JavaScript Session Establishment Protocol (JSEP)Google747 6th Street SouthKirklandWA98033United States of Americajustin@uberti.nameCisco400 3rd Avenue SWCalgaryABT2P 4H2Canadafluffy@iii.caMozillaekr@rtfm.comwebrtcsdpnegotiationsignalingpeerconnectionapiicertpofferanswerThis document describes the mechanisms for allowing a
JavaScript application to control the signaling plane of a
multimedia session via the interface specified in the W3C
RTCPeerConnection API and discusses how this relates to existing
signaling protocols.IntroductionThis document describes how the W3C Web Real-Time Communication (WebRTC) RTCPeerConnection
interface
is used to control the setup,
management, and teardown of a multimedia session.General Design of JSEPWebRTC call setup has been designed to focus on controlling
the media plane, leaving signaling-plane behavior up to the
application as much as possible. The rationale is that
different applications may prefer to use different protocols,
such as the existing SIP call signaling protocol, or something
custom to the particular application, perhaps for a novel use
case. In this approach, the key information that needs to be
exchanged is the multimedia session description, which
specifies the transport and media configuration
information necessary to establish the media plane.With these considerations in mind, this document describes
the JavaScript Session Establishment Protocol (JSEP), which
allows for full control of the signaling state machine from
JavaScript. As described above, JSEP assumes a model in which a
JavaScript application executes inside a runtime containing
WebRTC APIs (the "JSEP implementation"). The JSEP
implementation is almost entirely divorced from the core
signaling flow, which is instead handled by the JavaScript
making use of two interfaces: (1) passing in local and remote
session descriptions and (2) interacting with the Interactive
Connectivity Establishment (ICE) state
machine . The combination of the JSEP implementation and the
JavaScript application is referred to throughout this document
as a "JSEP endpoint".In this document, the use of JSEP is described as if it
always occurs between two JSEP endpoints. Note, though, that in many
cases it will actually be between a JSEP endpoint and some kind
of server, such as a gateway or Multipoint Control Unit (MCU). This distinction is
invisible to the JSEP endpoint; it just follows the
instructions it is given via the API.JSEP's handling of session descriptions is simple and
straightforward. Whenever an offer/answer exchange is needed,
the initiating side creates an offer by calling a createOffer
API. The application then uses that offer to set up its local
configuration via the setLocalDescription API. The offer is finally
sent off to the remote side over its preferred signaling
mechanism (e.g., WebSockets); upon receipt of that offer, the
remote party installs it using the setRemoteDescription
API.To complete the offer/answer exchange, the remote party uses
the createAnswer API to generate an appropriate answer,
applies it using the setLocalDescription API, and sends the
answer back to the initiator over the signaling channel. When
the initiator gets that answer, it installs it using the
setRemoteDescription API, and initial setup is complete. This
process can be repeated for additional offer/answer
exchanges.Regarding ICE
, JSEP decouples the ICE state
machine from the overall signaling state machine. The ICE
state machine must remain in the JSEP implementation because
only the implementation has the necessary knowledge of
candidates and other transport information. Performing this
separation provides additional flexibility in protocols that
decouple session descriptions from transport. For instance, in
traditional SIP, each offer or answer is self-contained,
including both the session descriptions and the transport
information. However,
allows SIP to
be used with Trickle ICE
, in which the session
description can be sent immediately and the transport
information can be sent when available. Sending transport
information separately can allow for faster ICE and DTLS
startup, since ICE checks can start as soon as any transport
information is available rather than waiting for all of it.
JSEP's decoupling of the ICE and signaling state machines
allows it to accommodate either model.Although it abstracts signaling, the JSEP approach
requires that the application be aware of the signaling process.
While the application does not need to understand the contents
of session descriptions to set up a call, the application must
call the right APIs at the right times, convert the session
descriptions and ICE information into the defined messages of
its chosen signaling protocol, and perform the reverse
conversion on the messages it receives from the other side.One way to make life easier for the application is to
provide a JavaScript library that hides this complexity from
the developer; said library would implement a given signaling
protocol along with its state machine and serialization code,
presenting a higher-level call-oriented interface to the
application developer. For example, libraries exist to provide
implementations of the SIP and Extensible Messaging
and Presence Protocol (XMPP) signaling
protocols atop the JSEP API.
Thus, JSEP
provides greater control for the experienced developer without
forcing any additional complexity on the novice developer.Other Approaches ConsideredOne approach that was considered instead of JSEP was to
include a lightweight signaling protocol. Instead of providing
session descriptions to the API, the API would produce and
consume messages from this protocol. While providing a more
high-level API, this put more control of signaling within the
JSEP implementation, forcing it to have to understand and
handle concepts like signaling glare (see
).A second approach that was considered but not chosen was to
decouple the management of the media control objects from
session descriptions, instead offering APIs that would control
each component directly. This was rejected based on the
argument that requiring exposure of this level of complexity to
the application programmer would not be beneficial; it would
(1) result in an API where even a simple example would require a
significant amount of code to orchestrate all the needed
interactions and (2) create a large API surface that
would need to be agreed upon and documented.
In addition, these API
points could be called in any order, resulting in a more
complex set of interactions with the media subsystem than the
JSEP approach, which specifies how session descriptions are to
be evaluated and applied.One variation on JSEP that was considered was to keep the
basic session-description-oriented API but to move the
mechanism for generating offers and answers out of the JSEP
implementation. Instead of providing createOffer/createAnswer
methods within the implementation, this approach would instead
expose a getCapabilities API, which would provide the
application with the information it needed in order to generate
its own session descriptions. This increases the amount of work
that the application needs to do; it needs to know how to
generate session descriptions from capabilities, and especially
how to generate the correct answer from an arbitrary offer and
the supported capabilities. While this could certainly be
addressed by using a library like the one mentioned above, it
basically forces the use of said library even for a simple
example. Providing createOffer/createAnswer avoids this
problem.Contradiction regarding bundle-only "m=" sectionsSince the approval of the WebRTC specification documents, the IETF has become
aware of an inconsistency between the document specifying JSEP and the document specifying BUNDLE (this RFC and , respectively). Rather than delaying publication further to come to a resolution, the documents are being published as they were originally approved. The IETF intends to restart work on these technologies, and revised versions of these documents will be published as soon as a resolution becomes available.The specific issue involves the handling of "m=" sections that are designated as bundle-only, as discussed in of this RFC. Currently, there is divergence between JSEP and BUNDLE, as well as between these specifications and existing browser implementations:
JSEP prescribes that said "m=" sections should use port zero and add an "a=bundle-only" attribute in initial offers, but not in answers or subsequent offers.
BUNDLE prescribes that these "m=" sections should be marked as described in the previous point, but in all offers and answers.
Most current browsers do not mark any "m=" sections with port zero and instead use the same port for all bundled "m=" sections; some others follow the JSEP behavior.
TerminologyThe key words "MUST", "MUST NOT",
"REQUIRED", "SHALL",
"SHALL NOT", "SHOULD",
"SHOULD NOT",
"RECOMMENDED", "NOT RECOMMENDED",
"MAY", and "OPTIONAL" in this document are
to be interpreted as described in BCP 14 when, and only when, they appear in all capitals,
as shown here.Semantics and SyntaxSignaling ModelJSEP does not specify a particular signaling model or state
machine, other than the generic need to exchange session
descriptions in the fashion described by
(offer/answer) in order for both
sides of the session to know how to conduct the session. JSEP
provides mechanisms to create offers and answers, as well as to
apply them to a session. However, the JSEP implementation is
totally decoupled from the actual mechanism by which these
offers and answers are communicated to the remote side,
including addressing, retransmission, forking, and glare
handling. These issues are left entirely up to the application;
the application has complete control over which offers and
answers get handed to the implementation, and when.Session Descriptions and State MachineIn order to establish the media plane, the JSEP
implementation needs specific parameters to indicate what to
transmit to the remote side, as well as how to handle the media
that is received. These parameters are determined by the
exchange of session descriptions in offers and answers, and
there are certain details to this process that must be handled
in the JSEP APIs.Whether a session description applies to the local side or
the remote side affects the meaning of that description. For
example, the list of codecs sent to a remote party indicates
what the local side is willing to receive, which, when
intersected with the set of codecs the remote side supports,
specifies what the remote side should send. However, not all
parameters follow this rule; some parameters are declarative,
and the remote side must either accept them or reject them
altogether. An example of such a parameter is the TLS
fingerprints
as used in the context of DTLS ;
these fingerprints are calculated based on
the local certificate(s) offered and are not subject to
negotiation.
In addition, various RFCs put different conditions on the
format of offers versus answers. For example, an offer may
propose an arbitrary number of "m=" sections (i.e., media
descriptions as described in
), but an answer must
contain the exact same number as the offer.Lastly, while the exact media parameters are known only
after an offer and an answer have been exchanged, the offerer
may receive ICE checks, and possibly media (e.g., in the case
of a re-offer after a connection has been established) before
it receives an answer. To properly process incoming media in
this case, the offerer's media handler must be aware of the
details of the offer before the answer arrives.Therefore, in order to handle session descriptions properly,
the JSEP implementation needs:
To know if a session description pertains to the local or
remote side.
To know if a session description is an offer or an
answer.
To allow the offer to be specified independently of the
answer.
JSEP addresses this by adding both setLocalDescription
and setRemoteDescription methods and having session description
objects contain a type field indicating the type of session
description being supplied. This satisfies the requirements
listed above for both the offerer, who first calls
setLocalDescription(sdp [offer]) and then later
setRemoteDescription(sdp [answer]), and the
answerer, who first calls setRemoteDescription(sdp [offer]) and
then later setLocalDescription(sdp [answer]).During the offer/answer exchange, the outstanding offer is
considered to be "pending" at the offerer and the answerer, as
it may be either accepted or rejected. If this is a re-offer,
each side will also have "current" local and remote
descriptions, which reflect the result of the last offer/answer
exchange. Sections
,
,
, and
provide more
detail on pending and current descriptions.JSEP also allows for an answer to be treated as provisional
by the application. Provisional answers provide a way for an
answerer to communicate initial session parameters back to the
offerer, in order to allow the session to begin, while allowing
a final answer to be specified later. This concept of a final
answer is important to the offer/answer model; when such an
answer is received, any extra resources allocated by the caller
can be released, now that the exact session configuration is
known. These "resources" can include things like extra ICE
components, Traversal Using Relays around NAT (TURN) candidates, or video decoders. Provisional
answers, on the other hand, do no such deallocation; as a
result, multiple dissimilar provisional answers, with their own
codec choices, transport parameters, etc., can be received and
applied during call setup. Note that the final answer itself
may be different than any received provisional answers.In
, the constraint at the signaling
level is that only one offer can be outstanding for a given
session, but at the JSEP level, a new offer can be
generated at any point. For example, when using SIP for
signaling, if one offer is sent and is then canceled using a SIP
CANCEL, another offer can be generated even though no answer
was received for the first offer. To support this, the JSEP
media layer can provide an offer via the createOffer method
whenever the JavaScript application needs one for the
signaling. The answerer can send back zero or more provisional
answers and then finally end the offer/answer exchange by sending a
final answer. The state machine for this is as follows:Aside from these state transitions, there is no other
difference between the handling of provisional ("pranswer") and
final ("answer") answers.Session Description FormatJSEP's session descriptions use Session Description Protocol (SDP) syntax for their
internal representation. While this format is not optimal for
manipulation from JavaScript, it is widely accepted and is
frequently updated with new features; any alternate encoding of
session descriptions would have to keep pace with the changes
to SDP, at least until the time that this new encoding eclipsed
SDP in popularity.However, to provide for future flexibility, the SDP syntax
is encapsulated within a SessionDescription object, which can
be constructed from SDP and be serialized out to SDP. If
future specifications agree on a JSON format for session
descriptions, we could easily enable this object to generate
and consume that JSON.As detailed below, most applications should be able to treat
the SessionDescriptions produced and consumed by these various
API calls as opaque blobs; that is, the application will not
need to read or change them.Session Description ControlIn order to give the application control over various common
session parameters, JSEP provides control surfaces that tell
the JSEP implementation how to generate session descriptions.
This avoids the need for JavaScript to modify session
descriptions in most cases.Changes to these objects result in changes to the session
descriptions generated by subsequent createOffer/createAnswer
calls.RtpTransceiversRtpTransceivers 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
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.RtpSendersRtpSenders 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).RtpReceiversRtpReceivers 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).ICEICE Gathering OverviewJSEP 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
.ICE Candidate TricklingCandidate 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
. 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.ICE Candidate FormatIn JSEP, ICE candidates are abstracted by an
IceCandidate object, and as with session descriptions, SDP
syntax is used for the internal representation.The candidate details are specified in an IceCandidate
field, using the same SDP syntax as the
"candidate-attribute" field defined in
. Note that this
field does not contain an "a=" prefix, as indicated in the
following example:The IceCandidate object contains a field to indicate
which ICE username fragment (ufrag) it is associated with, as defined in
. This value is used
to determine which session description (and thereby which
gathering phase) this IceCandidate belongs to, which helps
resolve ambiguities during ICE restarts. If this field is
absent in a received IceCandidate (perhaps when
communicating with a non-JSEP endpoint), the most recently
received session description is assumed.The IceCandidate object also contains fields to indicate
which "m=" section it is associated with, which can be
identified in one of two ways: either by an "m=" section
index or by a MID. The "m=" section index is a zero-based
index, with index N referring to the N+1th "m=" section in
the session description referenced by this IceCandidate.
The MID is a "media stream identification" value, as
defined in
, which provides a
more robust way to identify the "m=" section in the session
description, using the MID of the associated RtpTransceiver
object (which may have been locally generated by the
answerer when interacting with a non-JSEP endpoint that
does not support the MID attribute, as discussed in
below). If the
MID field is present in a received IceCandidate, it MUST be
used for identification; otherwise, the "m=" section index is
used instead.Implementations MUST
be prepared to receive objects with some fields missing, as
mentioned above.ICE Candidate PolicyTypically, 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
.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.ICE Candidate PoolJSEP 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".)ICE VersionsWhile this specification formally relies on , at the time of its publication, the
majority of WebRTC implementations support the version
of ICE described in . The "ice2" attribute defined in
can be used to detect the version in use by a remote endpoint
and to provide a smooth transition from the older specification
to the newer one. Implementations MUST be able to accept remote
descriptions that do not have the "ice2" attribute.Video Size NegotiationVideo size negotiation is the process through which a
receiver can use the "a=imageattr" SDP attribute
to indicate what video frame sizes it
is capable of receiving. A receiver may have hard limits on
what its video decoder can process, or it may have some maximum
set by policy. By specifying these limits in an "a=imageattr"
attribute, JSEP endpoints can attempt to ensure that the remote
sender transmits video at an acceptable resolution. However,
when communicating with a non-JSEP endpoint that does not
understand this attribute, any signaled limits may be exceeded,
and the JSEP implementation MUST handle this gracefully, e.g.,
by discarding the video.Note that certain codecs support transmission of samples
with aspect ratios other than 1.0 (i.e., non-square pixels).
JSEP implementations will not transmit non-square pixels but
SHOULD receive and render such video with the correct aspect
ratio. However, sample aspect ratio has no impact on the size
negotiation described below; all dimensions are measured in
pixels, whether square or not.Creating an imageattr AttributeThe 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:This declaration indicates that the receiver is capable of
decoding any image resolution from 48x48 up to 1280x720
pixels.Interpreting imageattr Attributes 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
() 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)
), 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.
SimulcastJSEP supports simulcast transmission of a MediaStreamTrack,
where multiple encodings of the source media can be transmitted
within the context of a single "m=" section. The current JSEP API
is designed to allow applications to send simulcasted media but
only to receive a single encoding. This allows for multi-user
scenarios where each sending client sends multiple encodings to
a server, which then, for each receiving client, chooses the
appropriate encoding to forward.Applications request support for simulcast by configuring
multiple encodings on an RtpSender. Upon generation of an offer
or answer, these encodings are indicated via SDP markings on
the corresponding "m=" section, as described below. Receivers
that understand simulcast and are willing to receive it will
also include SDP markings to indicate their support, and JSEP
endpoints will use these markings to determine whether
simulcast is permitted for a given RtpSender. If simulcast
support is not negotiated, the RtpSender will only use the
first configured encoding.Note that the exact simulcast parameters are up to the
sending application. While the aforementioned SDP markings are
provided to ensure that the remote side can receive and demux
multiple simulcast encodings, the specific resolutions and
bitrates to be used for each encoding are purely a send-side
decision in JSEP.JSEP currently does not provide a mechanism to configure
receipt of simulcast. This means that if simulcast is offered
by the remote endpoint, the answer generated by a JSEP endpoint
will not indicate support for receipt of simulcast, and as such
the remote endpoint will only send a single encoding per "m="
section.In addition, JSEP does not provide a mechanism to handle an
incoming offer requesting simulcast from the JSEP endpoint.
This means that setting up simulcast in the case where the JSEP
endpoint receives the initial offer requires out-of-band
signaling or SDP inspection. However, in the case where the
JSEP endpoint sets up simulcast in its initial offer, any
established simulcast streams will continue to work upon
receipt of an incoming re-offer. Future versions of this
specification may add additional APIs to handle the incoming
initial offer scenario.When using JSEP to transmit multiple encodings from an
RtpSender, the techniques from
and
are used. Specifically,
when multiple encodings have been configured for an RtpSender,
the "m=" section for the RtpSender will include an "a=simulcast"
attribute, as defined in
,
with a "send" simulcast stream description that lists each
desired encoding, and no "recv" simulcast stream description.
The "m=" section will also include an "a=rid" attribute for each
encoding, as specified in
; the use of
Restriction Identifiers (RIDs, also called rid-ids or RtpStreamIds)
allows the individual encodings to be
disambiguated even though they are all part of the same "m="
section.Interactions with ForkingSome call signaling systems allow various types of forking
where an SDP Offer may be provided to more than one device. For
example, SIP
defines both a "parallel search"
and "sequential search". Although these are primarily signaling-level issues that are outside the scope of JSEP, they do have
some impact on the configuration of the media plane that is
relevant. When forking happens at the signaling layer, the
JavaScript application responsible for the signaling needs to
make the decisions about what media should be sent or received
at any point in time, as well as which remote endpoint it
should communicate with; JSEP is used to make sure the media
engine can make the RTP and media perform as required by the
application. The basic operations that the applications can
have the media engine do are as follows:
Start exchanging media with a given remote peer, but keep
all the resources reserved in the offer.
Start exchanging media with a given remote peer, and free
any resources in the offer that are not being used.
Sequential ForkingSequential 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.Parallel ForkingParallel 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
. 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
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
) 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
.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
), 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.InterfaceThis section details the basic operations that must be present
to implement JSEP functionality. The actual API exposed in the
W3C API may have somewhat different syntax but should map easily
to these concepts.
PeerConnectionConstructorThe 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
, 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&nbhy;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
. 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
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".addTrackThe 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
. addTrack attempts
to minimize the number of transceivers as follows: if the
PeerConnection is in the "have&nbhy;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
.removeTrackThe 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).addTransceiverThe 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
).onaddtrack EventThe 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.
createDataChannelThe 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.ondatachannel EventThe 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.
createOfferThe createOffer method generates a blob of SDP that
contains an offer per 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
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
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.createAnswerThe createAnswer method generates a blob of SDP that
contains an SDP answer per 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
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.SessionDescriptionTypeSession 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
. The contents MUST be
empty.Use of Provisional AnswersMost applications will not need to create answers using
the "pranswer" type. While it is good practice to send an
immediate response to an offer, in order to warm up the
session transport and prevent media clipping, the preferred
handling for a JSEP application is to create and send a
"sendonly" final answer with a null MediaStreamTrack
immediately after receiving the offer, which will prevent
media from being sent by the caller and allow media to be
sent immediately upon answer by the callee. Later, when the
callee actually accepts the call, the application can plug
in the real MediaStreamTrack and create a new "sendrecv"
offer to update the previous offer/answer pair and start
bidirectional media flow. While this could also be done
with a "sendonly" pranswer followed by a "sendrecv"
answer, the initial pranswer leaves the offer/answer
exchange open, which means that the caller cannot send an
updated offer during this time. As an example, consider a typical JSEP application that
wants to set up audio and video as quickly as possible.
When the callee receives an offer with audio and video
MediaStreamTracks, it will send an immediate answer
accepting these tracks as sendonly (meaning that the caller
will not send the callee any media yet, and because the
callee has not yet added its own MediaStreamTracks, the
callee will not send any media either). It will then ask
the user to accept the call and acquire the needed local
tracks. Upon acceptance by the user, the application will
plug in the tracks it has acquired, which, because ICE handshaking
and DTLS handshaking have likely completed by this point, can
start transmitting immediately. The application will also
send a new offer to the remote side indicating call
acceptance and moving the audio and video to be two-way
media. A detailed example flow along these lines is shown
in
.Of course, some applications may not be able to perform
this double offer/answer exchange, particularly ones that
are attempting to gateway to legacy signaling protocols. In
these cases, pranswer can still provide the application
with a mechanism to warm up the transport.RollbackIn certain situations, it may be desirable to "undo" a
change made to setLocalDescription or setRemoteDescription.
Consider a case where a call is ongoing and one side wants
to change some of the session parameters; that side
generates an updated offer and then calls
setLocalDescription. However, the remote side, either
before or after setRemoteDescription, decides it does not
want to accept the new parameters and sends a reject
message back to the offerer. Now, the offerer, and possibly
the answerer as well, needs to return to a "stable" state and
the previous local/remote description. To support this, we
introduce the concept of "rollback", which discards any
proposed changes to the session, returning the state
machine to the "stable" state. A rollback is performed by
supplying a session description of type "rollback" with
empty contents to either setLocalDescription or
setRemoteDescription.setLocalDescriptionThe 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.
setRemoteDescriptionThe 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.currentLocalDescriptionThe 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.pendingLocalDescriptionThe 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".currentRemoteDescriptionThe 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.pendingRemoteDescriptionThe 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".canTrickleIceCandidatesThe 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
, 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.setConfigurationThe 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
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.addIceCandidateThe addIceCandidate method provides an update to the ICE
agent via an IceCandidate object
(). 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
.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
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.onicecandidate EventThe 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
and
subject to the restrictions discussed in
, or
(2) when an ICE gathering phase completes. The event contains a single
IceCandidate object, as defined in
.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
.RtpTransceiverstopThe 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.stoppedThe 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.setDirectionThe 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
.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
below.Note that while setDirection sets the direction property
of the transceiver immediately (), 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.directionThe 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.currentDirectionThe currentDirection property indicates the last
negotiated direction for the transceiver's associated "m="
section. More specifically, it indicates the
direction attribute 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".setCodecPreferencesThe 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.SDP Interaction ProceduresThis section describes the specific procedures to be followed
when creating and parsing SDP objects.Requirements OverviewJSEP implementations MUST comply with the specifications
listed below that govern the creation and processing of offers
and answers.Usage RequirementsAll 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
, 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 .
DTLS
or DTLS-SRTP
MUST be used, as
appropriate for the media type, as specified in
.
The SDP security descriptions mechanism for SRTP keying
MUST NOT be used, as discussed in
.Profile Names and InteroperabilityFor media "m=" sections, JSEP implementations MUST support
the "UDP/TLS/RTP/SAVPF" profile specified in
as well as the "TCP/DTLS/RTP/SAVPF"
profile specified in 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
.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
)
"RTP/AVPF" (defined in
)
"RTP/SAVP" (defined in
)
"RTP/SAVPF" (defined in
)
"TCP/DTLS/RTP/SAVP" (defined in
)
"TCP/DTLS/RTP/SAVPF" (defined in
)
"UDP/TLS/RTP/SAVP" (defined in
)
"UDP/TLS/RTP/SAVPF" (defined in
)
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.Constructing an OfferWhen createOffer is called, a new SDP description MUST be
created that includes the functionality specified in
. The exact
details of this process are explained below.Initial OffersWhen 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
. Specifically:
The first SDP line MUST be "v=0" as defined in
.
The second SDP line MUST be an "o=" line as defined
in
.
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
. As mentioned in
, 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
; 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
, 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
; 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 and
.
If WebRTC identity is being used, an "a=identity" line
MUST be added, as described in
.
The next step is to generate "m=" sections, as specified in
. 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
.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
, or these
values MUST be session-level attributes.Each "m=" section MUST be generated as specified in
. 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
.
To properly indicate use of DTLS, the <proto>
field MUST be set to "UDP/TLS/RTP/SAVPF", as specified in
.
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
and
.
The "m=" line MUST be followed immediately by a "c=" line,
as specified in
. Again, as no
candidates are available yet, the "c=" line MUST contain the
default value "IN IP4 0.0.0.0", as defined in
. 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
.
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
. 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
.
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
and
.
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
.
For each supported Forward Error Correction (FEC) mechanism, "a=rtpmap" and
"a=fmtp" lines, as specified in
. The FEC
mechanisms that MUST be supported are specified in
,
and specific usage for each media type is outlined in
Sections and .
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
. 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
.
For each supported RTP header extension, an "a=extmap"
line, as specified in
.
The list of
header extensions that SHOULD/MUST be supported is
specified in
. Any header extensions that require encryption MUST
be specified as indicated in
.
For each supported RTCP feedback mechanism, an
"a=rtcp-fb" line, as specified in
. The list of
RTCP feedback mechanisms that SHOULD/MUST be supported is
specified in
.
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
,
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
, 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 ), and SHOULD be 3 bytes
or less, to allow them to efficiently fit into the RTP
header extensions defined in
.
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
. 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
.
For each desired digest algorithm, one or more
"a=fingerprint" lines for each of the endpoint's
certificates, as specified in
.
An "a=setup" line, as specified in
and clarified
for use in DTLS-SRTP scenarios in
. The role value
in the offer MUST be "actpass".
An "a=tls-id" line, as specified in
.
An "a=rtcp" line, as specified in
, 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
.
If the RTP/RTCP multiplexing policy is "require", an
"a=rtcp-mux-only" line, as specified in
.
An "a=rtcp-rsize" line, as specified in
.
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"
. The <fmt>
value MUST be set to "webrtc-datachannel" as specified in
.
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
;
and, if appropriate, an "a=max-message-size" line, as defined
in
.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
. 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
. 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
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
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.Subsequent OffersWhen 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
and clarified in
.
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
() 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
. If no RTCP
candidates have yet been gathered, default values MUST be
used, as described in 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
), an
"a=candidate" line MUST be added, as defined in
.
If candidate gathering for the section has completed, an
"a=end-of-candidates" attribute MUST be added, as described
in
.
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
,
MUST NOT be added.
Instead, JSEP implementations MUST simply omit
parameters in the IDENTICAL and TRANSPORT categories for
bundled "m=" sections, as described in
.
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.Options HandlingThe createOffer method takes as a parameter an
RTCOfferOptions object. Special processing is performed when
generating an SDP description if the following options are
present.IceRestartIf the IceRestart option is specified, with a value of
"true", the offer MUST indicate an ICE restart by
generating new ICE ufrag and pwd attributes, as specified
in
. If this
option is specified on an initial offer, it has no effect
(since a new ICE ufrag and pwd are already generated).
Similarly, if the ICE configuration has changed, this
option has no effect, since new ufrag and pwd attributes
will be generated automatically. This option is primarily
useful for reestablishing connectivity in cases where
failures are detected by the application.VoiceActivityDetectionSilence suppression, also known as discontinuous
transmission ("DTX"), can reduce the bandwidth used for
audio by switching to a special encoding when voice
activity is not detected, at the cost of some fidelity.If the "VoiceActivityDetection" option is specified,
with a value of "true", the offer MUST indicate support for
silence suppression in the audio it receives by including
comfort noise ("CN") codecs for each offered audio codec,
as specified in
, except for
codecs that have their own internal silence suppression
support. For codecs that have their own internal silence
suppression support, the appropriate fmtp parameters for
that codec MUST be specified to indicate that silence
suppression for received audio is desired. For example,
when using the Opus codec
, the "usedtx=1" parameter,
specified in
, would be used in the offer.If the "VoiceActivityDetection" option is specified,
with a value of "false", the JSEP implementation MUST NOT
emit "CN" codecs. For codecs that have their own internal
silence suppression support, the appropriate fmtp
parameters for that codec MUST be specified to indicate
that silence suppression for received audio is not desired.
For example, when using the Opus codec, the "usedtx=0"
parameter would be specified in the offer. In addition, the
implementation MUST NOT use silence suppression for media
it generates, regardless of whether the "CN" codecs or
related fmtp parameters appear in the peer's description.
The impact of these rules is that silence suppression in
JSEP depends on mutual agreement of both sides, which
ensures consistent handling regardless of which codec is
used.The "VoiceActivityDetection" option does not have any
impact on the setting of the "vad" value in the signaling
of the client-to-mixer audio level header extension
described in
.Generating an AnswerWhen createAnswer is called, a new SDP description MUST be
created that is compatible with the supplied remote description
as well as the requirements specified in
. The exact
details of this process are explained below.Initial AnswersWhen 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
. 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 above, except that the "a=ice-options" line, with the
"trickle" option as specified in
and the "ice2" option as specified in
, 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
. 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
, a group of type "LS" has
been added that references each track's RtpTransceiver.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: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.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.The example in 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
. 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
. 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
, 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
.
Otherwise, each "m=" section in the answer MUST then be
generated as specified in
. 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
.
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
. 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
. Again, as no
candidates are available yet, the "c=" line MUST contain the
default value "IN IP4 0.0.0.0", as defined in
.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
. The MID
value MUST match that specified in the offer.
A direction attribute, determined by applying the rules
regarding the offered direction specified in
, 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
and
.
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
.
For each supported FEC mechanism, "a=rtpmap" and
"a=fmtp" lines, as specified in
. The FEC
mechanisms that MUST be supported are specified in
, and
specific usage for each media type is outlined in Sections
and .
If this "m=" section is for media with configurable
durations of media per packet, e.g., audio, an "a=maxptime"
line, as described in
.
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
.
For each supported RTP header extension that is present
in the offer, an "a=extmap" line, as specified in
. The list of
header extensions that SHOULD/MUST be supported is
specified in
. Any header extensions that require encryption MUST be
specified as indicated in
.
For each supported RTCP feedback mechanism that is
present in the offer, an "a=rtcp-fb" line, as specified in
. The list of
RTCP feedback mechanisms that SHOULD/MUST be supported is
specified in
.
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
,
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
.
For each desired digest algorithm, one or more
"a=fingerprint" lines for each of the endpoint's
certificates, as specified in
.
An "a=setup" line, as specified in
and clarified
for use in DTLS-SRTP scenarios in
. 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
.
If present in the offer, an "a=rtcp-mux" line, as
specified in
. Otherwise,
an "a=rtcp" line, as specified in
, 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
.
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
;
and, if appropriate, an "a=max-message-size" line, as defined
in
.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
. 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.Subsequent AnswersWhen 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
. 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
. 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
.
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
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
), an
"a=candidate" line MUST be added, as defined in
.
If candidate gathering for the section has completed, an
"a=end-of-candidates" attribute MUST be added, as described
in
.
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.
Options HandlingThe 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.VoiceActivityDetectionSilence suppression in the answer is handled as
described in
, with
one exception: if support for silence suppression was not
indicated in the offer, the VoiceActivityDetection
parameter has no effect, and the answer MUST be generated
as if VoiceActivityDetection was set to "false". This is done
on a per-codec basis (e.g., if the offerer somehow offered
support for CN but set "usedtx=0" for Opus, setting
VoiceActivityDetection to "true" would result in an answer
with CN codecs and "usedtx=0"). The impact of this rule is
that an answerer will not try to use silence suppression
with any endpoint that does not offer it, making silence
suppression support bilateral even with non-JSEP
endpoints.Modifying an Offer or AnswerThe SDP returned from createOffer or createAnswer MUST NOT
be changed before passing it to setLocalDescription. If precise
control over the SDP is needed, the aforementioned
createOffer/createAnswer options or RtpTransceiver APIs MUST be
used.After calling setLocalDescription with an offer or answer,
the application MAY modify the SDP to reduce its capabilities
before sending it to the far side, as long as it follows the
rules above that define a valid JSEP offer or answer. Likewise,
an application that has received an offer or answer from a peer
MAY modify the received SDP, subject to the same constraints,
before calling setRemoteDescription.As always, the application is solely responsible for what it
sends to the other party, and all incoming SDP will be
processed by the JSEP implementation to the extent of its
capabilities. It is an error to assume that all SDP is
well formed; however, one should be able to assume that any
implementation of this specification will be able to process,
as a remote offer or answer, unmodified SDP coming from any
other implementation of this specification.Processing a Local DescriptionWhen a SessionDescription is supplied to
setLocalDescription, the following steps MUST be performed:
If the description is of type "rollback", follow the
processing defined in
and skip the
processing described in the rest of this section.
Otherwise, the type of the SessionDescription is checked
against the current state of the PeerConnection:
If the type is "offer", the PeerConnection state MUST be
either "stable" or "have-local-offer".
If the type is "pranswer" or "answer", the
PeerConnection state MUST be either "have-remote-offer" or
"have-local-pranswer".
If the type is not correct for the current state,
processing MUST stop and an error MUST be returned.
The SessionDescription is then checked to ensure that its
contents are identical to those generated in the last call to
createOffer/createAnswer, and thus have not been altered, as
discussed in
; otherwise, processing
MUST stop and an error MUST be returned.
Next, the SessionDescription is parsed into a data
structure, as described in
below.
Finally, the parsed SessionDescription is applied as
described in
below.
Processing a Remote DescriptionWhen a SessionDescription is supplied to
setRemoteDescription, the following steps MUST be performed:
If the description is of type "rollback", follow the
processing defined in
and skip the
processing described in the rest of this section.
Otherwise, the type of the SessionDescription is checked
against the current state of the PeerConnection:
If the type is "offer", the PeerConnection state MUST be
either "stable" or "have-remote-offer".
If the type is "pranswer" or "answer", the
PeerConnection state MUST be either "have-local-offer" or
"have-remote-pranswer".
If the type is not correct for the current state,
processing MUST stop and an error MUST be returned.
Next, the SessionDescription is parsed into a data
structure, as described in
below. If parsing fails
for any reason, processing MUST stop and an error MUST be
returned.
Finally, the parsed SessionDescription is applied as
described in
below.
Processing a RollbackA rollback may be performed if the PeerConnection is in any
state except for "stable". This means that both offers and
provisional answers can be rolled back. Rollback can only be
used to cancel proposed changes; there is no support for
rolling back from a "stable" state to a previous "stable" state. If
a rollback is attempted in the "stable" state, processing MUST
stop and an error MUST be returned. Note that this implies that
once the answerer has performed setLocalDescription with its
answer, this cannot be rolled back.The effect of rollback MUST be the same regardless of
whether setLocalDescription or setRemoteDescription is
called.In order to process rollback, a JSEP implementation abandons
the current offer/answer transaction, sets the signaling state
to "stable", and sets the pending local and/or remote
description (see Sections
and
) to "null". Any
resources or candidates that were allocated by the abandoned
local description are discarded; any media that is received is
processed according to the previous local and remote
descriptions.A rollback disassociates any RtpTransceivers that were
associated with "m=" sections by the application of the
rolled-back session description (see Sections
and
).
This means that
some RtpTransceivers that were previously associated will no
longer be associated with any "m=" section; in such cases, the
value of the RtpTransceiver's mid property MUST be set to "null",
and the mapping between the transceiver and its "m=" section
index MUST be discarded. RtpTransceivers that were created by
applying a remote offer that was subsequently rolled back MUST
be stopped and removed from the PeerConnection. However, an
RtpTransceiver MUST NOT be removed if a track was attached to
the RtpTransceiver via the addTrack method. This is so that an
application may call addTrack, then call setRemoteDescription
with an offer, then roll back that offer, then call createOffer
and have an "m=" section for the added track appear in the
generated offer.Parsing a Session DescriptionThe SDP contained in the session description object consists
of a sequence of text lines, each containing a key-value
expression, as described in
. The SDP is read,
line by line, and converted to a data structure that contains
the deserialized information. However, SDP allows many types of
lines, not all of which are relevant to JSEP applications. For
each line, the implementation will first ensure that it is
syntactically correct according to its defining ABNF, check
that it conforms to the semantics used in
and
, and then either parse and
store or discard the provided value, as described below.If any line is not well formed or cannot be parsed as
described, the parser MUST stop with an error and reject the
session description, even if the value is to be discarded. This
ensures that implementations do not accidentally misinterpret
ambiguous SDP.Session-Level ParsingFirst, the session-level lines are checked and parsed.
These lines MUST occur in a specific order, and with a
specific syntax, as defined in
. 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
. 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
.
The "o=" line MUST be parsed as specified in
.
The "b=" line, if present, MUST be parsed as specified
in
, 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
, and the group's
semantics and mids are stored.
If present, a single "a=ice-lite" line is parsed as
specified in
, and a value
indicating the presence of ice-lite is stored.
If present, a single "a=ice-ufrag" line is parsed as
specified in
, and the ufrag value is stored.
If present, a single "a=ice-pwd" line is parsed as
specified in
, and the password value is stored.
If present, a single "a=ice-options" line is parsed as
specified in
, and the set of specified options is stored.
Any "a=fingerprint" lines are parsed as specified in
, and the set of
fingerprint and algorithm values is stored.
If present, a single "a=setup" line is parsed as
specified in
, and the setup value
is stored.
If present, a single "a=tls-id" line is parsed as
specified in , and
the attribute value is stored.
Any "a=identity" lines are parsed and the identity
values stored for subsequent verification, as specified in
.
Any "a=extmap" lines are parsed as specified in
, 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
, unknown
attribute lines MUST be ignored.Once all the session-level lines have been parsed,
processing continues with the lines in "m=" sections.Media Section ParsingLike the session-level lines, the media section lines MUST
occur in the specific order and with the specific syntax
defined in
.The "m=" line itself MUST be parsed as described in
, 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
, but its value is
not used.
The "b=" line, if present, MUST be parsed as specified
in
, 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
, and the ufrag value is stored.
If present, a single "a=ice-pwd" line is parsed as
specified in
, and the password value is stored.
If present, a single "a=ice-options" line is parsed as
specified in
,
and the set of specified options is stored.
Any "a=candidate" attributes MUST be parsed as specified
in
, and their values stored.
Any "a=remote-candidates" attributes MUST be parsed as
specified in
, but their values are ignored.
If present, a single "a=end-of-candidates" attribute
MUST be parsed as specified in
, and
its presence or absence flagged and stored.
Any "a=fingerprint" lines are parsed as specified in
, and the set of
fingerprint and algorithm values is stored.
If the "m=" <proto> value indicates use of RTP, as described
in
above, the following
attribute lines MUST be processed:
The "m=" <fmt> value MUST be parsed as specified in
, and the individual
values stored.
Any "a=rtpmap" or "a=fmtp" lines MUST be parsed as
specified in
, and their values
stored.
If present, a single "a=ptime" line MUST be parsed as
described in
, and its value
stored.
If present, a single "a=maxptime" line MUST be parsed as
described in
, and its value
stored.
If present, a single direction attribute line (e.g.,
"a=sendrecv") MUST be parsed as described in
, and its value
stored.
Any "a=ssrc" attributes MUST be parsed as specified in
, and their values
stored.
Any "a=extmap" attributes MUST be parsed as specified in
, and their values
stored.
Any "a=rtcp-fb" attributes MUST be parsed as specified
in
, and their values
stored.
If present, a single "a=rtcp-mux" attribute MUST be
parsed as specified in
, and its
presence or absence flagged and stored.
If present, a single "a=rtcp-mux-only" attribute MUST be
parsed as specified in
,
and its presence or absence flagged and stored.
If present, a single "a=rtcp-rsize" attribute MUST be
parsed as specified in
, and its presence or
absence flagged and stored.
If present, a single "a=rtcp" attribute MUST be parsed
as specified in
, but its value is
ignored, as this information is superfluous when using
ICE.
If present, "a=msid" attributes MUST be parsed as
specified in
, 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
, and their values
stored.
Any "a=rid" lines MUST be parsed as specified in
, and
their values stored.
If present, a single "a=simulcast" line MUST be parsed
as specified in
, 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
,
and the application protocol value stored.
An "a=sctp-port" attribute MUST be present, and it MUST
be parsed as specified in
,
and the value stored.
If present, a single "a=max-message-size" attribute MUST
be parsed as specified in
, 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
, unknown
attribute lines MUST be ignored.Semantics VerificationAssuming that parsing completes successfully, the parsed
description is then evaluated to ensure internal consistency
as well as proper support for mandatory features.
Specifically, the following checks are performed:
For each "m=" section, valid values for each of the
mandatory-to-use features enumerated in
MUST be present.
These values MAY be either present at the media level or
inherited from the session level.
ICE ufrag and password values, which MUST comply with
the size limits specified in
.
A tls-id value, which MUST be set according to
. If
this is a re-offer or a response to a re-offer and the
tls-id value is different from that presently in use, the
DTLS connection is not being continued and the remote
description MUST be part of an ICE restart, together with
new ufrag and password values.
A DTLS setup value, which MUST be set according to the
rules specified in
and MUST be
consistent with the selected role of the current DTLS
connection, if one exists and is being continued.
DTLS fingerprint values, where at least one
fingerprint MUST be present.
All rid-ids referenced in an "a=simulcast" line MUST
exist as "a=rid" lines.
Each "m=" section is also checked to ensure that prohibited
features are not used.
If the RTP/RTCP multiplexing policy is "require", each
"m=" section MUST contain an "a=rtcp-mux" attribute. If an "m="
section contains an "a=rtcp-mux-only" attribute, that
section MUST also contain an "a=rtcp-mux" attribute.
If an "m=" section was present in the previous answer, the
state of RTP/RTCP multiplexing MUST match what was
previously negotiated.
If this session description is of type "pranswer" or
"answer", the following additional checks are applied:
The session description MUST follow the rules defined in
, including the
requirement that the number of "m=" sections MUST exactly
match the number of "m=" sections in the associated
offer.
For each "m=" section, the media type and protocol values
MUST exactly match the media type and protocol values in
the corresponding "m=" section in the associated offer.
If any of the preceding checks failed, processing MUST
stop and an error MUST be returned.Applying a Local DescriptionThe following steps are performed at the media engine level
to apply a local description. If an error is returned, the
session MUST be restored to the state it was in before
performing these steps.First, "m=" sections are processed. For each "m=" section, the
following steps MUST be performed; if any parameters are out of
bounds or cannot be applied, processing MUST stop and an error
MUST be returned.
If this "m=" section is new, begin gathering candidates for
it, as defined in
, unless it is
definitively being bundled (either (1) this is an offer and the
"m=" section is marked bundle-only or (2) it is an answer and the
"m=" section is bundled into another "m=" section).
Or, if the ICE ufrag and password values have changed,
trigger the ICE agent to start an ICE restart as described in
, and begin
gathering new candidates for the "m=" section. If this
description is an answer, also start checks on that media
section.
If the "m=" section <proto> value indicates use of RTP:
If there is no RtpTransceiver associated with this "m="
section, find one and associate it with this "m=" section
according to the following steps. Note that this situation
will only occur when applying an offer.
Find the RtpTransceiver that corresponds to this "m="
section, using the mapping between transceivers and "m="
section indices established when creating the offer.
Set the value of this RtpTransceiver's mid property to
the MID of the "m=" section.
If RTCP mux is indicated, prepare to demux RTP and RTCP
from the RTP ICE component, as specified in
.
For each specified RTP header extension, establish a
mapping between the extension ID and URI, as described in
.
If the MID header extension is supported, prepare to
demux RTP streams intended for this "m=" section based on the
MID header extension, as described in
.
For each specified media format, establish a mapping
between the payload type and the actual media format, as
described in
. In addition,
prepare to demux RTP streams intended for this "m=" section
based on the media formats supported by this "m=" section, as
described in
.
For each specified "rtx" media format, establish a
mapping between the RTX payload type and its associated
primary payload type, as described in
Sections and of .
If the direction attribute is of type "sendrecv" or
"recvonly", enable receipt and decoding of media.
Finally, if this description is of type "pranswer" or
"answer", follow the processing defined in
below.Applying a Remote DescriptionThe following steps are performed to apply a remote
description. If an error is returned, the session MUST be
restored to the state it was in before performing these
steps.If the answer contains any "a=ice-options" attributes where
"trickle" is listed as an attribute, update the PeerConnection
canTrickleIceCandidates property to be "true". Otherwise, set this property to
"false".The following steps MUST be performed for attributes at the
session level; if any parameters are out of bounds or cannot
be applied, processing MUST stop and an error MUST be returned.
For any specified "CT" bandwidth value, set this value as the
limit for the maximum total bitrate for all "m=" sections, as
specified in
. Within this
overall limit, the implementation can dynamically decide how
to best allocate the available bandwidth between "m=" sections,
respecting any specific limits that have been specified for
individual "m=" sections.
For any specified "RR" or "RS" bandwidth values, handle as
specified in
.
Any "AS" bandwidth value ()
MUST be ignored, as the meaning
of this construct at the session level is not well
defined.
For each "m=" section, the following steps MUST be performed;
if any parameters are out of bounds or cannot be applied,
processing MUST stop and an error MUST be returned.
If the ICE ufrag or password changed from the previous
remote description:
If the description is of type "offer", the
implementation MUST note that an ICE restart is needed, as
described in
.
If the description is of type "answer" or "pranswer",
then check to see if the current local description is an
ICE restart, and if not, generate an error. If the
PeerConnection state is "have-remote-pranswer" and the ICE
ufrag or password changed from the previous provisional
answer, then signal the ICE agent to discard any previous
ICE checklist state for the "m=" section. Finally, signal
the ICE agent to begin checks.
If the current local description indicates an ICE restart
but neither the ICE ufrag nor the password has changed from the
previous remote description (as prescribed by
), generate an
error.
Configure the ICE components associated with this media
section to use the supplied ICE remote ufrag and password for
their connectivity checks.
Pair any supplied ICE candidates with any gathered local
candidates, as described in
, and start
connectivity checks with the appropriate credentials.
If an "a=end-of-candidates" attribute is present, process
the end-of-candidates indication as described in
.
If the "m=" section <proto> value indicates use of RTP:
If the "m=" section is being recycled (see
), disassociate
the currently associated RtpTransceiver by setting its mid
property to "null", and discard the mapping between the
transceiver and its "m=" section index.
If the "m=" section is not associated with any
RtpTransceiver (possibly because it was disassociated in the
previous step), either find an RtpTransceiver or create one
according to the following steps:
If the "m=" section is sendrecv or recvonly, and there
are RtpTransceivers of the same type that were added to
the PeerConnection by addTrack and are not associated
with any "m=" section and are not stopped, find the first
(according to the canonical order described in
) such
RtpTransceiver.
If no RtpTransceiver was found in the previous step,
create one with a recvonly direction.
Associate the found or created RtpTransceiver with the
"m=" section by setting the value of the RtpTransceiver's
mid property to the MID of the "m=" section, and establish
a mapping between the transceiver and the index of the "m="
section. If the "m=" section does not include a MID (i.e.,
the remote endpoint does not support the MID extension),
generate a value for the RtpTransceiver mid property,
following the guidance for "a=mid" mentioned in
.
For each specified media format that is also supported
by the local implementation, establish a mapping between
the specified payload type and the media format, as
described in
. Specifically, this
means that the implementation records the payload type to
be used in outgoing RTP packets when sending each specified
media format, as well as the relative preference for each
format that is indicated in their ordering. If any
indicated media format is not supported by the local
implementation, it MUST be ignored.
For each specified "rtx" media format, establish a
mapping between the RTX payload type and its associated
primary payload type, as described in
. If any referenced
primary payload types are not present, this MUST result in
an error. Note that RTX payload types may refer to primary
payload types that are not supported by the local media
implementation, in which case the RTX payload type MUST
also be ignored.
For each specified fmtp parameter that is supported by
the local implementation, enable them on the associated
media formats.
For each specified Synchronization Source (SSRC) that is signaled in the "m="
section, prepare to demux RTP streams intended for this "m="
section using that SSRC, as described in
.
For each specified RTP header extension that is also
supported by the local implementation, establish a mapping
between the extension ID and URI, as described in
. Specifically, this
means that the implementation records the extension ID to
be used in outgoing RTP packets when sending each specified
header extension. If any indicated RTP header extension is
not supported by the local implementation, it MUST be
ignored.
For each specified RTCP feedback mechanism that is
supported by the local implementation, enable them on the
associated media formats.
For any specified "TIAS" ("Transport
Independent Application Specific Maximum") bandwidth value, set this value
as a constraint on the maximum RTP bitrate to be used when
sending media, as specified in
. If a "TIAS" value is not
present but an "AS" value is specified, generate a "TIAS"
value using this formula:
TIAS = AS * 1000 * 0.95 - (50 * 40 * 8)
The 1000 changes the unit from kbps to bps (as required
by TIAS), and the 0.95 is to allocate 5% to RTCP.
An estimate of header overhead is then subtracted out, in which
the 50 is based on 50 packets per second, the 40 is based on
typical header size (in bytes), and the 8 converts bytes to bits.
Note that "TIAS" is preferred over
"AS" because it provides more accurate
control of bandwidth.
For any "RR" or "RS" bandwidth values, handle as
specified in
.
Any specified "CT" bandwidth value MUST be ignored, as
the meaning of this construct at the media level is not
well defined.
If the "m=" section is of type "audio":
For each specified "CN" media format, configure
silence suppression for all supported media formats with
the same clock rate, as described in
, except for formats
that have their own internal silence suppression
mechanisms. Silence suppression for such formats (e.g.,
Opus) is controlled via fmtp parameters, as discussed in
.
For each specified "telephone-event" media format,
enable dual-tone multifrequency (DTMF) transmission for all supported media formats
with the same clock rate, as described in
. If there are
any supported media formats that do not have a
corresponding telephone-event format, disable DTMF
transmission for those formats.
For any specified "ptime" value, configure the
available media formats to use the specified packet size
when sending. If the specified size is not supported for
a media format, use the next closest value instead.
Finally, if this description is of type "pranswer" or
"answer", follow the processing defined in
below.Applying an AnswerIn addition to the steps mentioned above for processing a
local or remote description, the following steps are performed
when processing a description of type "pranswer" or
"answer".For each "m=" section, the following steps MUST be performed:
If the "m=" section has been rejected (i.e., the <port> value is set to
zero in the answer), stop any reception or transmission of
media for this section, and, unless a non-rejected "m=" section
is bundled with this "m=" section, discard any associated ICE
components, as described in
.
If the remote DTLS fingerprint has been changed or the
value of the "a=tls-id" attribute has changed, tear down the DTLS connection. This
includes the case when the PeerConnection state is
"have-remote-pranswer". If a DTLS connection needs to be torn
down but the answer does not indicate an ICE restart or, in
the case of "have-remote-pranswer", new ICE credentials, an
error MUST be generated. If an ICE restart is performed
without a change in the tls-id value or fingerprint, then the same DTLS
connection is continued over the new ICE channel. Note that
although JSEP requires that answerers change the tls-id value
if and only if the offerer does, non-JSEP answerers are
permitted to change the tls-id value as long as the offer contained
an ICE restart. Thus, JSEP implementations that process DTLS
data prior to receiving an answer MUST be prepared to receive
either a ClientHello or data from the previous DTLS
connection.
If no valid DTLS connection exists, prepare to start a
DTLS connection, using the specified roles and fingerprints,
on any underlying ICE components, once they are active.
If the "m=" section <proto> value indicates use of RTP:
If the "m=" section references RTCP feedback mechanisms
that were not present in the corresponding "m=" section in
the offer, this indicates a negotiation problem and MUST
result in an error. However, new media formats and new RTP
header extension values are permitted in the answer, as
described in
and
.
If the "m=" section has RTCP mux enabled, discard the RTCP
ICE component, if one exists, and begin or continue muxing
RTCP over the RTP ICE component, as specified in
. Otherwise,
prepare to transmit RTCP over the RTCP ICE component; if no
RTCP ICE component exists because RTCP mux was previously
enabled, this MUST result in an error.
If the "m=" section has Reduced-Size RTCP enabled,
configure the RTCP transmission for this "m=" section to use
Reduced-Size RTCP, as specified in
.
If the direction attribute in the answer indicates
that the JSEP implementation should be sending media
("sendonly" for local answers, "recvonly" for remote
answers, or "sendrecv" for either type of answer), choose
the media format to send as the most preferred media format
from the remote description that is also locally supported,
as discussed in Sections and of , and start
transmitting RTP media using that format once the
underlying transport layers have been established. If an
SSRC has not already been chosen for this outgoing RTP
stream, choose a unique random one. If media is already being
transmitted, the same SSRC SHOULD be used unless the
clock rate of the new codec is different, in which case a
new SSRC MUST be chosen, as specified in
.
The payload type mapping from the remote description is
used to determine payload types for the outgoing RTP
streams, including the payload type for the send media
format chosen above. Any RTP header extensions that were
negotiated should be included in the outgoing RTP streams,
using the extension mapping from the remote description. If the MID
header extension has been negotiated, include it in the outgoing RTP
streams, as indicated in
.
If the RtpStreamId or RepairedRtpStreamId header extensions have been
negotiated and rid-ids have been established, include these header
extensions in the outgoing RTP streams, as indicated in
.
If the "m=" section is of type "audio", and silence
suppression was (1) configured for the send media format as a
result of processing the remote description and (2) also
enabled for that format in the local description, use
silence suppression for outgoing media, in accordance with
the guidance in
.
If these
conditions are not met, silence suppression MUST NOT be
used for outgoing media.
If simulcast has been negotiated, send the appropriate number of
Source RTP Streams as specified in
.
If the send media format chosen above has a
corresponding "rtx" media format or a FEC mechanism has
been negotiated, establish a redundancy RTP stream with a
unique random SSRC for each Source RTP Stream, and start or
continue transmitting RTX/FEC packets as needed.
If the send media format chosen above has a
corresponding "red" media format of the same clock rate,
allow redundant encoding using the specified format for
resiliency purposes, as discussed in
. Note
that unlike RTX or FEC media formats, the "red" format is
transmitted on the Source RTP Stream, not the redundancy
RTP stream.
Enable the RTCP feedback mechanisms referenced in the
media section for all Source RTP Streams using the
specified media formats. Specifically, begin or continue
sending the requested feedback types and reacting to
received feedback, as specified in
. When sending RTCP
feedback, follow the rules and recommendations from
to select
which SSRC to use.
If the direction attribute in the answer indicates
that the JSEP implementation should not be sending media
("recvonly" for local answers, "sendonly" for remote
answers, or "inactive" for either type of answer), stop
transmitting all RTP media, but continue sending RTCP, as
described in
.
If the "m=" section <proto> value indicates use of SCTP:
If an SCTP association exists and the remote SCTP port
has changed, discard the existing SCTP association. This
includes the case when the PeerConnection state is
"have-remote-pranswer".
If no valid SCTP association exists, prepare to initiate
an SCTP association over the associated ICE component and
DTLS connection, using the local SCTP port value from the
local description and the remote SCTP port value from the
remote description, as described in
.
If the answer contains valid bundle groups, discard any ICE
components for the "m=" sections that will be bundled onto the
primary ICE components in each bundle, and begin muxing these
"m=" sections accordingly, as described in
.If the description is of type "answer" and there are still
remaining candidates in the ICE candidate pool, discard
them.Processing RTP/RTCPWhen bundling, associating incoming RTP/RTCP with the proper
"m=" section is defined in
. When not bundling, the proper "m=" section is clear from the
ICE component over which the RTP/RTCP is received.Once the proper "m=" section or sections are known, RTP/RTCP is delivered
to the RtpTransceiver(s) associated with the "m=" section(s) and
further processing of the RTP/RTCP is done at the RtpTransceiver
level. This includes using the RID mechanism
and its associated RtpStreamId and
RepairedRtpStreamId identifiers to distinguish between
multiple encoded streams and determine which Source RTP
stream should be repaired by a given redundancy RTP stream.ExamplesNote that this example section shows several SDP fragments. To
accommodate RFC line-length restrictions, some of the SDP lines have been split
into multiple lines, where leading whitespace indicates that a
line is a continuation of the previous line. In addition, some
blank lines have been added to improve readability but are not
valid in SDP.More examples of SDP for WebRTC call flows, including examples
with IPv6 addresses, can be found in
.Simple ExampleThis section shows a very simple example that sets up a
minimal audio/video call between two JSEP endpoints without
using Trickle ICE. The example in the following section
provides a more detailed example of what could happen in a JSEP
session.The code flow below shows Alice's endpoint initiating the
session to Bob's endpoint. The messages from the JavaScript
application in Alice's browser to the JavaScript in Bob's
browser, abbreviated as "AliceJS" and "BobJS", respectively, are
assumed to flow over some signaling protocol via a web server.
The JavaScript on both Alice's side and Bob's side waits for
all candidates before sending the offer or answer, so the
offers and answers are complete; Trickle ICE is not used. The
user agents (JSEP implementations) in Alice's and Bob's browsers,
abbreviated as "AliceUA" and "BobUA", respectively, are both using the
default bundle policy of "balanced" and the default RTCP mux
policy of "require".AliceUA: create new PeerConnection
AliceJS->AliceUA: addTrack with two tracks: audio and video
AliceJS->AliceUA: createOffer to get offer
AliceJS->AliceUA: setLocalDescription with offer
AliceUA->AliceJS: multiple onicecandidate events with candidates
// wait for ICE gathering to complete
AliceUA->AliceJS: onicecandidate event with null candidate
AliceJS->AliceUA: get |offer-A1| from pendingLocalDescription
// |offer-A1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-A1|
WebServer->BobJS: signaling with |offer-A1|
// |offer-A1| arrives at Bob
BobJS->BobUA: create a PeerConnection
BobJS->BobUA: setRemoteDescription with |offer-A1|
BobUA->BobJS: ontrack events for audio and video tracks
// Bob accepts call
BobJS->BobUA: addTrack with local tracks
BobJS->BobUA: createAnswer
BobJS->BobUA: setLocalDescription with answer
BobUA->BobJS: multiple onicecandidate events with candidates
// wait for ICE gathering to complete
BobUA->BobJS: onicecandidate event with null candidate
BobJS->BobUA: get |answer-A1| from currentLocalDescription
// |answer-A1| is sent over signaling protocol
// to Alice
BobJS->WebServer: signaling with |answer-A1|
WebServer->AliceJS: signaling with |answer-A1|
// |answer-A1| arrives at Alice
AliceJS->AliceUA: setRemoteDescription with |answer-A1|
AliceUA->AliceJS: ontrack events for audio and video tracks
// media flows
BobUA->AliceUA: media sent from Bob to Alice
AliceUA->BobUA: media sent from Alice to Bob ]]>The SDP for |offer-A1| looks like:The SDP for |answer-A1| looks like:Detailed ExampleThis section shows a more involved example of a session
between two JSEP endpoints. Trickle ICE is used in full trickle
mode, with a bundle policy of "max-bundle", an RTCP mux policy
of "require", and a single TURN server. Initially, both Alice
and Bob establish an audio channel and a data channel. Later,
Bob adds two video flows -- one for his video feed and one for
screen sharing, both supporting FEC -- with the video feed
configured for simulcast. Alice accepts these video flows but
does not add video flows of her own, so they are handled as
recvonly. Alice also specifies a maximum video decoder
resolution.AliceUA: create new PeerConnection
AliceJS->AliceUA: addTrack with an audio track
AliceJS->AliceUA: createDataChannel to get data channel
AliceJS->AliceUA: createOffer to get |offer-B1|
AliceJS->AliceUA: setLocalDescription with |offer-B1|
// |offer-B1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-B1|
WebServer->BobJS: signaling with |offer-B1|
// |offer-B1| arrives at Bob
BobJS->BobUA: create a PeerConnection
BobJS->BobUA: setRemoteDescription with |offer-B1|
BobUA->BobJS: ontrack event with audio track from Alice
// candidates are sent to Bob
AliceUA->AliceJS: onicecandidate (host) |offer-B1-candidate-1|
AliceJS->WebServer: signaling with |offer-B1-candidate-1|
AliceUA->AliceJS: onicecandidate (srflx) |offer-B1-candidate-2|
AliceJS->WebServer: signaling with |offer-B1-candidate-2|
AliceUA->AliceJS: onicecandidate (relay) |offer-B1-candidate-3|
AliceJS->WebServer: signaling with |offer-B1-candidate-3|
WebServer->BobJS: signaling with |offer-B1-candidate-1|
BobJS->BobUA: addIceCandidate with |offer-B1-candidate-1|
WebServer->BobJS: signaling with |offer-B1-candidate-2|
BobJS->BobUA: addIceCandidate with |offer-B1-candidate-2|
WebServer->BobJS: signaling with |offer-B1-candidate-3|
BobJS->BobUA: addIceCandidate with |offer-B1-candidate-3|
// Bob accepts call
BobJS->BobUA: addTrack with local audio
BobJS->BobUA: createDataChannel to get data channel
BobJS->BobUA: createAnswer to get |answer-B1|
BobJS->BobUA: setLocalDescription with |answer-B1|
// |answer-B1| is sent to Alice
BobJS->WebServer: signaling with |answer-B1|
WebServer->AliceJS: signaling with |answer-B1|
AliceJS->AliceUA: setRemoteDescription with |answer-B1|
AliceUA->AliceJS: ontrack event with audio track from Bob
// candidates are sent to Alice
BobUA->BobJS: onicecandidate (host) |answer-B1-candidate-1|
BobJS->WebServer: signaling with |answer-B1-candidate-1|
BobUA->BobJS: onicecandidate (srflx) |answer-B1-candidate-2|
BobJS->WebServer: signaling with |answer-B1-candidate-2|
BobUA->BobJS: onicecandidate (relay) |answer-B1-candidate-3|
BobJS->WebServer: signaling with |answer-B1-candidate-3|
WebServer->AliceJS: signaling with |answer-B1-candidate-1|
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-1|
WebServer->AliceJS: signaling with |answer-B1-candidate-2|
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-2|
WebServer->AliceJS: signaling with |answer-B1-candidate-3|
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-3|
// data channel opens
BobUA->BobJS: ondatachannel event
AliceUA->AliceJS: ondatachannel event
BobUA->BobJS: onopen
AliceUA->AliceJS: onopen
// media is flowing between endpoints
BobUA->AliceUA: audio+data sent from Bob to Alice
AliceUA->BobUA: audio+data sent from Alice to Bob
// some time later, Bob adds two video streams
// note: no candidates exchanged, because of bundle
BobJS->BobUA: addTrack with first video stream
BobJS->BobUA: addTrack with second video stream
BobJS->BobUA: createOffer to get |offer-B2|
BobJS->BobUA: setLocalDescription with |offer-B2|
// |offer-B2| is sent to Alice
BobJS->WebServer: signaling with |offer-B2|
WebServer->AliceJS: signaling with |offer-B2|
AliceJS->AliceUA: setRemoteDescription with |offer-B2|
AliceUA->AliceJS: ontrack event with first video track
AliceUA->AliceJS: ontrack event with second video track
AliceJS->AliceUA: createAnswer to get |answer-B2|
AliceJS->AliceUA: setLocalDescription with |answer-B2|
// |answer-B2| is sent over signaling protocol
// to Bob
AliceJS->WebServer: signaling with |answer-B2|
WebServer->BobJS: signaling with |answer-B2|
BobJS->BobUA: setRemoteDescription with |answer-B2|
// media is flowing between endpoints
BobUA->AliceUA: audio+video+data sent from Bob to Alice
AliceUA->BobUA: audio+video+data sent from Alice to Bob ]]>The SDP for |offer-B1| looks like:|offer-B1-candidate-1| looks like:|offer-B1-candidate-2| looks like:|offer-B1-candidate-3| looks like:The SDP for |answer-B1| looks like:|answer-B1-candidate-1| looks like:|answer-B1-candidate-2| looks like:|answer-B1-candidate-3| looks like:The SDP for |offer-B2| is shown below. In addition to the
new "m=" sections for video, both of which are offering FEC and
one of which is offering simulcast, note the increment of the
version number in the "o=" line; changes to the "c=" line,
indicating the local candidate that was selected; and the
inclusion of gathered candidates as a=candidate lines.The SDP for |answer-B2| is shown below. In addition to the
acceptance of the video "m=" sections, the use of a=recvonly to
indicate one-way video, and the use of a=imageattr to limit the
received resolution, note the use of setup:passive to maintain
the existing DTLS roles.Early Transport Warmup ExampleThis example demonstrates the early-warmup technique
described in
. Here, Alice's
endpoint sends an offer to Bob's endpoint to start an
audio/video call. Bob immediately responds with an answer that
accepts the audio/video "m=" sections but marks them as sendonly
(from his perspective), meaning that Alice will not yet send
media. This allows the JSEP implementation to start negotiating
ICE and DTLS immediately. Bob's endpoint then prompts him to
answer the call, and when he does, his endpoint sends a second
offer, which enables the audio and video "m=" sections, and
thereby bidirectional media transmission. The advantage of such
a flow is that as soon as the first answer is received, the
implementation can proceed with ICE and DTLS negotiation and
establish the session transport. If the transport setup
completes before the second offer is sent, then media can be
transmitted by the callee immediately upon
answering the call, minimizing perceived post-dial delay. The
second offer/answer exchange can also change the preferred
codecs or other session parameters.This example also makes use of the "relay" ICE candidate
policy described in
to minimize the ICE
gathering and checking needed.AliceUA: create new PeerConnection with "relay" ICE policy
AliceJS->AliceUA: addTrack with two tracks: audio and video
AliceJS->AliceUA: createOffer to get |offer-C1|
AliceJS->AliceUA: setLocalDescription with |offer-C1|
// |offer-C1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-C1|
WebServer->BobJS: signaling with |offer-C1|
// |offer-C1| arrives at Bob
BobJS->BobUA: create new PeerConnection with "relay" ICE policy
BobJS->BobUA: setRemoteDescription with |offer-C1|
BobUA->BobJS: ontrack events for audio and video
// a relay candidate is sent to Bob
AliceUA->AliceJS: onicecandidate (relay) |offer-C1-candidate-1|
AliceJS->WebServer: signaling with |offer-C1-candidate-1|
WebServer->BobJS: signaling with |offer-C1-candidate-1|
BobJS->BobUA: addIceCandidate with |offer-C1-candidate-1|
// Bob prepares an early answer to warm up the
// transport
BobJS->BobUA: addTransceiver with null audio and video tracks
BobJS->BobUA: transceiver.setDirection(sendonly) for both
BobJS->BobUA: createAnswer
BobJS->BobUA: setLocalDescription with answer
// |answer-C1| is sent over signaling protocol
// to Alice
BobJS->WebServer: signaling with |answer-C1|
WebServer->AliceJS: signaling with |answer-C1|
// |answer-C1| (sendonly) arrives at Alice
AliceJS->AliceUA: setRemoteDescription with |answer-C1|
AliceUA->AliceJS: ontrack events for audio and video
// a relay candidate is sent to Alice
BobUA->BobJS: onicecandidate (relay) |answer-B1-candidate-1|
BobJS->WebServer: signaling with |answer-B1-candidate-1|
WebServer->AliceJS: signaling with |answer-B1-candidate-1|
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-1|
// ICE and DTLS establish while call is ringing
// Bob accepts call, starts media, and sends
// new offer
BobJS->BobUA: transceiver.setTrack with audio and video tracks
BobUA->AliceUA: media sent from Bob to Alice
BobJS->BobUA: transceiver.setDirection(sendrecv) for both
transceivers
BobJS->BobUA: createOffer
BobJS->BobUA: setLocalDescription with offer
// |offer-C2| is sent over signaling protocol
// to Alice
BobJS->WebServer: signaling with |offer-C2|
WebServer->AliceJS: signaling with |offer-C2|
// |offer-C2| (sendrecv) arrives at Alice
AliceJS->AliceUA: setRemoteDescription with |offer-C2|
AliceJS->AliceUA: createAnswer
AliceJS->AliceUA: setLocalDescription with |answer-C2|
AliceUA->BobUA: media sent from Alice to Bob
// |answer-C2| is sent over signaling protocol
// to Bob
AliceJS->WebServer: signaling with |answer-C2|
WebServer->BobJS: signaling with |answer-C2|
BobJS->BobUA: setRemoteDescription with |answer-C2| ]]>The SDP for |offer-C1| looks like:|offer-C1-candidate-1| looks like:The SDP for |answer-C1| looks like:|answer-C1-candidate-1| looks like:The SDP for |offer-C2| looks like:The SDP for |answer-C2| looks like:Security ConsiderationsThe IETF has published separate documents
describing the security
architecture for WebRTC as a whole. The remainder of this section
describes security considerations for this document.While formally the JSEP interface is an API, it is better to
think of it as an Internet protocol, with the application
JavaScript being untrustworthy from the perspective of the JSEP
implementation. Thus, the threat model of
applies. In particular, JavaScript can
call the API in any order and with any inputs, including
malicious ones. This is particularly relevant when we consider
the SDP that is passed to setLocalDescription. While correct
API usage requires that the application pass in SDP that was
derived from createOffer or createAnswer, there is no
guarantee that applications do so. The JSEP implementation MUST
be prepared for the JavaScript to pass in bogus data instead.Conversely, the application programmer needs to be aware that
the JavaScript does not have complete control of endpoint
behavior. One case that bears particular mention is that editing
ICE candidates out of the SDP or suppressing trickled candidates
does not have the expected behavior: implementations will still
perform checks from those candidates even if they are not sent to
the other side. Thus, for instance, it is not possible to prevent
the remote peer from learning your public IP address by removing
server-reflexive candidates. Applications that wish to conceal
their public IP address MUST instead configure the ICE agent to
use only relay candidates.IANA ConsiderationsThis document has no IANA actions.ReferencesNormative ReferencesA Session Initiation Protocol (SIP) Usage for Incremental
Provisioning of Candidates for the Interactive Connectivity
Establishment (Trickle ICE)RTP Stream Identifier Source Description (SDES)Trickle ICE: Incremental Provisioning of Candidates for the
Interactive Connectivity Establishment (ICE) ProtocolSession Description Protocol (SDP) Offer/Answer Considerations for
Datagram Transport Layer Security (DTLS) and Transport Layer Security (TLS)Session Description Protocol (SDP) Offer/Answer Procedures for Interactive Connectivity Establishment (ICE)WebRTC MediaStream Identification in the Session Description ProtocolIndicating Exclusive Support of RTP and RTP Control Protocol (RTCP)
Multiplexing Using the Session Description Protocol (SDP)RTP Payload Format RestrictionsSession Description Protocol (SDP) Offer/Answer Procedures for
Stream Control Transmission Protocol (SCTP) over Datagram Transport Layer
Security (DTLS) TransportNegotiating Media Multiplexing Using the Session Description Protocol (SDP)A Framework for Session Description Protocol (SDP)
Attributes When MultiplexingUsing Simulcast in Session Description Protocol (SDP) and RTP
SessionsWebRTC Forward Error Correction RequirementsMedia Transport and Use of RTP in WebRTCSecurity Considerations for WebRTCWebRTC Security ArchitectureInformative ReferencesWebRTC IP Address Handling RequirementsWebRTC 1.0: Real-time Communication Between BrowsersCiscoGoogleMozillaWorld Wide Web Consortium PR PR-webrtc-202012153rd Generation Partnership Project; Technical
Specification Group Services and System Aspects; IP
Multimedia Subsystem (IMS); Multimedia Telephony; Media
handling and interaction (Release 16)3GPPSDP ABNF SyntaxFor the syntax validation performed in
, the following list of ABNF
definitions is used:
SDP ABNF References
Attribute
Reference
ptime
maxptime
rtpmap
recvonly
sendrecv
sendonly
inactive
fmtp
rtcp
setup
fingerprint
rtcp-fb
extmap
mid
group
imageattr
extmap (encrypt option)
candidate
remote-candidates
ice-lite
ice-ufrag
ice-pwd
ice-options
msid
rid
simulcast
tls-id
Acknowledgements, , , and
provided significant text for this
document. , , ,
, , ,
, , , ,
, , , ,
, ,
, , , and all provided valuable feedback on
this document.