3.4.1. RtpTransceivers

RtpTransceivers allow the application to control the RTP media associated with one "m=" section. Each RtpTransceiver has an RtpSender and an RtpReceiver, which an application can use to control the sending and receiving of RTP media. The application may also modify the RtpTransceiver directly, for instance, by stopping it.¶

RtpTransceivers generally have a 1:1 mapping with "m=" sections, although there may be more RtpTransceivers than "m=" sections when RtpTransceivers are created but not yet associated with an "m=" section, or if RtpTransceivers have been stopped and disassociated from "m=" sections. An RtpTransceiver is said to be associated with an "m=" section if its media identification (mid) property is non-null; otherwise, it is said to be disassociated. The associated "m=" section is determined using a mapping between transceivers and "m=" section indices, formed when creating an offer or applying a remote offer.¶

An RtpTransceiver is never associated with more than one "m=" section, and once a session description is applied, an "m=" section is always associated with exactly one RtpTransceiver. However, in certain cases where an "m=" section has been rejected, as discussed in Section 5.2.2 below, that "m=" section will be "recycled" and associated with a new RtpTransceiver with a new MID value.¶

RtpTransceivers can be created explicitly by the application or implicitly by calling setRemoteDescription with an offer that adds new "m=" sections.¶

3.4.2. RtpSenders

RtpSenders allow the application to control how RTP media is sent. An RtpSender is conceptually responsible for the outgoing RTP stream(s) described by an "m=" section. This includes encoding the attached MediaStreamTrack, sending RTP media packets, and generating/processing the RTP Control Protocol (RTCP) for the outgoing RTP streams(s).¶

3.4.3. RtpReceivers

RtpReceivers allow the application to inspect how RTP media is received. An RtpReceiver is conceptually responsible for the incoming RTP stream(s) described by an "m=" section. This includes processing received RTP media packets, decoding the incoming stream(s) to produce a remote MediaStreamTrack, and generating/processing RTCP for the incoming RTP stream(s).¶

3.5.1. ICE Gathering Overview

JSEP gathers ICE candidates as needed by the application. Collection of ICE candidates is referred to as a gathering phase, and this is triggered either by the addition of a new or recycled "m=" section to the local session description or by new ICE credentials in the description, indicating an ICE restart. Use of new ICE credentials can be triggered explicitly by the application or implicitly by the JSEP implementation in response to changes in the ICE configuration.¶

When the ICE configuration changes in a way that requires a new gathering phase, a 'needs-ice-restart' bit is set. When this bit is set, calls to the createOffer API will generate new ICE credentials. This bit is cleared by a call to the setLocalDescription API with new ICE credentials from either an offer or an answer, i.e., from either a locally or remotely initiated ICE restart.¶

When a new gathering phase starts, the ICE agent will notify the application that gathering is occurring through a state change event. Then, when each new ICE candidate becomes available, the ICE agent will supply it to the application via an onicecandidate event; these candidates will also automatically be added to the current and/or pending local session description. Finally, when all candidates have been gathered, a final onicecandidate event will be dispatched to signal that the gathering process is complete.¶

Note that gathering phases only gather the candidates needed by new/recycled/restarting "m=" sections; other "m=" sections continue to use their existing candidates. Also, if an "m=" section is bundled (either by a successful bundle negotiation or by being marked as bundle-only), then candidates will be gathered and exchanged for that "m=" section if and only if its MID item is a BUNDLE-tag, as described in [RFC8843].¶

3.5.2. 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 [RFC8838]. 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 media setup in cases where gathering is not performed prior to initiating the call.¶

JSEP supports optional candidate trickling by providing APIs, as described above, 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 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 that time.¶

Upon receipt of trickled candidates, the receiving application will supply them to its ICE agent. This triggers the ICE agent to start using the new remote candidates for connectivity checks.¶

candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host

3.5.3. ICE Candidate Policy

Typically, when gathering ICE candidates, the JSEP implementation will gather all possible forms of initial candidates -- host, server-reflexive, and relay. However, in certain cases, applications may want to have more specific control over the gathering process, due to privacy or related concerns. For example, one may want to only use relay candidates, to leak as little location information as possible (keeping in mind that this choice comes with corresponding operational costs). To accomplish this, JSEP allows the application to restrict which ICE candidates are used in a session. Note that this filtering is applied on top of any restrictions the implementation chooses to enforce regarding which IP addresses are permitted for the application, as discussed in [RFC8828].¶

There may also be cases where the application wants to change which types of candidates are used while the session is active. A prime example is where a callee may initially want to use only relay candidates, to avoid leaking location information to an arbitrary caller, but then change to use all candidates (for lower operational cost) once the user has indicated that they want to take the call. For this scenario, the JSEP implementation MUST allow the candidate policy to be changed in mid-session, subject to the aforementioned interactions with local policy.¶

To administer the ICE candidate policy, the JSEP implementation will determine the current setting at the start of each gathering phase. Then, during the gathering phase, the implementation MUST NOT expose candidates disallowed by the current policy to the application, use them as the source of connectivity checks, or indirectly expose them via other fields, such as the raddr/rport attributes for other ICE candidates. Later, if a different policy is specified by the application, the application can apply it by kicking off a new gathering phase via an ICE restart.¶

3.5.4. ICE Candidate Pool

JSEP applications typically inform the JSEP implementation to begin ICE gathering via the information supplied to setLocalDescription, as the local description indicates the number of ICE components that will be needed and for which candidates must be gathered. However, to accelerate cases where the application knows the number of ICE components to use ahead of time, it may ask the implementation to gather a pool of potential ICE candidates to help ensure rapid media setup.¶

When setLocalDescription is eventually called and the JSEP implementation prepares to gather the needed ICE candidates, it SHOULD start by checking if any candidates are available in the pool. If there are candidates in the pool, they SHOULD be handed to the application immediately via the ICE candidate event. If the pool becomes depleted, either because a larger-than-expected number of ICE components are used or because the pool has not had enough time to gather candidates, the remaining candidates are gathered as usual. This only occurs for the first offer/answer exchange, after which the candidate pool is emptied and no longer used.¶

One example of where this concept is useful is an application that expects an incoming call at some point in the future, and wants to minimize the time it takes to establish connectivity, to avoid clipping of initial media. By pre-gathering candidates into the pool, it can exchange and start sending connectivity checks from these candidates almost immediately upon receipt of a call. Note, though, that by holding on to these pre-gathered candidates, which will be kept alive as long as they may be needed, the application will consume resources on the STUN/TURN servers it is using. ("STUN" stands for "Session Traversal Utilities for NAT".)¶

3.6.1. Creating an imageattr Attribute

The receiver will first combine any known local limits (e.g., hardware decoder capabilities or local policy) to determine the absolute minimum and maximum sizes it can receive. If there are no known local limits, the "a=imageattr" attribute SHOULD be omitted. If these local limits preclude receiving any video, i.e., the degenerate case of no permitted resolutions, the "a=imageattr" attribute MUST be omitted, and the "m=" section MUST be marked as sendonly/inactive, as appropriate.¶

Otherwise, an "a=imageattr" attribute is created with a "recv" direction, and the resulting resolution space formed from the aforementioned intersection is used to specify its minimum and maximum "x=" and "y=" values.¶

The rules here express a single set of preferences, and therefore, the "a=imageattr" "q=" value is not important. It SHOULD be set to "1.0".¶

The "a=imageattr" field is payload type specific. When all video codecs supported have the same capabilities, use of a single attribute, with the wildcard payload type (*), is RECOMMENDED. However, when the supported video codecs have different limitations, specific "a=imageattr" attributes MUST be inserted for each payload type.¶

As an example, consider a system with a multiformat video decoder, which is capable of decoding any resolution from 48x48 to 720p. In this case, the implementation would generate this attribute:¶

          a=imageattr:* recv [x=[48:1280],y=[48:720],q=1.0]

This declaration indicates that the receiver is capable of decoding any image resolution from 48x48 up to 1280x720 pixels.¶

3.6.2. Interpreting imageattr Attributes

[RFC6236] defines "a=imageattr" to be an advisory field. This means that it does not absolutely constrain the video formats that the sender can use but gives an indication of the preferred values.¶

This specification prescribes behavior that is more specific. When a MediaStreamTrack, which is producing video of a certain resolution (the "track resolution"), is attached to an RtpSender, which is encoding the track video at the same or lower resolution(s) (the "encoder resolutions"), and a remote description is applied that references the sender and contains valid "a=imageattr recv" attributes, it MUST follow the rules below to ensure that the sender does not transmit a resolution that would exceed the size criteria specified in the attributes. These rules MUST be followed as long as the attributes remain present in the remote description, including cases in which the track changes its resolution or is replaced with a different track.¶

Depending on how the RtpSender is configured, it may be producing a single encoding at a certain resolution or, if simulcast (Section 3.7) has been negotiated, multiple encodings, each at their own specific resolution. In addition, depending on the configuration, each encoding may have the flexibility to reduce resolution when needed or may be locked to a specific output resolution.¶

For each encoding being produced by the RtpSender, the set of "a=imageattr recv" attributes in the corresponding "m=" section of the remote description is processed to determine what should be transmitted. Only attributes that reference the media format selected for the encoding are considered; each such attribute is evaluated individually, starting with the attribute with the highest "q=" value. If multiple attributes have the same "q=" value, they are evaluated in the order they appear in their containing "m=" section. Note that while JSEP endpoints will include at most one "a=imageattr recv" attribute per media format, JSEP endpoints may receive session descriptions from non-JSEP endpoints with "m=" sections that contain multiple such attributes.¶

For each "a=imageattr recv" attribute, the following rules are applied. If this processing is successful, the encoding is transmitted accordingly, and no further attributes are considered for that encoding. Otherwise, the next attribute is evaluated, in the aforementioned order. If none of the supplied attributes can be processed successfully, the encoding MUST NOT be transmitted, and an error SHOULD be raised to the application.¶

• The limits from the attribute are compared to the encoder resolution. Only the specific limits mentioned below are considered; any other values, such as picture aspect ratio, MUST be ignored. When considering a MediaStreamTrack that is producing rotated video, the unrotated resolution MUST be used for the checks. This is required regardless of whether the receiver supports performing receive-side rotation (e.g., through Coordination of Video Orientation (CVO) [TS26.114]), as it significantly simplifies the matching logic.¶
• If the attribute includes a "sar=" (sample aspect ratio) value set to something other than "1.0", indicating that the receiver wants to receive non-square pixels, this cannot be satisfied and the attribute MUST NOT be used.¶
• If the encoder resolution exceeds the maximum size permitted by the attribute and the encoder is allowed to adjust its resolution, the encoder SHOULD apply downscaling in order to satisfy the limits. Downscaling MUST NOT change the picture aspect ratio of the encoding, ignoring any trivial differences due to rounding. For example, if the encoder resolution is 1280x720 and the attribute specified a maximum of 640x480, the expected output resolution would be 640x360. If downscaling cannot be applied, the attribute MUST NOT be used.¶
• If the encoder resolution is less than the minimum size permitted by the attribute, the attribute MUST NOT be used; the encoder MUST NOT apply upscaling. JSEP implementations SHOULD avoid this situation by allowing receipt of arbitrarily small resolutions, perhaps via fallback to a software decoder.¶
• If the encoder resolution is within the maximum and minimum sizes, no action is needed.¶

3.8.1. Sequential Forking

Sequential forking involves a call being dispatched to multiple remote callees, where each callee can accept the call, but only one active session ever exists at a time; no mixing of received media is performed.¶

JSEP handles sequential forking well, allowing the application to easily control the policy for selecting the desired remote endpoint. When an answer arrives from one of the callees, the application can choose to apply it as either (1) a provisional answer, leaving open the possibility of using a different answer in the future or (2) a final answer, ending the setup flow.¶

In a "first-one-wins" situation, the first answer will be applied as a final answer, and the application will reject any subsequent answers. In SIP parlance, this would be ACK + BYE.¶

In a "last-one-wins" situation, all answers would be applied as provisional answers, and any previous call leg will be terminated. At some point, the application will end the setup process, perhaps with a timer; at this point, the application could reapply the pending remote description as a final answer.¶

3.8.2. Parallel Forking

Parallel forking involves a call being dispatched to multiple remote callees, where each callee can accept the call and multiple simultaneous active signaling sessions can be established as a result. If multiple callees send media at the same time, the possibilities for handling this are described in [RFC3960], Section 3.1. Most SIP devices today only support exchanging media with a single device at a time and do not try to mix multiple early media audio sources, as that could result in a confusing situation. For example, consider having a European ringback tone mixed together with the North American ringback tone -- the resulting sound would not be like either tone and would confuse the user. If the signaling application wishes to only exchange media with one of the remote endpoints at a time, then from a media engine point of view, this is exactly like the sequential forking case.¶

In the parallel forking case where the JavaScript application wishes to simultaneously exchange media with multiple peers, the flow is slightly more complex, but the JavaScript application can follow the strategy that [RFC3960] describes, using UPDATE. 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 (see Section 4.1) and making sure that the same local media streams have been added into this new PeerConnection. Then the new PeerConnection object would produce an SDP offer that could be used by the signaling to perform the UPDATE strategy discussed in [RFC3960].¶

As a result of sharing the media streams, the application will end up with N parallel PeerConnection sessions, each with a local and remote description and their own local and remote addresses. The media flow from these sessions can be managed using setDirection (see Section 4.2.3), 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.¶

4.1.1. Constructor

The PeerConnection constructor allows the application to specify global parameters for the media session, such as the STUN/TURN servers and credentials to use when gathering candidates, as well as the initial ICE candidate policy and pool size, and also the bundle policy to use.¶

If an ICE candidate policy is specified, it functions as described in Section 3.5.3, causing the JSEP implementation to only surface the permitted candidates (including any implementation-internal filtering) to the application and only use those candidates for connectivity checks. The set of available policies is as follows:¶

all:

All candidates permitted by implementation policy will be gathered and used.¶

relay:

All candidates except relay candidates will be filtered out. This obfuscates the location information that might be ascertained by the remote peer from the received candidates. Depending on how the application deploys and chooses relay servers, this could obfuscate location to a metro or possibly even global level.¶

The default ICE candidate policy MUST be set to "all", as this is generally the desired policy and also typically reduces the use of application TURN server resources significantly.¶

If a size is specified for the ICE candidate pool, this indicates the number of ICE components to pre-gather candidates for. Because pre‑gathering results in utilizing STUN/TURN server resources for potentially long periods of time, this MUST only occur upon application request, and therefore the default candidate pool size MUST be zero.¶

The application can specify its preferred policy regarding use of bundle, the multiplexing mechanism defined in [RFC8843]. Regardless of policy, the application will always try to negotiate bundle onto a single transport and will offer a single bundle group across all "m=" sections; use of this single transport is contingent upon the answerer accepting bundle. However, by specifying a policy from the list below, the application can control exactly how aggressively it will try to bundle media streams together, which affects how it will interoperate with a non-bundle-aware endpoint. When negotiating with a non-bundle-aware endpoint, only the streams not marked as bundle-only streams will be established.¶

The set of available policies is as follows:¶

balanced:

The first "m=" section of each type (audio, video, or application) will contain transport parameters, which will allow an answerer to unbundle that section. The second and any subsequent "m=" sections of each type will be marked bundle-only. The result is that if there are N distinct media types, then candidates will be gathered for N media streams. This policy balances desire to multiplex with the need to ensure that basic audio and video can still be negotiated in legacy cases. When acting as answerer, if there is no bundle group in the offer, the implementation will reject all but the first "m=" section of each type.¶

max-compat:

All "m=" sections will contain transport parameters; none will be marked as bundle-only. This policy will allow all streams to be received by non-bundle-aware endpoints but will require separate candidates to be gathered for each media stream.¶

max-bundle:

Only the first "m=" section will contain transport parameters; all streams other than the first will be marked as bundle-only. This policy aims to minimize candidate gathering and maximize multiplexing, at the cost of less compatibility with legacy endpoints. When acting as answerer, the implementation will reject any "m=" sections other than the first "m=" section, unless they are in the same bundle group as that "m=" section.¶

As it provides the best trade-off between performance and compatibility with legacy endpoints, the default bundle policy MUST be set to "balanced".¶

The application can specify its preferred policy regarding use of RTP/RTCP multiplexing [RFC5761] using one of the following policies:¶

negotiate:

The JSEP implementation will gather both RTP and RTCP candidates but also will offer "a=rtcp-mux", thus allowing for compatibility with either multiplexing or non-multiplexing endpoints.¶

require:

The JSEP implementation will only gather RTP candidates and will insert an "a=rtcp-mux-only" indication into any new "m=" sections in offers it generates. This halves the number of candidates that the offerer needs to gather. Applying a description with an "m=" section that does not contain an "a=rtcp-mux" attribute will cause an error to be returned.¶

The default multiplexing policy MUST be set to "require". Implementations MAY choose to reject attempts by the application to set the multiplexing policy to "negotiate".¶

4.1.2. addTrack

The addTrack method adds a MediaStreamTrack to the PeerConnection, using the MediaStream argument to associate the track with other tracks in the same MediaStream, so that they can be added to the same "LS" (Lip Synchronization) group when creating an offer or answer. Adding tracks to the same "LS" group indicates that the playback of these tracks should be synchronized for proper lip sync, as described in [RFC5888], Section 7.  addTrack attempts to minimize the number of transceivers as follows: if the PeerConnection is in the "have‑remote-offer" state, the track will be attached to the first compatible transceiver that was created by the most recent call to setRemoteDescription and does not have a local track. Otherwise, a new transceiver will be created, as described in Section 4.1.4.¶

4.1.3. removeTrack

The removeTrack method removes a MediaStreamTrack from the PeerConnection, using the RtpSender argument to indicate which sender should have its track removed. The sender's track is cleared, and the sender stops sending. Future calls to createOffer will mark the "m=" section associated with the sender as recvonly (if transceiver.direction is sendrecv) or as inactive (if transceiver.direction is sendonly).¶

4.1.4. addTransceiver

The addTransceiver method adds a new RtpTransceiver to the PeerConnection. If a MediaStreamTrack argument is provided, then the transceiver will be configured with that media type and the track will be attached to the transceiver. Otherwise, the application MUST explicitly specify the type; this mode is useful for creating recvonly transceivers as well as for creating transceivers to which a track can be attached at some later point.¶

At the time of creation, the application can also specify a transceiver direction attribute, a set of MediaStreams that the transceiver is associated with (allowing "LS" group assignments), and a set of encodings for the media (used for simulcast as described in Section 3.7).¶

4.1.5. onaddtrack Event

The onaddtrack event is dispatched to the application when a new remote track has been signaled as a result of a setRemoteDescription call. The new track is supplied as a MediaStreamTrack object in the event, along with the MediaStream(s) the track is part of.¶

4.1.6. createDataChannel

The createDataChannel method creates a new data channel and attaches it to the PeerConnection. If no data channel currently exists for this PeerConnection, then a new offer/answer exchange is required. All data channels on a given PeerConnection share the same SCTP/DTLS association ("SCTP" stands for "Stream Control Transmission Protocol") and therefore the same "m=" section, so subsequent creation of data channels does not have any impact on the JSEP state.¶

The createDataChannel method also includes a number of arguments that are used by the PeerConnection (e.g., maxPacketLifetime) but are not reflected in the SDP and do not affect the JSEP state.¶

4.1.7. ondatachannel Event

The ondatachannel event is dispatched to the application when a new data channel has been negotiated by the remote side, which can occur at any time after the underlying SCTP/DTLS association has been established. The new data channel object is supplied in the event.¶

4.1.8. createOffer

The createOffer method generates a blob of SDP that contains an offer per [RFC3264] with the supported configurations for the session, including descriptions of the media added to this PeerConnection, the codec, RTP, and RTCP options supported by this implementation, and any candidates that have been gathered by the ICE agent. An options parameter may be supplied to provide additional control over the generated offer. This options parameter allows an application to trigger an ICE restart, for the purpose of reestablishing connectivity.¶

In the initial offer, the generated SDP will contain all desired functionality for the session (functionality that is supported but not desired by default may be omitted); for each SDP line, the generation of the SDP will follow the process defined for generating an initial offer from the specification that defines 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 stopping RtpTransceivers, 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 RFC 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. 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.¶

Calling this method may do things such as generating new ICE credentials, but it does not change the PeerConnection state, trigger candidate gathering, or cause media to start or stop flowing. Specifically, the offer is not applied, and does not become the pending local description, until setLocalDescription is called.¶

4.1.9. createAnswer

The createAnswer method generates a blob of SDP that contains an SDP answer per [RFC3264] with the supported configuration for the session that is compatible with the parameters supplied in the most recent call to setRemoteDescription; setRemoteDescription MUST have been called prior to calling createAnswer. Like createOffer, the returned blob contains descriptions of the media added to this PeerConnection, the codec/RTP/RTCP options negotiated for this session, and any candidates that have been gathered by the ICE agent. An options 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 specification that defines the given 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.¶

Calling this method may do things such as generating new ICE credentials, but it does not change the PeerConnection state, trigger candidate gathering, or cause a media state change. Specifically, the answer is not applied, and does not become the current local description, until setLocalDescription is called.¶

4.1.10. SessionDescriptionType

Session description objects (RTCSessionDescription) may be of type "offer", "pranswer", "answer", or "rollback". 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 MUST be parsed as an offer; said description may include many possible media configurations. A description used as an "offer" may be applied any time the PeerConnection is in a "stable" state or applied as an update to a previously supplied but unanswered "offer".¶

"pranswer" indicates that a description MUST be parsed as an answer, but not a final answer, and so MUST NOT result in the freeing of allocated resources. It may result in the start of media transmission, if the answer does not specify an inactive media direction. A description used as a "pranswer" may be applied as a response to an "offer" or as an update to a previously sent "pranswer".¶

"answer" indicates that a description MUST be parsed as an answer, the offer/answer exchange MUST be considered complete, and any resources (decoders, candidates) that are no longer needed SHOULD be released. A description used as an "answer" may be applied as a response to an "offer" or as an update to a previously sent "pranswer".¶

The only difference between a provisional and final answer is that the final answer results in the freeing of any unused resources that were allocated as a result of the offer. As such, the application can use some discretion on whether an answer should be applied as provisional or final and can change the type of the session 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).¶

"rollback" is a special session description type indicating that the state machine MUST be rolled back to the previous "stable" state, as described in Section 4.1.10.2. The contents MUST be empty.¶

4.1.11. setLocalDescription

The setLocalDescription method instructs the PeerConnection to apply the supplied session description as its local configuration. The type field indicates whether the description should be processed as an offer, provisional answer, final answer, or rollback; 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 current and pending local descriptions (e.g., support the codecs that exist in either description). This dual processing begins when the PeerConnection enters the "have-local-offer" state, and it continues until setRemoteDescription is called with either (1) a final answer, at which point the PeerConnection can fully adopt the pending local description or (2) a rollback, which results in a revert to the current local description.¶

This API indirectly controls the candidate gathering process. When a local description is supplied and the number of transports currently in use does not match the number of transports needed by the local description, the PeerConnection will create transports as needed and begin gathering candidates for each transport, using ones from the candidate pool if available.¶

If (1) setRemoteDescription was previously called with an offer, (2) setLocalDescription is called with an answer (provisional or final), (3) the media directions are compatible, and (4) media is available to send, this will result in the starting of media transmission.¶

4.1.12. setRemoteDescription

The setRemoteDescription method instructs the PeerConnection to apply the supplied session description as the desired remote configuration. As in setLocalDescription, the type field of the description indicates how it should be processed.¶

This API changes the local media state; among other things, it sets up local resources for sending and encoding media.¶

If (1) setLocalDescription was previously called with an offer, (2) setRemoteDescription is called with an answer (provisional or final), (3) the media directions are compatible, and (4) media is available to send, this will result in the starting of media transmission.¶

4.1.13. currentLocalDescription

The currentLocalDescription method returns the current negotiated local description -- i.e., the local description from the last successful offer/answer exchange -- in addition to any local candidates that have been generated by the ICE agent since the local description was set.¶

A null object will be returned if an offer/answer exchange has not yet been completed.¶

4.1.14. pendingLocalDescription

The pendingLocalDescription method returns a copy of the local description currently in negotiation -- i.e., a local offer set without any corresponding remote answer -- in addition to any local candidates that have been generated by the ICE agent since the local description was set.¶

A null object will be returned if the state of the PeerConnection is "stable" or "have-remote-offer".¶

4.1.15. currentRemoteDescription

The currentRemoteDescription method returns a copy of the current negotiated remote description -- i.e., the remote description from the last successful offer/answer exchange -- in addition to any remote candidates that have been supplied via processIceMessage since the remote description was set.¶

A null object will be returned if an offer/answer exchange has not yet been completed.¶

4.1.16. pendingRemoteDescription

The pendingRemoteDescription method returns a copy of the remote description currently in negotiation -- i.e., a remote offer set without any corresponding local answer -- in addition to any remote candidates that have been supplied via processIceMessage since the remote description was set.¶

A null object will be returned if the state of the PeerConnection is "stable" or "have-local-offer".¶

4.1.17. canTrickleIceCandidates

The canTrickleIceCandidates property indicates whether the remote side supports receiving trickled candidates. There are three potential values:¶

null:

No SDP has been received from the other side, so it is not known if it can handle trickle. This is the initial value before setRemoteDescription is called.¶

true:

SDP has been received from the other side indicating that it can support trickle.¶

false:

SDP has been received from the other side indicating that it cannot support trickle.¶

As described in Section 3.5.2, JSEP implementations always provide candidates to the application individually, consistent with what is needed for Trickle ICE. However, applications can use the canTrickleIceCandidates property to determine whether their peer can actually do Trickle ICE, i.e., whether it is safe to send an initial offer or answer followed later by candidates as they are gathered. As "true" is the only value that definitively indicates remote Trickle ICE support, an application that compares canTrickleIceCandidates against "true" will by default attempt Half Trickle on initial offers and Full Trickle on subsequent interactions with a Trickle ICE-compatible agent.¶

4.1.18. setConfiguration

The setConfiguration method allows the global configuration of the PeerConnection, which was initially set by constructor parameters, to be changed during the session. The effects of calling this method depend on when it is invoked, and they will differ, depending on which specific parameters are changed:¶

• Any changes to the STUN/TURN servers to use affect the next gathering phase. If an ICE gathering phase has already started or completed, the 'needs-ice-restart' bit mentioned in Section 3.5.1 will be set. This will cause the next call to createOffer to generate new ICE credentials, for the purpose of forcing an ICE restart and kicking off a new gathering phase, in which the new servers will be used. If the ICE candidate pool has a nonzero size and a local description has not yet been applied, any existing candidates will be discarded, and new candidates will be gathered from the new servers.¶
• Any change to the ICE candidate policy affects the next gathering phase. If an ICE gathering phase has already started or completed, the 'needs-ice-restart' bit will be set. Either way, changes to the policy have no effect on the candidate pool, because pooled candidates are not made available to the application until a gathering phase occurs, and so any necessary filtering can still be done on any pooled candidates.¶
• The ICE candidate pool size MUST NOT be changed after applying a local description. If a local description has not yet been applied, any changes to the ICE candidate pool size take effect immediately; if increased, additional candidates are pre-gathered; if decreased, the now-superfluous candidates are discarded.¶
• The bundle and RTCP-multiplexing policies MUST NOT be changed after the construction of the PeerConnection.¶

Calling this method may result in a change to the state of the ICE agent.¶

4.1.19. addIceCandidate

The addIceCandidate method provides an update to the ICE agent via an IceCandidate object (Section 3.5.2.1). If the IceCandidate's candidate field is non-null, the IceCandidate is treated as a new remote ICE candidate, which will be added to the current and/or pending remote description according to the rules defined for Trickle ICE. Otherwise, the IceCandidate is treated as an end-of-candidates indication, as defined in [RFC8838], Section 14.¶

In either case, the "m=" section index, MID, and ufrag fields from the supplied IceCandidate are used to determine which "m=" section and ICE candidate generation the IceCandidate belongs to, as described in Section 3.5.2.1 above. In the case of an end-of-candidates indication, null values for the "m=" section index and MID fields are interpreted to mean that the indication applies to all "m=" sections in the specified ICE candidate generation. However, if both fields are null for a new remote candidate, this MUST be treated as an invalid condition, as specified below.¶

If any IceCandidate fields contain invalid values or an error occurs during the processing of the IceCandidate object, the supplied IceCandidate MUST be ignored and an error MUST be returned.¶

Otherwise, the new remote candidate or end-of-candidates indication is supplied to the ICE agent. In the case of a new remote candidate, connectivity checks will be sent to the new candidate, assuming setLocalDescription has already been called to initialize the ICE gathering process.¶

4.1.20. onicecandidate Event

The onicecandidate event is dispatched to the application in two situations: (1) when the ICE agent has discovered a new allowed local ICE candidate during ICE gathering, as outlined in Section 3.5.1 and subject to the restrictions discussed in Section 3.5.3, or (2) when an ICE gathering phase completes. The event contains a single IceCandidate object, as defined in Section 3.5.2.1.¶

In the first case, the newly discovered candidate is reflected in the IceCandidate object, and all of its fields MUST be non-null. This candidate will also be added to the current and/or pending local description according to the rules defined for Trickle ICE.¶

In the second case, the event's IceCandidate object MUST have its candidate field set to null to indicate that the current gathering phase is complete, i.e., there will be no further onicecandidate events in this phase. However, the IceCandidate's ufrag field MUST be specified to indicate which ICE candidate generation is ending. The IceCandidate's "m=" section index and MID fields MAY be specified to indicate that the event applies to a specific "m=" section, or set to null to indicate it applies to all "m=" sections in the current ICE candidate generation. This event can be used by the application to generate an end-of-candidates indication, as defined in [RFC8838], Section 13.¶

4.2.1. stop

The stop method stops an RtpTransceiver. This will cause future calls to createOffer to generate a zero port for the associated "m=" section. See below for more details.¶

4.2.2. stopped

The stopped property indicates whether the transceiver has been stopped, either by a call to stop or by applying an answer that rejects the associated "m=" section. In either of these cases, it is set to "true" and otherwise will be set to "false".¶

A stopped RtpTransceiver does not send any outgoing RTP or RTCP or process any incoming RTP or RTCP. It cannot be restarted.¶

4.2.3. setDirection

The setDirection method sets the direction of a transceiver, which affects the direction property of the associated "m=" section on future calls to createOffer and createAnswer. The permitted values for direction are "recvonly", "sendrecv", "sendonly", and "inactive", mirroring the identically named direction attributes defined in [RFC4566], Section 6.¶

When creating offers, the transceiver direction is directly reflected in the output, even for re-offers. When creating answers, the transceiver direction is intersected with the offered direction, as explained in Section 5.3 below.¶

Note that while setDirection sets the direction property of the transceiver immediately (Section 4.2.4), this property does not immediately affect whether the transceiver's RtpSender will send or its RtpReceiver will receive. The direction in effect is represented by the currentDirection property, which is only updated when an answer is applied.¶

4.2.4. direction

The direction property indicates the last value passed into setDirection. If setDirection has never been called, it is set to the direction the transceiver was initialized with.¶

4.2.5. currentDirection

The currentDirection property indicates the last negotiated direction for the transceiver's associated "m=" section. More specifically, it indicates the direction attribute [RFC3264] of the associated "m=" section in the last applied answer (including provisional answers), with "send" and "recv" directions reversed if it was a remote answer. For example, if the direction attribute for the associated "m=" section in a remote answer is "recvonly", currentDirection is set to "sendonly".¶

If an answer that references this transceiver has not yet been applied or if the transceiver is stopped, currentDirection is set to "null".¶

4.2.6. setCodecPreferences

The setCodecPreferences method sets the codec preferences of a transceiver, which in turn affect the presence and order of codecs of the associated "m=" section on future calls to createOffer and createAnswer. Note that setCodecPreferences does not directly affect which codec the implementation decides to send. It only affects which codecs the implementation indicates that it prefers to receive, via the offer or answer. Even when a codec is excluded by setCodecPreferences, it still may be used to send until the next offer/answer exchange discards it.¶

The codec preferences of an RtpTransceiver can cause codecs to be excluded by subsequent calls to createOffer and createAnswer, in which case the corresponding media formats in the associated "m=" section will be excluded. The codec preferences cannot add media formats that would otherwise not be present.¶

The codec preferences of an RtpTransceiver can also determine the order of codecs in subsequent calls to createOffer and createAnswer, in which case the order of the media formats in the associated "m=" section will follow the specified preferences.¶

5.1.1. Usage Requirements

All session descriptions handled by JSEP implementations, both local and remote, MUST indicate support for the following specifications. If any of these are absent, this omission MUST be treated as an error.¶

• ICE, as specified in [RFC8445], MUST be used. Note that the remote endpoint may use a lite implementation; implementations MUST properly handle remote endpoints that use ICE-lite. The remote endpoint may also use an older version of ICE; implementations MUST properly handle remote endpoints that use ICE as specified in [RFC5245].¶
• DTLS [RFC6347] or DTLS-SRTP [RFC5763] MUST be used, as appropriate for the media type, as specified in [RFC8827].¶

The SDP security descriptions mechanism for SRTP keying [RFC4568] MUST NOT be used, as discussed in [RFC8827].¶

5.1.2. Profile Names and Interoperability

For media "m=" sections, JSEP implementations MUST support the "UDP/TLS/RTP/SAVPF" profile specified in [RFC5764] as well as the "TCP/DTLS/RTP/SAVPF" profile specified in [RFC7850] and MUST indicate one of these profiles for each media "m=" line they produce in an offer. For data "m=" sections, implementations MUST support the "UDP/DTLS/SCTP" profile as well as the "TCP/DTLS/SCTP" profile and MUST indicate one of these profiles for each data "m=" line they produce in an offer. The exact profile to use is determined by the protocol associated with the current default or selected ICE candidate, as described in [RFC8839], Section 4.2.1.2.¶

Unfortunately, in an attempt at compatibility, some endpoints generate other profile strings even when they mean to support one of these profiles. For instance, an endpoint might generate "RTP/AVP" but supply "a=fingerprint" and "a=rtcp-fb" attributes, indicating its willingness to support "UDP/TLS/RTP/SAVPF" or "TCP/DTLS/RTP/SAVPF". In order to simplify compatibility with such endpoints, JSEP implementations MUST follow the following rules when processing the media "m=" sections in a received offer:¶

• Any profile in the offer matching one of the following MUST be accepted:¶

  • "RTP/AVP" (defined in [RFC4566], Section 8.2.2)¶
  • "RTP/AVPF" (defined in [RFC4585], Section 9)¶
  • "RTP/SAVP" (defined in [RFC3711], Section 12)¶
  • "RTP/SAVPF" (defined in [RFC5124], Section 6)¶
  • "TCP/DTLS/RTP/SAVP" (defined in [RFC7850], Section 3.4)¶
  • "TCP/DTLS/RTP/SAVPF" (defined in [RFC7850], Section 3.5)¶
  • "UDP/TLS/RTP/SAVP" (defined in [RFC5764], Section 9)¶
  • "UDP/TLS/RTP/SAVPF" (defined in [RFC5764], Section 9)¶
• The profile in any "m=" line in any generated answer MUST exactly match the profile provided in the offer.¶
• Because DTLS-SRTP is REQUIRED, the choice of SAVP or AVP has no effect; support for DTLS-SRTP is determined by the presence of one or more "a=fingerprint" attributes. Note that lack of an "a=fingerprint" attribute will lead to negotiation failure.¶
• The use of AVPF or AVP simply controls the timing rules used for RTCP feedback. If AVPF is provided or an "a=rtcp-fb" attribute is present, assume AVPF timing, i.e., a default value of "trr-int=0". Otherwise, assume that AVPF is being used in an AVP-compatible mode and use a value of "trr-int=4000".¶
• For data "m=" sections, implementations MUST support receiving the "UDP/DTLS/SCTP", "TCP/DTLS/SCTP", or "DTLS/SCTP" (for backwards compatibility) profiles.¶

Note that re-offers by JSEP implementations MUST use the correct profile strings even if the initial offer/answer exchange used an (incorrect) older profile string. This simplifies JSEP behavior, with minimal downside, as any remote endpoint that fails to handle such a re-offer will also fail to handle a JSEP endpoint's initial offer.¶

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:¶

• The first SDP line MUST be "v=0" as defined in [RFC4566], Section 5.1.¶
• The second SDP line MUST be an "o=" line as defined in [RFC4566], Section 5.2. The value of the <username> field SHOULD be "-". The <sess-id> MUST be representable by a 64-bit signed integer, and the value MUST be less than 2⁶³-1. It is RECOMMENDED that the <sess-id> be constructed by generating a 64-bit quantity with the highest bit set to zero and the remaining 63 bits being cryptographically random. The value of the <nettype> <addrtype> <unicast-address> tuple SHOULD be set to a non-meaningful address, such as IN IP4 0.0.0.0, to prevent leaking a local IP address in this field; this problem is discussed in [RFC8828]. As mentioned in [RFC4566], the entire "o=" line needs to be unique, but selecting a random number for <sess-id> is sufficient to accomplish this.¶
• The third SDP line MUST be a "s=" line as defined in [RFC4566], Section 5.3; to match the "o=" line, a single dash SHOULD be used as the session name, e.g., "s=-". Note that this differs from the advice in [RFC4566], which proposes a single space, but as both "o=" and "s=" are meaningless in JSEP, having the same meaningless value seems clearer.¶
• Session Information ("i="), URI ("u="), Email Address ("e="), Phone Number ("p="), Repeat Times ("r="), and Time Zones ("z=") lines are not useful in this context and SHOULD NOT be included.¶
• Encryption Keys ("k=") lines do not provide sufficient security and MUST NOT be included.¶
• A "t=" line MUST be added, as specified in [RFC4566], Section 5.9; both <start-time> and <stop-time> SHOULD be set to zero, e.g., "t=0 0".¶
• An "a=ice-options" line with the "trickle" and "ice2" options MUST be added, as specified in [RFC8840], Section 4.1.1 and [RFC8445], Section 10.¶
• If WebRTC identity is being used, an "a=identity" line MUST be added, as described in [RFC8827], Section 5.¶

The next step is to generate "m=" sections, as specified in [RFC4566], Section 5.14. An "m=" section is generated for each RtpTransceiver that has been added to the PeerConnection, excluding any stopped RtpTransceivers; this is done in the order the RtpTransceivers were added to the PeerConnection. If there are no such RtpTransceivers, no "m=" sections are generated; more can be added later, as discussed in [RFC3264], Section 5.¶

For each "m=" section generated for an RtpTransceiver, establish a mapping between the transceiver and the index of the generated "m=" section.¶

Each "m=" section, provided it is not marked as bundle-only, MUST contain a unique set of ICE credentials and a unique set of ICE candidates. Bundle-only "m=" sections MUST NOT contain any ICE credentials and MUST NOT gather any candidates.¶

For DTLS, all "m=" sections MUST use any and all certificates that have been specified for the PeerConnection; as a result, they MUST all have the same fingerprint value or values [RFC8122], or these values MUST be session-level attributes.¶

Each "m=" section MUST be generated as specified in [RFC4566], Section 5.14. For the "m=" line itself, the following rules MUST be followed:¶

• If the "m=" section is marked as bundle-only, then the <port> value MUST be set to zero. Otherwise, the <port> value is set to the port of the default ICE candidate for this "m=" section, but given that no candidates are available yet, the default port value of 9 (Discard) MUST be used, as indicated in [RFC8840], Section 4.1.1.¶
• To properly indicate use of DTLS, the <proto> field MUST be set to "UDP/TLS/RTP/SAVPF", as specified in [RFC5764], Section 8.¶
• If codec preferences have been set for the associated transceiver, media formats MUST be generated in the corresponding order and MUST exclude any codecs not present in the codec preferences.¶
• Unless excluded by the above restrictions, the media formats MUST include the mandatory audio/video codecs as specified in [RFC7874], Section 3 and [RFC7742], Section 5.¶

The "m=" line MUST be followed immediately by a "c=" line, as specified in [RFC4566], Section 5.7. Again, as no candidates are available yet, the "c=" line MUST contain the default value "IN IP4 0.0.0.0", as defined in [RFC8840], Section 4.1.1.¶

[RFC8859] groups SDP attributes into different categories. To avoid unnecessary duplication when bundling, attributes of category IDENTICAL or TRANSPORT MUST NOT be repeated in bundled "m=" sections, repeating the guidance from [RFC8843], Section 7.1.3. This includes "m=" sections for which bundling has been negotiated and is still desired, as well as "m=" sections marked as bundle-only.¶

The following attributes, which are of a category other than IDENTICAL or TRANSPORT, MUST be included in each "m=" section:¶

• An "a=mid" line, as specified in [RFC5888], Section 4. All MID values MUST be generated in a fashion that does not leak user information, e.g., randomly or using a per-PeerConnection counter, and SHOULD be 3 bytes or less, to allow them to efficiently fit into the RTP header extension defined in [RFC8843], Section 15.2. Note that this does not set the RtpTransceiver mid property, as that only occurs when the description is applied. The generated MID value can be considered a "proposed" MID at this point.¶
• A direction attribute that is the same as that of the associated transceiver.¶
• For each media format on the "m=" line, "a=rtpmap" and "a=fmtp" lines, as specified in [RFC4566], Section 6 and [RFC3264], Section 5.1.¶
• 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 of the primary codec, as specified in [RFC4588], Section 8.1.¶
• For each supported Forward Error Correction (FEC) mechanism, "a=rtpmap" and "a=fmtp" lines, as specified in [RFC4566], Section 6. The FEC mechanisms that MUST be supported are specified in [RFC8854], Section 7, and specific usage for each media type is outlined in Sections 4 and 5.¶
• If this "m=" section is for media with configurable durations of media per packet, e.g., audio, an "a=maxptime" line, indicating the maximum amount of media, specified in milliseconds, that can be encapsulated in each packet, as specified in [RFC4566], Section 6. This value is set to the smallest of the maximum duration values across all the codecs included in the "m=" section.¶
• If this "m=" section is for video media and there are known limitations on the size of images that can be decoded, an "a=imageattr" line, as specified in Section 3.6.¶
• 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 [RFC8834], Section 5.2. Any header extensions that require encryption MUST be specified as indicated in [RFC6904], Section 4.¶
• For each supported RTCP feedback mechanism, an "a=rtcp-fb" line, as specified in [RFC4585], Section 4.2. The list of RTCP feedback mechanisms that SHOULD/MUST be supported is specified in [RFC8834], Section 5.1.¶

• If the RtpTransceiver has a sendrecv or sendonly direction:¶

  • For each MediaStream that was associated with the transceiver when it was created via addTrack or addTransceiver, an "a=msid" line, as specified in [RFC8830], Section 2, but omitting the "appdata" field.¶
• If the RtpTransceiver has a sendrecv or sendonly direction, and the application has specified a rid-id for an encoding, or has specified more than one encoding in the RtpSenders's parameters, an "a=rid" line for each encoding specified. The "a=rid" line is specified in [RFC8851], and its direction MUST be "send". If the application has chosen a rid-id, it MUST be used; otherwise, a rid-id MUST be generated by the implementation. rid-ids MUST be generated in a fashion that does not leak user information, e.g., randomly or using a per-PeerConnection counter (see guidance at the end of [RFC8852], Section 3.3), and SHOULD be 3 bytes or less, to allow them to efficiently fit into the RTP header extensions defined in [RFC8852], Section 3.3. If no encodings have been specified, or only one encoding is specified but without a rid-id, then no "a=rid" lines are generated.¶
• If the RtpTransceiver has a sendrecv or sendonly direction and more than one "a=rid" line has been generated, an "a=simulcast" line, with direction "send", as defined in [RFC8853], Section 5.1. The associated set of rid-ids MUST include all of the rid-ids used in the "a=rid" lines for this "m=" section.¶
• If (1) the bundle policy for this PeerConnection is set to "max-bundle" and this is not the first "m=" section or (2) the bundle policy is set to "balanced" and this is not the first "m=" section for this media type, an "a=bundle-only" line.¶

The following attributes, which are of category IDENTICAL or TRANSPORT, MUST appear only in "m=" sections that either have a unique address or are associated with the BUNDLE-tag. (In initial offers, this means those "m=" sections that do not contain an "a=bundle-only" attribute.)¶

• "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC8839], Section 5.4.¶
• For each desired digest algorithm, one or more "a=fingerprint" lines for each of the endpoint's certificates, as specified in [RFC8122], Section 5.¶
• 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".¶
• An "a=tls-id" line, as specified in [RFC8842], Section 5.2.¶
• An "a=rtcp" line, as specified in [RFC3605], Section 2.1, containing the default value "9 IN IP4 0.0.0.0", because no candidates have yet been gathered.¶
• An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.3.¶
• If the RTP/RTCP multiplexing policy is "require", an "a=rtcp-mux-only" line, as specified in [RFC8858], Section 4.¶
• An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5.¶

Lastly, if a data channel has been created, an "m=" section MUST be generated for data. The <media> field MUST be set to "application", and the <proto> field MUST be set to "UDP/DTLS/SCTP" [RFC8841]. The <fmt> value MUST be set to "webrtc-datachannel" as specified in [RFC8841], Section 4.2.2.¶

Within the data "m=" section, an "a=mid" line MUST be generated and included as described above, along with an "a=sctp-port" line referencing the SCTP port number, as defined in [RFC8841], Section 5.1; and, if appropriate, an "a=max-message-size" line, as defined in [RFC8841], Section 6.1.¶

As discussed above, the following attributes of category IDENTICAL or TRANSPORT are included only if the data "m=" section either has a unique address or is associated with the BUNDLE-tag (e.g., if it is the only "m=" section):¶

• "a=ice-ufrag"¶
• "a=ice-pwd"¶
• "a=fingerprint"¶
• "a=setup"¶
• "a=tls-id"¶

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 MUST include the mid identifiers of each "m=" section. The effect of this is that the JSEP implementation offers all "m=" sections as one bundle group. However, whether the "m=" sections are bundle-only or not depends on the bundle policy.¶

The next step is to generate session-level lip sync groups as defined in [RFC5888], Section 7. For each MediaStream referenced by more than one RtpTransceiver (by passing those MediaStreams as arguments to the addTrack and addTransceiver methods), a group of type "LS" MUST be added that contains the MID values for each RtpTransceiver.¶

Attributes that SDP permits to be at either the session level or the media level SHOULD generally be at the media level even if they are identical. This assists development and debugging by making it easier to understand individual media sections, especially if one of a set of initially identical attributes is subsequently changed. However, implementations MAY choose to aggregate attributes at the session level, and JSEP implementations MUST be prepared to receive attributes in either location.¶

Attributes other than the ones specified above MAY be included, except for the following attributes, which are specifically incompatible with the requirements of [RFC8834] and MUST NOT be included:¶

• "a=crypto"¶
• "a=key-mgmt"¶
• "a=ice-lite"¶

Note that when bundle is used, any additional attributes that are added MUST follow the advice in [RFC8859] on how those attributes interact with bundle.¶

Note that these requirements are in some cases stricter than those of SDP. Implementations MUST be prepared to accept compliant SDP even if it would not conform to the requirements for generating SDP in this specification.¶

5.2.2. Subsequent Offers

When createOffer is called a second (or later) time or is called after a local description has already been installed, the processing is somewhat different than for an initial offer.¶

If the previous offer was not applied using setLocalDescription, meaning the PeerConnection is still in the "stable" state, the steps for generating an initial offer MUST be followed, subject to the following restriction:¶

• The fields of the "o=" line MUST stay the same except for the <session-version> field, which MUST increment by one on each call to createOffer if the offer might differ from the output of the previous call to createOffer; implementations MAY opt to increment <session-version> on every call. The value of the generated <session-version> is independent of the <session-version> of the current local description; in particular, in the case where the current version is N, an offer is created and applied with version N+1, and then that offer is rolled back so that the current version is again N, the next generated offer will still have version N+2.¶

Note that if the application creates an offer by reading currentLocalDescription instead of calling createOffer, the returned SDP may be different than when setLocalDescription was originally called, due to the addition of gathered ICE candidates, but the <session-version> will not have changed. There are no known scenarios in which this causes problems, but if this is a concern, the solution is simply to use createOffer to ensure a unique <session-version>.¶

If the previous offer was applied using setLocalDescription, but a corresponding answer from the remote side has not yet been applied, meaning the PeerConnection is still in the "have-local-offer" state, an offer is generated by following the steps in the "stable" state above, along with these exceptions:¶

• The "s=" and "t=" lines MUST stay the same.¶
• If any RtpTransceiver has been added and there exists an "m=" section with a zero port in the current local description or the current remote description, that "m=" section MUST be recycled by generating an "m=" section for the added RtpTransceiver as if the "m=" section were being added to the session description (including a new MID value) and placing it at the same index as the "m=" section with a zero port.¶
• If an RtpTransceiver is stopped and is not associated with an "m=" section, an "m=" section MUST NOT be generated for it. This prevents adding back RtpTransceivers whose "m=" sections were recycled and used for a new RtpTransceiver in a previous offer/ answer exchange, as described above.¶
• If an RtpTransceiver has been stopped and is associated with an "m=" section, and the "m=" section is not being recycled as described above, an "m=" section MUST be generated for it with the port set to zero and all "a=msid" lines removed.¶
• For RtpTransceivers that are not stopped, the "a=msid" line or lines MUST stay the same if they are present in the current description, regardless of changes to the transceiver's direction or track. If no "a=msid" line is present in the current description, "a=msid" line(s) MUST be generated according to the same rules as for an initial offer.¶
• Each "m=" and "c=" line MUST be filled in with the port, relevant RTP profile, and address of the default candidate for the "m=" section, as described in [RFC8839], Section 4.2.1.2 and clarified in Section 5.1.2. If no RTP candidates have yet been gathered, default values MUST still be used, as described above.¶
• Each "a=mid" line MUST stay the same.¶
• Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless the ICE configuration has changed (e.g., changes to either the supported STUN/TURN servers or the ICE candidate policy) or the IceRestart option (Section 5.2.3.1) was specified. If the "m=" section is bundled into another "m=" section, it still MUST NOT contain any ICE credentials.¶
• If the "m=" section is not bundled into another "m=" section, its "a=rtcp" attribute line MUST be filled in with the port and address of the default RTCP candidate, as indicated in [RFC5761], Section 5.1.3. If no RTCP candidates have yet been gathered, default values MUST be used, as described in Section 5.2.1 above.¶
• If the "m=" section is not bundled into another "m=" section, for each candidate that has been gathered during the most recent gathering phase (see Section 3.5.1), an "a=candidate" line MUST be added, as defined in [RFC8839], Section 5.1. If candidate gathering for the section has completed, an "a=end-of-candidates" attribute MUST be added, as described in [RFC8840], Section 8.2. If the "m=" section is bundled into another "m=" section, both "a=candidate" and "a=end-of-candidates" MUST be omitted.¶
• For RtpTransceivers that are still present, the "a=rid" lines MUST stay the same.¶
• For RtpTransceivers that are still present, any "a=simulcast" line MUST stay the same.¶

If the previous offer was applied using setLocalDescription, and a corresponding answer from the remote side has been applied using setRemoteDescription, meaning the PeerConnection is in the "have-remote-pranswer" state or the "stable" state, an offer is generated based on the negotiated session descriptions by following the steps mentioned for the "have-local-offer" state above.¶

In addition, for each existing, non-recycled, 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 local or remote description, as appropriate:¶

• The "m=" line and corresponding "a=rtpmap" and "a=fmtp" lines MUST only include media formats that have not been excluded by the codec preferences of the associated transceiver and also MUST include all currently available formats. Media formats that were previously offered but are no longer available (e.g., a shared hardware codec) MAY be excluded.¶
• Unless codec preferences have been set for the associated transceiver, the media formats on the "m=" line MUST be generated in the same order as in the most recent answer. Any media formats that were not present in the most recent answer MUST be added after all existing formats.¶
• The RTP header extensions MUST only include those that are present in the most recent answer.¶
• The RTCP feedback mechanisms MUST only include those that are present in the most recent answer, except for the case of format-specific mechanisms that are referencing a newly added media format.¶
• The "a=rtcp" line MUST NOT be added if the most recent answer included an "a=rtcp-mux" line.¶
• The "a=rtcp-mux" line MUST be the same as that in the most recent answer.¶
• The "a=rtcp-mux-only" line MUST NOT be added.¶
• The "a=rtcp-rsize" line MUST NOT be added unless present in the most recent answer.¶
• An "a=bundle-only" line, as defined in [RFC8843], Section 6, MUST NOT be added. Instead, JSEP implementations MUST simply omit parameters in the IDENTICAL and TRANSPORT categories for bundled "m=" sections, as described in [RFC8843], Section 7.1.3.¶
• Note that if media "m=" sections are bundled into a data "m=" section, then certain TRANSPORT and IDENTICAL attributes may appear in the data "m=" section even if they would otherwise only be appropriate for a media "m=" section (e.g., "a=rtcp-mux"). This cannot happen in initial offers because in the initial offer JSEP implementations always list media "m=" sections (if any) before the data "m=" section (if any), and at least one of those media "m=" sections will not have the "a=bundle-only" attribute. Therefore, in initial offers, any "a=bundle-only" "m=" sections will be bundled into a preceding non-bundle-only media "m=" section.¶

The "a=group:BUNDLE" attribute MUST include the MID identifiers specified in the bundle group in the most recent answer, minus any "m=" sections that have been marked as rejected, plus any newly added or re-enabled "m=" sections. In other words, the bundle attribute MUST contain all "m=" sections that were previously bundled, as long as they are still alive, as well as any new "m=" sections.¶

"a=group:LS" attributes are generated in the same way as for initial offers, with the additional stipulation that any lip sync groups that were present in the most recent answer MUST continue to exist and MUST contain any previously existing MID identifiers, as long as the identified "m=" sections still exist and are not rejected, and the group still contains at least two MID identifiers. This ensures that any synchronized "recvonly" "m=" sections continue to be synchronized in the new offer.¶

5.2.3. Options Handling

The createOffer method takes as a parameter an RTCOfferOptions object. Special processing is performed when generating an SDP description if the following options are present.¶

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 JSEP 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 above is not present, this MUST be treated as an error and MUST cause the affected "m=" sections to be marked as rejected.¶

The first step in generating an initial answer is to generate session-level attributes. The process here is identical to that indicated in Section 5.2.1 above, except that the "a=ice-options" line, with the "trickle" option as specified in [RFC8840], Section 4.1.3 and the "ice2" option as specified in [RFC8445], Section 10, is only included if such an option was present in the offer.¶

The next step is to generate session-level lip sync groups, as defined in [RFC5888], Section 7. For each group of type "LS" present in the offer, select the local RtpTransceivers that are referenced by the MID values in the specified group, and determine which of them either reference a common local MediaStream (specified in the calls to addTrack/addTransceiver used to create them) or have no MediaStream to reference because they were not created by addTrack/addTransceiver. If at least two such RtpTransceivers exist, a group of type "LS" with the MID values of these RtpTransceivers MUST be added. Otherwise, the offered "LS" group MUST be ignored and no corresponding group generated in the answer.¶

As a simple example, consider the following offer of a single audio and single video track contained in the same MediaStream. SDP lines not relevant to this example have been removed for clarity. As explained in Section 5.2, a group of type "LS" has been added that references each track's RtpTransceiver.¶

          a=group:LS a1 v1
          m=audio 10000 UDP/TLS/RTP/SAVPF 0
          a=mid:a1
          a=msid:ms1
          m=video 10001 UDP/TLS/RTP/SAVPF 96
          a=mid:v1
          a=msid:ms1

If the answerer uses a single MediaStream when it adds its tracks, both of its transceivers will reference this stream, and so the subsequent answer will contain a "LS" group identical to that in the offer, as shown below:¶

          a=group:LS a1 v1
          m=audio 20000 UDP/TLS/RTP/SAVPF 0
          a=mid:a1
          a=msid:ms2
          m=video 20001 UDP/TLS/RTP/SAVPF 96
          a=mid:v1
          a=msid:ms2

However, if the answerer groups its tracks into separate MediaStreams, its transceivers will reference different streams, and so the subsequent answer will not contain a "LS" group.¶

          m=audio 20000 UDP/TLS/RTP/SAVPF 0
          a=mid:a1
          a=msid:ms2a
          m=video 20001 UDP/TLS/RTP/SAVPF 96
          a=mid:v1
          a=msid:ms2b

Finally, if the answerer does not add any tracks, its transceivers will not reference any MediaStreams, causing the preferences of the offerer to be maintained, and so the subsequent answer will contain an identical "LS" group.¶

          a=group:LS a1 v1
          m=audio 20000 UDP/TLS/RTP/SAVPF 0
          a=mid:a1
          a=recvonly
          m=video 20001 UDP/TLS/RTP/SAVPF 96
          a=mid:v1
          a=recvonly

The example in Section 7.2 shows a more involved case of "LS" group generation.¶

The next step is to generate an "m=" section 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 are considered to be present in each "m=" section. Each offered "m=" section will have an associated RtpTransceiver, as described in Section 5.10. If there are more RtpTransceivers than there are "m=" sections, the unmatched RtpTransceivers will need to be associated in a subsequent offer.¶

For each offered "m=" section, if any of the following conditions are true, 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 further processing for this "m=" section can be skipped:¶

• The associated RtpTransceiver has been stopped.¶
• There is no offered media format that is both supported and, if applicable, allowed by codec preferences.¶
• The bundle policy is "max-bundle", and this is not the first "m=" section or in the same bundle group as the first "m=" section.¶
• The bundle policy is "balanced", and this is not the first "m=" section for this media type or in the same bundle group as the first "m=" section for this media type.¶
• This "m=" section is in a bundle group, and the group's offerer tagged "m=" section is being rejected due to one of the above reasons. This requires all "m=" sections in the bundle group to be rejected, as specified in [RFC8843], Section 7.3.3.¶

Otherwise, each "m=" section in the answer MUST then be generated as specified in [RFC3264], Section 6.1. For the "m=" line itself, the following rules MUST be followed:¶

• The <port> value would normally be set to the port of the default ICE candidate for this "m=" section, but given that no candidates are available yet, the default <port> value of 9 (Discard) MUST be used, as indicated in [RFC8840], Section 4.1.1.¶
• The <proto> field MUST be set to exactly match the <proto> field for the corresponding "m=" line in the offer.¶
• If codec preferences have been set for the associated transceiver, media formats MUST be generated in the corresponding order, regardless of what was offered, and MUST exclude any codecs not present in the codec preferences.¶
• Otherwise, the media formats on the "m=" line MUST be generated in the same order as those offered in the current remote description, excluding any currently unsupported formats. Any currently available media formats that are not present in the current remote description MUST be added after all existing formats.¶
• In either case, the media formats in the answer MUST include at least one format that is present in the offer but MAY include formats that are locally supported but not present in the offer, as mentioned in [RFC3264], Section 6.1. If no common format exists, the "m=" section is rejected as described above.¶

The "m=" line MUST be followed immediately by a "c=" line, as specified in [RFC4566], Section 5.7. Again, as no candidates are available yet, the "c=" line MUST contain the default value "IN IP4 0.0.0.0", as defined in [RFC8840], Section 4.1.3.¶

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 contain the following attributes (which are of attribute types other than IDENTICAL or TRANSPORT):¶

• If and only 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.¶
• A direction attribute, determined by applying the rules regarding the offered direction specified in [RFC3264], Section 6.1, and then intersecting with the direction of the associated RtpTransceiver. For example, in the case where an "m=" section is offered as "sendonly" and the local transceiver is set to "sendrecv", the result in the answer is a "recvonly" direction.¶
• For each media format on the "m=" line, "a=rtpmap" and "a=fmtp" lines, as specified in [RFC4566], Section 6 and [RFC3264], Section 6.1.¶
• 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 of the primary codec, as specified in [RFC4588], Section 8.1.¶
• For each supported FEC mechanism, "a=rtpmap" and "a=fmtp" lines, as specified in [RFC4566], Section 6. The FEC mechanisms that MUST be supported are specified in [RFC8854], Section 7, and specific usage for each media type is outlined in Sections 4 and 5.¶
• If this "m=" section is for media with configurable durations of media per packet, e.g., audio, an "a=maxptime" line, as described in Section 5.2.¶
• If this "m=" section is for video media and there are known limitations on the size of images that can be decoded, an "a=imageattr" line, as specified in Section 3.6.¶
• 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 [RFC8834], Section 5.2. Any header extensions that require encryption MUST be specified as indicated in [RFC6904], Section 4.¶
• For each supported RTCP feedback mechanism that is present in the offer, an "a=rtcp-fb" line, as specified in [RFC4585], Section 4.2. The list of RTCP feedback mechanisms that SHOULD/MUST be supported is specified in [RFC8834], Section 5.1.¶

• If the RtpTransceiver has a sendrecv or sendonly direction:¶

  • For each MediaStream that was associated with the transceiver when it was created via addTrack or addTransceiver, an "a=msid" line, as specified in [RFC8830], Section 2, but omitting the "appdata" field.¶

Each "m=" section that is not bundled into another "m=" section MUST contain the following attributes (which are of category IDENTICAL or TRANSPORT):¶

• "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC8839], Section 5.4.¶
• For each desired digest algorithm, one or more "a=fingerprint" lines for each of the endpoint's certificates, as specified in [RFC8122], Section 5.¶
• 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". When the offer contains the "actpass" value, as will always be the case with JSEP endpoints, the answerer SHOULD use the "active" role. Offers from non-JSEP endpoints MAY send other values for "a=setup", in which case the answer MUST use a value consistent with the value in the offer.¶
• An "a=tls-id" line, as specified in [RFC8842], Section 5.3.¶
• If present in the offer, an "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.3. Otherwise, an "a=rtcp" line, as specified in [RFC3605], Section 2.1, containing the default value "9 IN IP4 0.0.0.0" (because no candidates have yet been gathered).¶
• If present in the offer, an "a=rtcp-rsize" line, as specified in [RFC5506], Section 5.¶

If a data channel "m=" section has been offered, an "m=" section MUST also be generated for data. The <media> field MUST be set to "application", and the <proto> and <fmt> fields MUST be set to exactly match the fields in the offer.¶

Within the data "m=" section, an "a=mid" line MUST be generated and included as described above, along with an "a=sctp-port" line referencing the SCTP port number, as defined in [RFC8841], Section 5.1; and, if appropriate, an "a=max-message-size" line, as defined in [RFC8841], Section 6.1.¶

As discussed above, the following attributes of category IDENTICAL or TRANSPORT are included only if the data "m=" section is not bundled into another "m=" section:¶

• "a=ice-ufrag"¶
• "a=ice-pwd"¶
• "a=fingerprint"¶
• "a=setup"¶
• "a=tls-id"¶

Note that if media "m=" sections are bundled into a data "m=" section, then certain TRANSPORT and IDENTICAL attributes may also appear in the data "m=" section even if they would otherwise only be appropriate for a media "m=" section (e.g., "a=rtcp-mux").¶

If "a=group" attributes with semantics of "BUNDLE" are offered, corresponding session-level "a=group" attributes MUST be added as specified in [RFC5888]. These attributes MUST have semantics "BUNDLE" and MUST include all mid identifiers from the offered bundle groups that have not been rejected. Note that regardless of the presence of "a=bundle-only" in the offer, all "m=" sections in the answer MUST NOT have an "a=bundle-only" line.¶

Attributes that are common between all "m=" sections MAY be moved to the session level if explicitly defined to be valid at the session level.¶

The attributes prohibited in the creation of offers are also prohibited in the creation of answers.¶

5.3.2. Subsequent Answers

When createAnswer is called a second (or later) time or is called after a local description has already been installed, the processing is somewhat different than for an initial answer.¶

If the previous answer was not applied using setLocalDescription, meaning the PeerConnection is still in the "have-remote-offer" state, the steps for generating an initial answer MUST be followed, subject to the following restriction:¶

• The fields of the "o=" line MUST stay the same except for the <session-version> field, which MUST increment if the session description changes in any way from the previously generated answer.¶

If any session description was previously supplied to setLocalDescription, an answer is generated by following the steps in the "have-remote-offer" state above, along with these exceptions:¶

• The "s=" and "t=" lines MUST stay the same.¶
• Each "m=" and "c=" line MUST be filled in with the port and address of the default candidate for the "m=" section, as described in [RFC8839], Section 4.2.1.2. Note that in certain cases, the "m=" line protocol may not match that of the default candidate, because the "m=" line protocol value MUST match what was supplied in the offer, as described above.¶
• Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless the "m=" section is restarting, in which case new ICE credentials MUST be created as specified in [RFC8839], Section 4.4.1.1.1. If the "m=" section is bundled into another "m=" section, it still MUST NOT contain any ICE credentials.¶
• Each "a=tls-id" line MUST stay the same, unless the offerer's "a=tls-id" line changed, in which case a new tls-id value MUST be created, as described in [RFC8842], Section 5.2.¶
• Each "a=setup" line MUST use an "active" or "passive" role value consistent with the existing DTLS association, if the association is being continued by the offerer.¶
• RTCP multiplexing MUST be used, and an "a=rtcp-mux" line inserted if and only if the "m=" section previously used RTCP multiplexing.¶
• If the "m=" section is not bundled into another "m=" section and RTCP multiplexing is not active, an "a=rtcp" attribute line MUST be filled in with the port and address of the default RTCP candidate. If no RTCP candidates have yet been gathered, default values MUST be used, as described in Section 5.3.1 above.¶
• If the "m=" section is not bundled into another "m=" section, for each candidate that has been gathered during the most recent gathering phase (see Section 3.5.1), an "a=candidate" line MUST be added, as defined in [RFC8839], Section 5.1. If candidate gathering for the section has completed, an "a=end-of-candidates" attribute MUST be added, as described in [RFC8840], Section 8.2. If the "m=" section is bundled into another "m=" section, both "a=candidate" and "a=end-of-candidates" MUST be omitted.¶
• For RtpTransceivers that are not stopped, the "a=msid" line(s) MUST stay the same, regardless of changes to the transceiver's direction or track. If no "a=msid" line is present in the current description, "a=msid" line(s) MUST be generated according to the same rules as for an initial answer.¶

5.3.3. Options Handling

The createAnswer method takes as a parameter an RTCAnswerOptions object. The set of parameters for RTCAnswerOptions is different than those supported in RTCOfferOptions; the IceRestart option is unnecessary, as ICE credentials will automatically be changed for all "m=" sections where the offerer chose to perform ICE restart.¶

The following options are supported in RTCAnswerOptions.¶

5.8.1. Session-Level Parsing

First, the session-level lines are checked and parsed. These lines MUST occur in a specific order, and with a specific syntax, as defined in [RFC4566], Section 5. Note that while the specific line types (e.g., "v=", "c=") MUST occur in the defined order, lines of the same type (typically "a=") can occur in any order.¶

The following non-attribute lines are not meaningful in the JSEP context and MAY be discarded once they have been checked.¶

• The "c=" line MUST be checked for syntax, but its value is only used for ICE mismatch detection, as defined in [RFC8445], Section 5.4. Note that JSEP implementations should never encounter this condition because ICE is required for WebRTC.¶
• The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k=" lines MUST be checked for syntax, but their values are not otherwise used.¶

The remaining non-attribute lines are processed as follows:¶

• The "v=" line MUST have a version of 0, as specified in [RFC4566], Section 5.1.¶
• The "o=" line MUST be parsed as specified in [RFC4566], Section 5.2.¶
• The "b=" line, if present, MUST be parsed as specified in [RFC4566], Section 5.8, and the bwtype and bandwidth values stored.¶

Finally, the attribute lines are processed. Specific processing MUST be applied for the following session-level attribute ("a=") lines:¶

• Any "a=group" lines are parsed as specified in [RFC5888], Section 5, and the group's semantics and mids are stored.¶
• If present, a single "a=ice-lite" line is parsed as specified in [RFC8839], Section 5.3, and a value indicating the presence of ice-lite is stored.¶
• If present, a single "a=ice-ufrag" line is parsed as specified in [RFC8839], Section 5.4, and the ufrag value is stored.¶
• If present, a single "a=ice-pwd" line is parsed as specified in [RFC8839], Section 5.4, and the password value is stored.¶
• If present, a single "a=ice-options" line is parsed as specified in [RFC8839], Section 5.6, and the set of specified options is stored.¶
• Any "a=fingerprint" lines are parsed as specified in [RFC8122], Section 5, and the set of fingerprint and algorithm values is stored.¶
• If present, a single "a=setup" line is parsed as specified in [RFC4145], Section 4, and the setup value is stored.¶
• If present, a single "a=tls-id" line is parsed as specified in [RFC8842], Section 5, and the attribute value is stored.¶
• Any "a=identity" lines are parsed and the identity values stored for subsequent verification, as specified in [RFC8827], Section 5.¶
• Any "a=extmap" lines are parsed as specified in [RFC5285], Section 5, and their values are stored.¶

Other attributes that are not relevant to JSEP may also be present, and implementations SHOULD process any that they recognize. As required by [RFC4566], Section 5.13, unknown attribute lines MUST be ignored.¶

Once all the session-level lines have been parsed, processing continues with the lines in "m=" sections.¶

5.8.2. Media Section Parsing

Like the session-level lines, the media section lines MUST occur in the specific order and with the specific syntax defined in [RFC4566], Section 5.¶

The "m=" line itself MUST be parsed as described in [RFC4566], Section 5.14, and the <media>, <port>, <proto>, and <fmt> values stored.¶

Following the "m=" line, specific processing MUST be applied for the following non-attribute lines:¶

• As with the "c=" line at the session level, the "c=" line MUST be parsed according to [RFC4566], Section 5.7, but its value is not used.¶
• The "b=" line, if present, MUST be parsed as specified in [RFC4566], Section 5.8, and the bwtype and bandwidth values stored.¶

Specific processing MUST also be applied for the following attribute lines:¶

• If present, a single "a=ice-ufrag" line is parsed as specified in [RFC8839], Section 5.4, and the ufrag value is stored.¶
• If present, a single "a=ice-pwd" line is parsed as specified in [RFC8839], Section 5.4, and the password value is stored.¶
• If present, a single "a=ice-options" line is parsed as specified in [RFC8839], Section 5.6, and the set of specified options is stored.¶
• Any "a=candidate" attributes MUST be parsed as specified in [RFC8839], Section 5.1, and their values stored.¶
• Any "a=remote-candidates" attributes MUST be parsed as specified in [RFC8839], Section 5.2, but their values are ignored.¶
• If present, a single "a=end-of-candidates" attribute MUST be parsed as specified in [RFC8840], Section 8.1, and its presence or absence flagged and stored.¶
• Any "a=fingerprint" lines are parsed as specified in [RFC8122], Section 5, and the set of fingerprint and algorithm values is stored.¶

If the "m=" <proto> value indicates use of RTP, as described in Section 5.1.2 above, the following attribute lines MUST be processed:¶

• The "m=" <fmt> value MUST be parsed as specified in [RFC4566], Section 5.14, and the individual values stored.¶
• Any "a=rtpmap" or "a=fmtp" lines MUST be parsed as specified in [RFC4566], Section 6, and their values stored.¶
• If present, a single "a=ptime" line MUST be parsed as described in [RFC4566], Section 6, and its value stored.¶
• If present, a single "a=maxptime" line MUST be parsed as described in [RFC4566], Section 6, and its value stored.¶
• If present, a single direction attribute line (e.g., "a=sendrecv") MUST be parsed as described in [RFC4566], Section 6, and its value stored.¶
• Any "a=ssrc" attributes MUST be parsed as specified in [RFC5576], Section 4.1, and their values stored.¶
• Any "a=extmap" attributes MUST be parsed as specified in [RFC5285], Section 5, and their values stored.¶
• Any "a=rtcp-fb" attributes MUST be parsed as specified in [RFC4585], Section 4.2, and their values stored.¶
• If present, a single "a=rtcp-mux" attribute MUST be parsed as specified in [RFC5761], Section 5.1.3, and its presence or absence flagged and stored.¶
• If present, a single "a=rtcp-mux-only" attribute MUST be parsed as specified in [RFC8858], Section 3, and its presence or absence flagged and stored.¶
• If present, a single "a=rtcp-rsize" attribute MUST be parsed as specified in [RFC5506], Section 5, and its presence or absence flagged and stored.¶
• If present, a single "a=rtcp" attribute MUST be parsed as specified in [RFC3605], Section 2.1, but its value is ignored, as this information is superfluous when using ICE.¶
• If present, "a=msid" attributes MUST be parsed as specified in [RFC8830], Section 3.2, and their values stored, ignoring any "appdata" field. If no "a=msid" attributes are present, a random msid-id value is generated for a "default" MediaStream for the session, if not already present, and this value is stored.¶
• Any "a=imageattr" attributes MUST be parsed as specified in [RFC6236], Section 3, and their values stored.¶
• Any "a=rid" lines MUST be parsed as specified in [RFC8851], Section 10, and their values stored.¶
• If present, a single "a=simulcast" line MUST be parsed as specified in [RFC8853], and its values stored.¶

Otherwise, if the "m=" <proto> value indicates use of SCTP, the following attribute lines MUST be processed:¶

• The "m=" <fmt> value MUST be parsed as specified in [RFC8841], Section 4.3, and the application protocol value stored.¶
• An "a=sctp-port" attribute MUST be present, and it MUST be parsed as specified in [RFC8841], Section 5.2, and the value stored.¶
• If present, a single "a=max-message-size" attribute MUST be parsed as specified in [RFC8841], Section 6, and the value stored. Otherwise, use the specified default.¶

Other attributes that are not relevant to JSEP may also be present, and implementations SHOULD process any that they recognize. As required by [RFC4566], Section 5.13, unknown attribute lines MUST be ignored.¶