing + Tone Generation in the Session Initiation Protocol (SIP)", + RFC 3960, December 2004. [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session Description Protocol (SDP) Security Descriptions for Media Streams", RFC 4568, July 2006. + [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. + Hakenberg, "RTP Retransmission Payload Format", RFC 4588, + 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, + 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 @@ -1356,49 +1913,234 @@ 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 +A.2. Example Session Descriptions + +A.2.1. createOffer + + This SDP shows a typical initial offer, created by createOffer for a + PeerConnection with a single audio MediaStreamTrack, a single video + MediaStreamTrack, and a single data channel. Host candidates have + also already been gathered. Note some lines have been broken into + two lines for formatting reasons. + + v=0 + o=- 4962303333179871722 1 IN IP4 0.0.0.0 + s=- + t=0 0 + a=group:BUNDLE audio video data + m=audio 56500 RTP/SAVPF 111 0 8 126 + c=IN IP4 192.0.2.1 + a=rtcp:56501 IN IP4 192.0.2.1 + a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56500 + typ host generation 0 + a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56501 + typ host generation 0 + a=ice-ufrag:ETEn1v9DoTMB9J4r + a=ice-pwd:OtSK0WpNtpUjkY4+86js7ZQl + a=ice-options:trickle + a=mid:audio + a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level + a=sendrecv + a=rtcp-mux + a=rtcp-rsize + a=fingerprint:sha-256 + 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04 + :BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 + a=setup:actpass + a=rtpmap:111 opus/48000/2 + a=fmtp:111 minptime=10 + a=rtpmap:0 PCMU/8000 + a=rtpmap:8 PCMA/8000 + a=rtpmap:126 telephone-event/8000 + a=maxptime:60 + a=ssrc:1732846380 cname:EocUG1f0fcg/yvY7 + a=msid:47017fee-b6c1-4162-929c-a25110252400 + f83006c5-a0ff-4e0a-9ed9-d3e6747be7d9 + m=video 56502 RTP/SAVPF 100 115 116 117 + c=IN IP4 192.0.2.1 + a=rtcp:56503 IN IP4 192.0.2.1 + a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56502 + typ host generation 0 + a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56503 + typ host generation 0 + a=ice-ufrag:BGKkWnG5GmiUpdIV + a=ice-pwd:mqyWsAjvtKwTGnvhPztQ9mIf + a=ice-options:trickle + a=mid:video + a=extmap:2 urn:ietf:params:rtp-hdrext:toffset + a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time + a=sendrecv + a=rtcp-mux + a=rtcp-rsize + a=fingerprint:sha-256 + 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04 + :BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 + a=setup:actpass + a=rtpmap:100 VP8/90000 + a=rtcp-fb:100 ccm fir + a=rtcp-fb:100 nack + a=rtcp-fb:100 goog-remb + a=rtpmap:115 rtx/90000 + a=fmtp:115 apt=100 + a=rtpmap:116 red/90000 + a=rtpmap:117 ulpfec/90000 + a=ssrc:1366781083 cname:EocUG1f0fcg/yvY7 + a=ssrc:1366781084 cname:EocUG1f0fcg/yvY7 + a=ssrc:1366781085 cname:EocUG1f0fcg/yvY7 + a=ssrc-group:FID 1366781083 1366781084 + a=ssrc-group:FEC 1366781083 1366781085 + a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae + f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0 + m=application 56504 DTLS/SCTP 5000 + c=IN IP4 192.0.2.1 + a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56504 + typ host generation 0 + a=ice-ufrag:VD5v2BnbZm3mgP3d + a=ice-pwd:+Jlkuox+VVIUDqxcfIDuTZMH + a=ice-options:trickle + a=mid:data + a=fingerprint:sha-256 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04 + :BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 + a=setup:actpass + a=fmtp:5000 protocol=webrtc-datachannel; streams=10 + +A.2.2. createAnswer + + This SDP shows a typical initial answer to the above offer, created + by createAnswer for a PeerConnection with a single audio + MediaStreamTrack, a single video MediaStreamTrack, and a single data + channel. Host candidates have also already been gathered. Note some + lines have been broken into two lines for formatting reasons. + + v=0 + o=- 6729291447651054566 1 IN IP4 0.0.0.0 + s=- + t=0 0 + a=group:BUNDLE audio video data + m=audio 20000 RTP/SAVPF 111 0 8 126 + c=IN IP4 192.0.2.2 + a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000 + typ host generation 0 + a=ice-ufrag:6sFvz2gdLkEwjZEr + a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2 + a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35 + :DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08 + a=setup:active + a=mid:audio + a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level + a=sendrecv + a=rtcp-mux + a=rtpmap:111 opus/48000/2 + a=fmtp:111 minptime=10 + a=rtpmap:0 PCMU/8000 + a=rtpmap:8 PCMA/8000 + a=rtpmap:126 telephone-event/8000 + a=maxptime:60 + a=ssrc:3429951804 cname:Q/NWs1ao1HmN4Xa5 + a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1 + PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1a0 + m=video 20000 RTP/SAVPF 100 115 116 117 + c=IN IP4 192.0.2.2 + a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000 + typ host generation 0 + a=ice-ufrag:6sFvz2gdLkEwjZEr + a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2 + a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35 + :DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08 + a=setup:active + a=mid:video + a=extmap:2 urn:ietf:params:rtp-hdrext:toffset + a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time + a=sendrecv + a=rtcp-mux + a=rtpmap:100 VP8/90000 + a=rtcp-fb:100 ccm fir + a=rtcp-fb:100 nack + a=rtcp-fb:100 goog-remb + a=rtpmap:115 rtx/90000 + a=fmtp:115 apt=100 + a=rtpmap:116 red/90000 + a=rtpmap:117 ulpfec/90000 + a=ssrc:3229706345 cname:Q/NWs1ao1HmN4Xa5 + a=ssrc:3229706346 cname:Q/NWs1ao1HmN4Xa5 + a=ssrc:3229706347 cname:Q/NWs1ao1HmN4Xa5 + a=ssrc-group:FID 3229706345 3229706346 + a=ssrc-group:FEC 3229706345 3229706347 + a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1 + PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1v0 + m=application 20000 DTLS/SCTP 5000 + c=IN IP4 192.0.2.2 + a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000 + typ host generation 0 + a=ice-ufrag:6sFvz2gdLkEwjZEr + a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2 + a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35 + :DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08 + a=setup:active + a=mid:data + a=fmtp:5000 protocol=webrtc-datachannel; streams=10 + Appendix B. Change log + Changes in draft-04: + + o Filled in sections on createOffer and createAnswer. + + o Added SDP examples. + + o Fixed references. 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 Removed stuff that has moved to W3C specification + o Align SDP handling with W3C draft + o Clarified section on forking. Changes in draft -01: + o Added diagrams for architecture and state machine. + o Added sections on forking and rehydration. + o Clarified meaning of "pranswer" and "answer". + o Reworked how ICE restarts and media directions are controlled. + o Added list of parameters that can be changed in a description. + o Updated suggested API and examples to match latest thinking. + o Suggested API and examples have been moved to an appendix. Changes in draft -00: o Migrated from draft-uberti-rtcweb-jsep-02. Authors' Addresses - Justin Uberti Google 747 6th Ave S Kirkland, WA 98033 USA Email: justin@uberti.name Cullen Jennings Cisco