Network Working Group R. Raymond
Internet-Draft E. Lagerway
Intended status: Informational Hookflash
Expires: January 07, 2014 I. Baz Castillo
July 06, 2013
approach is a far better solution than using SDP [RFC4566] as a
surface API for interfacing with WebRTC. The document outlines the
issues and pitfalls as well as use cases that are difficult (or
impossible) with SDP with offer / answer [RFC3264], and explains the
Status of This Memo
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This Internet-Draft will expire on January 07, 2014.
Copyright (c) 2013 IETF Trust and the persons identified as the
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Raymond, et al. Expires January 07, 2014 [Page 1]
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 4
2. Issues with a Universal Session Description Format (and Offer
/ Answer) . . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1. Goal of Minimized Requirements . . . . . . . . . . . . . 6
2.2. Offer / Answer State Machine . . . . . . . . . . . . . . 7
2.2.1. Offer / Answer Violations . . . . . . . . . . . . . . 8
2.3. Browser to Browser Format Compatibility Issue . . . . . . 8
2.6. Is SDP allowed to be mangled? . . . . . . . . . . . . . . 10
2.7. SDP errata and bugs compatibility issues . . . . . . . . 11
2.7.1. SDP Bugs Become Enshrined . . . . . . . . . . . . . . 11
2.8. SIP/SDP compatibility worsened . . . . . . . . . . . . . 12
2.9. Increased surface API . . . . . . . . . . . . . . . . . . 12
2.10. Impossible API to implement to achieve browser
compatibility . . . . . . . . . . . . . . . . . . . . . . 13
2.10.1. Example Oddities That Need Definition . . . . . . . 13
2.11. Plan A, Plan B vs NoPlan . . . . . . . . . . . . . . . . 14
2.12. SIP Forking Issue . . . . . . . . . . . . . . . . . . . . 15
3. Alternatives to Fixing these Issues Now . . . . . . . . . . . 15
3.1. Waiting for WebRTC 2.0 . . . . . . . . . . . . . . . . . 15
3.1.1. Cost now to fix versus fixing later . . . . . . . . . 16
3.1.2. If starting over, would even SIP people want SDP as a
surface API? . . . . . . . . . . . . . . . . . . . . 16
3.1.3. Incremental Approach may make Compatibility Worse . . 16
3.2. Session Description Format Construction API . . . . . . . 17
4. Example Difficult Usage Cases with Current Model . . . . . . 19
4.1. On / off hold example usage case . . . . . . . . . . . . 19
4.2. One-Sided Constraints Negotiation use Case Scenario . . . 20
4.3. Meet-me Negotiation Use Case Scenario . . . . . . . . . . 22
4.4. Browser to Browser Compatibility Extension Compatibility
Issue Scenario . . . . . . . . . . . . . . . . . . . . . 22
4.5. Building Interoperability between WebRTC and a SIP
Service Scenario . . . . . . . . . . . . . . . . . . . . 23
4.6. Bit-rate Change Scenario . . . . . . . . . . . . . . . . 24
4.7. Video Codec Option Change Scenario . . . . . . . . . . . 25
4.8. Video Upgrade Scenario . . . . . . . . . . . . . . . . . 25
Raymond, et al. Expires January 07, 2014 [Page 2]
5.1. Overview . . . . . . . . . . . . . . . . . . . . . . . . 26
5.2. Benefits . . . . . . . . . . . . . . . . . . . . . . . . 26
5.2.1. Greater compatibility . . . . . . . . . . . . . . . . 26
5.2.2. Easier to extend . . . . . . . . . . . . . . . . . . 26
5.2.3. Faster Reaction Time To Issues . . . . . . . . . . . 27
5.2.4. Decreased surface API . . . . . . . . . . . . . . . . 27
5.2.5. Greater compatibility for SIP . . . . . . . . . . . . 27
5.2.6. Alternative formats . . . . . . . . . . . . . . . . . 28
5.3. Design Goals and Considerations . . . . . . . . . . . . . 28
5.3.1. Objects Model Kept Simple . . . . . . . . . . . . . . 28
5.3.2. Simple to Gather Negotiation Information . . . . . . 28
5.3.3. Offer / Answer . . . . . . . . . . . . . . . . . . . 28
5.3.4. Extensions . . . . . . . . . . . . . . . . . . . . . 28
5.3.5. Well Defined Behaviors . . . . . . . . . . . . . . . 29
5.3.6. Data Channel . . . . . . . . . . . . . . . . . . . . 29
5.3.7. Satisfy the expectations of the RTCWEB charter . . . 29
5.3.8. SIP/SDP and current WebRTC API shim compatibility
statement . . . . . . . . . . . . . . . . . . . . . . 29
5.3.9. Greater Separation of RTCWEB Working Group and Other
Working Groups . . . . . . . . . . . . . . . . . . . 30
6. Security Considerations . . . . . . . . . . . . . . . . . . . 30
7. References . . . . . . . . . . . . . . . . . . . . . . . . . 30
7.1. Normative References . . . . . . . . . . . . . . . . . . 30
7.2. Informative References . . . . . . . . . . . . . . . . . 31
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 32
While the IETF RTCWEB WG is not specifically tasked with providing an
API by the W3C, the group has effectively defined a surface API with
the mandate to use SDP [RFC4566] with offer / answer [RFC3264].
SDP is a condensed text based format that typically describes all of
the real-time media streams, networking properties, codecs, media
state and media attributes. SDP is completely extensible and can be
used to describe absolutely anything so long as it is formatted
correctly within its minimally defined limitations.
The points for mandating SDP with an offer / answer API typically
boils down to:
1. It's really easy to establish communication, especially with SIP
2. The decision was already made.
3. SDP yields greater compatibility (especially with SIP networks).
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4. We must have some kind of universal exchange format.
5. There is no alternative to this approach except destroying
everything created and starting from scratch.
This document will explain why these reasons are insufficient to
continue with an SDP with offer / answer mandate approach given
strong logical arguments and reasons with real world scenarios where
this approach fails and due in no small part to its lasting
consequences (including negative consequences for SIP).
The document highlights the benefits and goals for a different
charter's requirements, yields greater compatibility and offers a
road-map where future potential extensions can be readily added
without breaking existing implementations.
This Shim will provide the same level of "ease of use" as experienced
not mandatory to use for those who do not require an "SDP with offer
/ answer" model.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
2. Issues with a Universal Session Description Format (and Offer /
The issue with SDP is not the expressiveness of the format but its
usage as an arbitrary universal format and an API surface instead of
with methods, properties and events. Today, in many real-world use
cases, controlling WebRTC requires modifying SDP directly.
requests into a text format (via modifications of SDP existing blobs)
is only one aspect of the many issues the SDP approach creates for
developers. Needlessly, an offer / answer state machine is imposed
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While the currently mandated SDP based API allows developers to
quickly implement basic calling demos and interoperability with some
SIP networks, it has many issues that will be explored and explained
in this document and include (but not limited to):
1. Defining a standard universal all-encompassing session
description format for use with WebRTC that describes all
connections, media, constraints, streams and tracks for all
scenarios is especially challenging.
2. Rather than focusing and defining the properties needed for
communication, the focus is put on the best way to express the
format where every nuance and behavior will need to be detailed
for any browser vendor to capably implement the SDP based WebRTC
3. The bar for browsers (or other applications with WebRTC engines)
to produce a WebRTC engine is raised substantially by forcing
the browser to implement an entire SDP offer / answer engine
too, with little to no added benefit.
4. A universal format built into the browser's API is entirely
unneeded and goes well beyond the RTCWEB chartered mandate for
the RTCWEB Working Group.
5. A flexible and expendable universal exchange format leads to
greater interpretations and mistakes in various implementations,
which in turn leads to increased incompatibilities.
6. Given the format is entirely flexible and open to
interpretation, resulting implementations will more likely be
prone to errors relative to the other truly needed aspects of
RTC (which have better defined boundaries, behaviours, and
7. Mistakes in the format won't be fixed until a new browser binary
update is released and deployed amongst users.
8. Mistakes in implementation of the session description format can
become enshrined and difficult to deprecate (for the sake of
9. Compatibility issues caused by the format will not be limited to
browsers-only as many hybrid browser-engine based applications
now exist too.
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10. Using alternative signaling formats will require complete
understanding of the universal format to be able to translate it
into other alternative signaling formats.
output session description format with 100% precision and
without loss. They will also require pre-knowledge of what each
browser produces and expects, despite the likelihood of a
multitude of outputted flavors, on various platforms, and from
version to version and despite the inability to easily predict
or detect the variants.
need to manipulate any defined universal format rather than
13. Offer / answer is mandated and the state machine is required but
the exact rules and violations of the rules ill defined when
used within WebRTC.
14. The rules of how a universal format can be modified before being
delivered to remote parties need to be meticulously defined or
compatibility issues will arise (including the allowed rules of
post browser format regeneration as to what can be modified and
fed back into the browser).
15. Due to the issues defined above, SIP compatibility will worsen,
An alternative to all of the issues caused by a universal format and
state machine are described later in the document. This alternative
plumbing while providing extensible and modifiable shims written
formats for the specific network where those formats operate.
2.1. Goal of Minimized Requirements
While the primary goal of WebRTC is to enable browser to browser
communication, the definition of a "browser" is ever expanding.
Beyond just traditional hand-held applications, hybrid applications
that are part HTML-5 and part native code exist. Servers will become
as much as part of the WebRTC infrastructure as browsers. Minimizing
the requirements to the basic wire compatibility necessary to achieve
RTC is essential for maximum compatibility, flexibility and varying
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The mandate for the RTCWEB charter is to simply define requirements,
provide basic "on-the-wire" compatibility, and define security
requirements (such as enforcing ICE connection agreements). The
RTCWEB charter goals have been exceeded by going well beyond that
scope by mandating an API that works fine for simple SIP
interoperability demos but does not provide easy compatibility to the
basic constructs needed as outlined from the charter for use with
other on-the-wire signaling protocols (other than SIP). If SIP is
the only end goal of the WG, then that goal must be specifically
stated rather than effectively mandated by making alternative
signaling approaches unreasonably difficult to achieve.
2.2. Offer / Answer State Machine
The current SDP approach requires an offer / answer state machine.
Mandating an offer / answer state machine implies that:
1. SDP be generated by browser A and sent to browser B
2. Browser B must respond with the offer with an answer
3. If either party issues a new offer but the offer is rejected, the
state must revert to the previous agreed SDP (or fail to none)
4. If one side receives an offer while the other side has an
outstanding offer, a conflict occurs and both sides must reject
and revert and perform SDP conflict resolution to issue an offer
5. The only changes to the media that are allowed happens if both
6. Any change required to the SDP requires a network round trip
where both sides mutually agree (at least as traditionally
defined in offer / answer but the rules are in flux)
This offer / answer model is defined as required with the current
implementation. Not only do the browser vendors have to enforce the
signaling. While WebRTC does not dictate the signaling mechanism
between browsers, effectively it is imposing this signaling state
machine on all implementations (which is not a mandate of the RTCWEB
There are other models for signaling other than offer / answer. For
example, one-sided constraints based negotiation is an alternative
model. This type of negotiation requires each side to determine what
it wants to receive independent of the other. This signaling is akin
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to saying "if you plan to send anything, make sure it conforms to the
following". Changes to the media may occur without agreement from
the remote party where each side decides what is acceptable to
receive without agreement from the other. The remote side can decide
if it wants to send within those constraints or not. There is no
round trip offer / answer required in this model to affect change.
Offer / answer introduces the unnecessary asynchronism to the API and
expecting to receive or the current sending codec can be done
immediately without the need for asynchronous calls.
Offer / answer is not required to achieve RTC wire compatibility but
it is currently mandated when alternatives could exist.
2.2.1. Offer / Answer Violations
The offer / answer SDP state machine is already violated in WebRTC.
Trickle ICE precludes offer / answer round trips and other proposed
standards like NoPlan [I-D.ivov-rtcweb-noplan] suggest relaxing the
offer / answer model even more. The rules of what offer / answer at
this point is undefined and in clear violation of the strict previous
rules without clear direction on what exactly constitutes offer /
answer anymore and where it should and should not be used.
A new state for offer / answer called PRANSWER is now defined, which
did not exist as part of the standard offer / answer state machine.
Offer rollback is not adequately defined either should an offer /
answer conflict occur.
Currently, switching codecs requires an SDP offer / answer should
perform a round trip even though it is not technically needed for an
RTC engine to change codecs. Should this be another exception to the
offer / answer state machine?
2.3. Browser to Browser Format Compatibility Issue
SDP is a flexible format, and it allows many alternative methods to
express the same intentions. The smallest change may alter the SDP's
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This creates a parsing and SDP generation compatibility issues. If
then each browser must support every single possible variant of SDP
for every browser version and platform in existence. They must do
this without failure. To maximize compatibility, a browser should
generate the SDP format in the variant expected by the remote party
(despite not having sufficient knowledge about the remote party to
provide the correct SDP).
Since WebRTC is not supposed to mandate the format on the wire for
signaling, one supported use case for WebRTC must be allowing the
browser generated SDP to be converted into alternative on-the-wire
browser, or later by an intermediate gateway. In either case, the
converter must be entirely aware of all variants to the SDP possible
from every browser platform and version, despite browser version
detection being heavily frowned upon by industry best practices.
correct SDP for all browsers and versions before passing the
serialize SDP blob into the browser. Generating compatible SDP may
be impossible unless the exact formats and restrictions are
unquestionably clear by all implementers of the specification (which
is anything but clearly described in the current WebRTC SDP based API
that developers are mandated to use).
The current SDP based API is limited to placing a call and answering
a call and adding media. To perform common edge cases or to utilize
RTC features beyond the basic API typically requires SDP mangling.
from RTC will be through serialization to / from the SDP instead of a
developer must learn an entirely new protocol called "SDP" and be
able to parse and generate not only basic SDP but any SDP extensions
without introducing a single compatibility issue.
streams. The developer must use a widely adopted but hidden feature
to parse the SDP from the browser, change it to add the appropriate
"hold" state, send that hold state to the remote side, wait for the
"answer" to accept the hold, parse the result on the return to see if
the hold was accepted and feed the result to the browser.
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Worse, a flood of extensions to SDP for WebRTC are being written to
"enhance" and "extend" the functionality of the browser with new
features. Many basic things are ill defined in the current SDP based
API, for example, changing non-negotiated codec parameters, such as
currently using or capable of delivering. The developer has no idea
of the extensions available, or what SDP will be produced, or what
handle everything generated by the browser for any use case beyond
basic call, answer and hang-up. This is a heavy burden to place on a
concepts as expressed in SDP, and is a challenge even for those who
Effective APIs are meant to be contracts between a producer and
consumer, whereas this SDP methodology offers little in the form of
any such contract.
developers must learn SDP to use RTC's available features and build
new features. Alternatively, accessors will need to be provided to
not move to an object model straight away and do away with SDP?).
2.6. Is SDP allowed to be mangled?
The choice must be made if SDP may be modified or not. If
modifications are the only way to achieve RTC features available then
what is allowed to be modified must be clearly defined in exact
detail and the expected behavior of each feature (and modification of
each feature), as expressed in SDP, must be defined. Anything short
of exact specifications will cause incompatibility. Again, the
utilize the available RTC features and they must learn the rules of
modification equally well, which virtually do not exist at all today.
If the choice is to not allow complete SDP modification at all, then
the protocol becomes extremely tied to SDP based protocols like SIP.
Yet, there is no mandate for SIP to be the standardized protocol in
WebRTC. In fact, the mandate to require SIP was explicitly denied,
which presents the argument that SDP manipulation must be allowed.
The SDP mangling issue isn't just an issue when the format is sent
on-the-wire. If Browser A sends Browser B an SDP, the current
philosophy is that the SDP is allowed to be modified. However, there
is the possibility of modifying the SDP generated by Browser A and
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giving that modified SDP back to Browser A to change it's behavior
(i.e. a serialized text based API call) before the offer is given to
Browser B (and likewise with Browser B when it responds with its SDP
How much of the SDP is allowed to be modified before giving the SDP
back to the local browser? SDP is a free-form format so anything can
theoretically get changed, but should it be allowed? If not, what
can and cannot be modified? CODECS? SSRC? SDES? Fingerprints?
Transports? M-lines? And so on...
This issue becomes further compounded when extensions are factored in
2.7. SDP errata and bugs compatibility issues
With the SDP baked into the browser binary, the only way SDP
compatibility issues can be fixed is by releasing a new browser
flaws until the browser vendors deliver the fix and the user base
upgrades their browsers.
While it could be argued that any bug must be worked around, SDP is a
unique problem. SDP is a free-form format. Being compatible isn't
as easy as implementing a limited wire protocol for media transport
or a API contract with well defined features and attributes. The
likelihood of free-form SDP containing errors is far greater than a
typical well defined API due to SDPs many flavors, interpretations
and lack of strong definition.
2.7.1. SDP Bugs Become Enshrined
To illustrate a scenario:
1. Browser Vendor A has a bug
2. Browser Vendor B can't work with A because of the bug so it
implements a "work around"
3. Browser Vendor A fixes the bug but implements a work around to be
compatible with Browser Vendor B's "work around"
This situation demonstrates is how browser bugs can become enshrined
as there's no way to update the SDP produced by the browser binary
once it's released until the next update release cycle occurs. This
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to exacting expectations for their network regardless of the browser
The lower level RTC wire protocols that need to be mandated by the
RTCWEB Working Group have limited scopes and well defined behaviors.
Any mistakes are obvious, likely to present very rapidly, and easy to
spot which party is doing something wrong and much easier to fix
earlier as a result. This is not true with a free form highly
descriptive language for sessions. The combinations are limitless
and every scenario is difficult to test, especially in concert with
every other browser vendor with every version released. The session
description will be the likely place of failure across the browsers
when the session description is generated inside the browser's
2.8. SIP/SDP compatibility worsened
One of the main arguments for using SDP with offer / answer was
supposed to be ease of compatibility with existing signaling
networks, like SIP. Instead, variations in the browser's SDP will
likely worsen SIP compatibility instead of enhance it.
A SIP provider must now be compatible with every browser's SDP on
every platform and version and the browser's SDP must be compatible
(Session Border Controller) must be used to re-write any incompatible
SDP to be compatible. However, this moves the problem from the
generation code to run, for maximum compatibility.
2.9. Increased surface API
By mandating SDP, the requirement for compatibility with WebRTC is
increased substantially with little benefit. Instead of just
supporting basic media RTP [RFC3550], STUN/ICE/TURN [RFC5389]/
[RFC5245]/[RFC5766], DTLS [RFC6347] and CODECS an additional bar must
be passed, i.e. a browser or other WebRTC compliant API must support
SDP with a full offer / answer state machine (or a state machine with
additional rules to make it flexible for various scenarios).
With an alternative approach, the entire requirement for SDP could be
removed without any loss of compatibility or increase in complexity
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2.10. Impossible API to implement to achieve browser compatibility
The current mandated SDP based API cannot be implemented as a
standard by independent browser vendors in its current form. A list
of subsequent behaviors regarding the usage, parsing, handling,
extensions, behaviors, constraints and other such reference documents
must be meticulously defined for SDP with the modified offer / answer
state machine or no browser can ever claim to be "compliant". The
current definition process is far from complete.
The current WebRTC SDP based API is far from achieving that goal due
to the inclusion of free-form SDP with offer / answer and it is
grounds for removing it as it goes beyond the RTCWEB's charter and
Any incremental approach that does not remove the offer / answer
model requirement yields a road block to achieving alternative WebRTC
signaling protocols other than SIP.
require an all-encompassing session description and related state
machine is being proposed as an alternative solution so the RTCWEB
charter can complete its defined goals in a timely fashion.
2.10.1. Example Oddities That Need Definition
There are many oddities in the SDP RFC [RFC4566] and the various
For example; will RTP CODEC maps be required or not? They are not
required for basic CODECs according to the SDP RFC. However, with
all the flavors of CODECs being offered, defining a mapping between
payloads is critical to compatibility and not just a good idea.
Another example; should "t=0 0" be respected? Is that allowed to be
changed? Do the browser vendors need to enforce the attribute, or
start until the NTP time stamp and close when the NTP time completes?
These are just small samples of questions that must all be completely
addressed in detail. This could also cause a cascade of updated
reference drafts and confusion as to which version is to be adhered
by browsers as well as what each browser specifically supports.
Nominally referencing the SDP RFC will not be sufficient, and deltas
from the established standards when violated will need to be defined
when the rules change.
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2.11. Plan A, Plan B vs NoPlan
At the time of authoring this document, three plans on how to handle
large number of media streams in SDP have emerged currently under
consideration from the IETF, referred to as PlanA
[I-D.roach-rtcweb-plan-a], PlanB [I-D.uberti-rtcweb-plan] and NoPlan
PlanA and PlanB acknowledge that using SDP as it is historically
defined in SIP is inefficient and problematic for large number of
media streams, especially factoring in that each media line must have
its own unique ports.
way and goes a long way towards improving the situation from SDP by
taking out the mapping of the streams from the SDP but does not
remove the reliance upon SDP. This creates a dual format system
where some information is initially carried over SDP and other
information is signaled through an alternative approach (including
the possibility of SDP offer/answer). NoPlan could have been the
sufficient approach if it took one step further and removed SDP
PlanA, PlanB and NoPlan are a perfect example of why not to use SDP
as the basis for WebRTC. SDP has some arbitrary limitations as a
description protocol for multiple streams whereas no such limitations
exist at the lower layer transports themselves. RTP allows for
multiplexing multiple SSRCs. In other words, the problem is SDP, not
the real time transportation technologies.
These drafts illustrate the limitations of SDP and attempt to solve
it by introducing even more complex descriptions around SDP and / or
by "relaxation" of the offer answer model combined with altering the
description language of SDP.
None of these drafts address most of the concerns outlined in this
draft. If anything, they further illustrate how divergent the SDP
will become as more and more effort is put into working around
problems inherent to the nature of utilizing SDP (or any universal
The issue that SDP implementers face should be isolated to those who
require SDP for their signaling protocols (namely SIP) where they can
choose the best practices for their networks for interoperability.
These complex approaches do not have to be forced on other signaling
protocols that do not have or require such limitations.
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by such limitations by introducing SDP (or any universal format) into
the mix when it adds zero value and fails in its primary objectives,
namely: interoperability with existing SIP vendors & networks.
This further illustrates why SDP baked into the browser binary is not
beneficial for SIP vendors either. They will be forced to upgrade
their SIP infrastructure to support SDP packets from browsers with
re-write of SDP approach to "fix" these incompatibilities.
With an object approach, newer signaling protocols could describe
multiple media streams with ease and SIP providers could ensure they
only generate compatible SDP with their networks and agree on their
best practices and launch new features that incorporate approaches
like as PlanA, PlanB or NoPlan in a manner they deem fit rather then
when the browser vendors decide to upgrade the SDP arbitrarily.
2.12. SIP Forking Issue
The current SDP based API model does not allow for SIP parallel
forking even though the RTC engine can allow for demuxing a media
stream. The current model does not allow for one offer to be
transmitted but accepts multiple answers, which is legal in SIP. A
complex UPDATE process is described on how to work around the problem
instead of fixing the original problem, i.e. the state machine being
but does not care if an upper shim supports SDP / SIP style forking
in the negotiation or not, so long as the basic rules of the RTC
media engine is respected.
3. Alternatives to Fixing these Issues Now
3.1. Waiting for WebRTC 2.0
If we don't get WebRTC 1.0 correct, fixing the API in WebRTC 2.0 may
become even more difficult.
At this stage, prototypes are underway but to our knowledge there are
no major commercial services deployed by more that one major vendor
using the current WebRTC API. Yet, the argument to even consider an
alternative is that 'it's too late'. Imagine trying to argue fixing
it after major networks are reliant upon specific browser
implementation. Having a good but simple architecture from the start
could alleviate a lot of pressure to fix a broken 1.0 in a 2.0
release before APIs become entrenched.
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3.1.1. Cost now to fix versus fixing later
The cost of fixing the API issues today may pale in comparison to the
cost of compatibility problems spread across entire sets of
industries where constant fixes and work around may be required.
3.1.2. If starting over, would even SIP people want SDP as a surface
Even SIP providers and vendors have started to realize that baking
SDP into the browser is not necessarily in their best interests, but
they do have an interest in a simple API to use since they aren't
API shim they desire and achieves greater interoperability because of
predictable, controllable and tailored SDP for their network, would
they not prefer such a model over the current "baked in the browser"
If the current WebRTC specification was ever rebooted, the current
mandated SDP based API would undoubtedly be scrapped in favor of a
better approach without its inherent design and use case flaws with
negative long term compatibility consequences.
3.1.3. Incremental Approach may make Compatibility Worse
One argument put forward, to keep the current SDP model, proposes the
current WebRTC SDP-based API must be completed soon and an
incremental improvement approach can be used to gradually move away
from these obvious problems.
The trouble with an incremental approach is that it may increase
incompatibility further. Not all browser vendors will match the
incremental improvements in unison nor will all customers upgrade
support multiple versions of the WebRTC API and increase the number
still perform all the workarounds required for the current API even
if they support the increments. This limits their willingness to use
any additional APIs until all browsers universally support the
incremental improvements. This will likely slow innovation and
adoption of future improvements.
This will likely create a situation where browser vendors cannot
easily achieve compliance because they too must support the existing
API and incremental improvements along the way, or break those
reliant upon the current methods.
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Having a good solid simple foundation is key to ensuring basic
compatibility while allowing for innovation to occur for those
developers who are willing to give new APIs a trial without needing
to support multiple sets of equivalent but incompatible APIs
3.2. Session Description Format Construction API
other than the model advocated in this draft. That model creates a
session description format rather than allowing direct control of how
While using SDP as the chosen format for WebRTC highlights the issues
described in this draft particularly well, using an alternative
format like JSON instead of SDP does not remove many of the issues
presented in this draft. The issues expressed are not solely caused
by the lack of expressiveness of the SDP format but the nature of
creating a universal all-encompassing format to describe all
transport, media, constraints, and negotiations with an attached
inflexible state machine is the nature of the issue. This format
must do everything and encompass all concepts and becomes the
effective mandate for signaling even if not explicitly required to
A few years ago there was an attempt to create a new "SDP 2.0" format
with a draft named Session Description and Capability Negotiation
[I-D.ietf-mmusic-sdpng]. This effort to create the "ultimate" SDP
format in XML was ultimately abandoned, in no small part because of
the difficulties in coming up with a single solution that works for
Given the difficulty in creating a universal all-encompassing format
based API that constructs a similar flexible, but well defined
highly suspect to fail equally. The reality is that such an effort
Even if successful, this format is not necessarily the format that
will be sent on-the-wire, especially for existing alternative
signaling protocols. As such, the format will still need to be
If the format must be parsed or interpreted by an intermediate then
the format becomes an interaction point to the browser no matter how
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implementation. Whatever format is selected, each browser or
alternative protocol format will have to decide how to convert and
interpret the output and generate new compatible inputs and deal with
the variations that will undoubtedly arrive from browser to browser
and from version to version.
construction or interpretation of a defined format, this format would
still become a do-everything serialization access point for the
browser and the defined exchange point for the local and remote
browser. Therefore the format itself must be described in meticulous
The standardization requirements for such an approach would increase
formatting rules and state machine it relies upon needs to become
standardized in detail as well.
Every combination of this all-encompassing format would need to be
outlined, rather than minimal definition of fixed properties needed
other browsers to break their implementations. Obtaining 100%
stability in such an output equally across all browsers, on all
platforms with all versions is highly doubtful.
the time of writing this draft, any proposal will need to be vetted
to see if it addresses all the concerns and issues brought up in this
still puts the emphasis in driving the developer towards building up
a media signaling exchange format rather than in the logic of how the
media should be controlled and pipelined.
alternative. In a follow-up to this draft the model will describe
pipelining for the browser's media/RTC engine and thus free the
whatever mechanism desired. A simplified shim implemented entirely
independently of a browser's binary release. Should any changes be
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is strictly under the control of the service provider and not the
4. Example Difficult Usage Cases with Current Model
4.1. On / off hold example usage case
This is a typical scenario widely adopted SIP technique of an SDP
attribute to place a stream on / off hold. This is the accepted
methodology and performing alternative approaches would deviate from
the expected practices for use with SIP and its manipulation of SDP.
Although not officially documented as supported, it is effectively
supported in WebRTC implementations. This is a typical use case need
by media application:
1. Browser A establishes a connection with Browser B
2. Browser A and browser B are streaming media
1. createOffer to obtain the SDP from Browser A
2. Parse the SDP
3. Add "a=sendonly" or "a=inactive" to all media
4. Regenerate the SDP, feed back to browser
5. Send the SDP to Browser B
6. Receive the answer from Browser B (which should respond with
a=recvonly if it still wishes media)
7. Parse the received SDP and modify with "a=recvonly" if it did not
respond correctly (to ensure the local side hold back its media)
8. Pass the modified SDP answer back into Browser A
This also implies that:
1. All future SDP events received from Browser B must be mangled to
ensure the "sendonly/recvonly/inactive" attribute is maintained
while on hold
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2. All future createOffer/createAnswer calls from Browser A must be
modified to ensure the "sendonly" property is maintained
3. We need to handle alternative formats to describe hold, e.g.
"c=0.0.0.0" from Browser B which may not utilize the latest SDP
specifications depending on the remote device / platform
Ironically, hold is a very SIP and telephony specific concept. The
better approach would be to allow the streams to be pause/unpaused at
will as that does not require interaction with the SDP, and allow the
higher layers to signal the desire to pause the session to the remote
peer in whatever manner desired.
This is a very basic use case that is extremely complex for a
particular action which is effectively supported by the browsers,
except only via the "SDP surface API". Even if this particular use
by the browser, there are countless other scenarios where tweaking a
field to modify the behavior in the format will only be only
available via SDP manipulation.
4.2. One-Sided Constraints Negotiation use Case Scenario
As WebRTC is a web API and not a SIP API, the API must be capable of
allowing for alternative signaling methods without enforcing it's own
signaling aspects (other than basic principles like ensure ICE
agreement has been achieved for security reasons).
Consider the following scenario:
1. Browser A and Browser B establish a connection
2. Browser A and Browser B use one-sided constraints negotiation
where each party independently decides what "it expects to
3. Browser A decides that it wishes to alter the properties of the
video it expects to receive
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With this model, browser A must be capable of independently modifying
its expectations without waiting for an answer from the remote side
(as that's illegal by the nature of the offer / answer signaling),
unless the rules are relaxed and special exceptions are made. For
the model to work, browser A's receive constraints must be applied to
the send constraints of the remote peer. This model does not require
an SDP offer / answer exchange since the sending peer can monitor the
expectations of the receiving peer and set its send constraints as
To achieve this a for one-sided negotiation:
extracting out last known expectations from the remote SDP last
received as part of the answer
Browser A thus it initiates a fake offer from the remote party
(generating the intentions of the constraint and generating an
have changed, and if so, it may trigger another reverse situation
where step 1 is repeated, except with Browser A and B's role
Is this really doable? Maybe, with a great deal of difficulty and
SDP mangling but it is unquestionably a hack and a violation of offer
/ answer (and relaxed rules create exceptions and exceptions require
additional logic to handle). The offer / answer rules are violated
because no round trip was performed at the time when the constraints
This is also fragile because if Browser B failed to accept the fake
offer there is no way to enforce the constraint nor can the
machine in Browser A expected an offer to be generated before a new
offer would be accepted, the conflict resolution process would be
extremely difficult and messy.
This offer / answer state machine is not even required to fulfill the
mandate of the RTCWEB Working Group charter but it is currently
mandated because it supposedly makes producing "SIP interoperability"
easier (which is highly suspect at best).
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without offer / answer could achieve the same (or better) "SIP
interoperability" without breaking other stateless negotiation
models, such as one-sided negotiation.
4.3. Meet-me Negotiation Use Case Scenario
1. WebRTC client A generates an offer and sends to a server
2. WebRTC client B generates an offer and sends to a server
3. WebRTC client C generates an offer and sends to a server
4. The server returns all the exchanges to each of these clients
5. WebRTC client A, B and C interconnect
Technically, there is no need for independent SDP offer / answer
negotiation amongst all these peers to achieve a mesh scenario for
this use case. Each client has enough information about the other
clients to establish a peer connection. The current WebRTC SDP API
imposes independent round trip negotiations that are not technically
necessary. If WebRTC client D was added later, the original
connection can be forked and re-use the same DTLS fingerprints to
negotiate new encryptions keys for media or data. Fingerprint or
identity signature reuse should not introduce any additional security
concerns since identities will be verified and keys negotiated for
each peer-to-peer connection.
scenario without independent round trip negotiations for each WebRTC
client in the mesh.
4.4. Browser to Browser Compatibility Extension Compatibility Issue
Consider the following scenario:
1. Browser A has implemented an extension to SDP (which is allowed)
2. Browser B has no knowledge of such an extension
sends it to Browser B
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Under this scenario, what should browser B do? To reject the offer
means communication cannot occur. To accept the offer has ambiguous
meaning because the answer might have misunderstood the extension's
intention and does not allow for the appropriate behavior.
The exact rules of what is allowed in SDP and what is not and how
extensions are treated must be defined clearly and non ambiguously.
Even though current SDP offer / answer API can deal with some
extensions, like new codecs being introduced, it is ambiguous on how
to deal with more major extensions such as new SDP profiles,
transports, or encryption methods.
Assuming that a lack of response to an extension is non-agreement to
use the extension is not acceptable. For example, if the extension
was security related dictating some security precondition to opening
a stream, the offer must be rejected as the precondition cannot be
met. Ignoring the extension would mean the offer was accepted where
it cannot be accepted. Another example would be introduction of new
SDP profile, like AVPF2. Offer/answer negotiation simply fails when
it encounters an unknown profile even if it is backwards compatible,
like for instance, most of the calls to current SIP devices will fail
if AVPF is used instead of AVP. A better approach is to define the
rules for how extensions can be made, whereas SDP has no such rules.
Currently, in SIP networks, such extensions are agreed upon in
advanced and extensively tested before they are introduced. SBCs
(Session Border Controllers) are often used to make devices with
control over the format generated on the wire, feature roll out is
under strict control of the provider, and not whenever a browser
vendor decides to produce an update.
4.5. Building Interoperability between WebRTC and a SIP Service
Consider the following scenario:
1. Developer takes SDP produced by browser and send to SIP gateway
(which is supposed to be SIP "compatible")
2. Users happily use this service
3. Browser Vendor A updates the browser SDP generator and a slight
variation in SDP changes
4. Users are now broken
5. SIP gateway must be updated to handle new SDP (and old SDP)
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6. Browser Vendor B updates their browser SDP generator (with a
different SDP variation)
7. Users are now broken again
8. SIP gateway must be updated to handle another variation of SDP
(and maintain the old variations)
9. Repeat to step 3, but add Browser Vendor C, D and multiple
This is not an unrealistic scenario by any stretch of the
imagination. This currently happens in the SIP world, but at least
in that world new devices are tested to ensure compatibility before
roll outs occur on the network so issues can be addressed before the
user's experience is broken. Since the SIP provider and gateway
vendor do not have control over the update cycle of the browsers,
their users are much more prone to breakage by taking the SDP from
the browser and sending to their network.
1. Developer uses shim to generate SDP by browser and sends to SIP
gateway (with SDP that is compatible)
2. Users happily use this service
3. Browser Vendor A updates the browser with a new RTC feature.
4. Repeat to step 2
The reason why the browser update does not affect the gateway is
to the browser do not change the SDP generation logic. The SDP is
entirely in control of SIP network provider. Any bugs with SDP
compatibility can be addressed by the SIP provider without changes in
the browser's binary. Bugs, updates and improvements are completely
within the boundary and control of the SIP network provider.
4.6. Bit-rate Change Scenario
Consider the follow scenario:
1. User is connected to a conference server
2. While user is listening, the user transmits a low bit-rate
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3. The users starts to communicate and the bit-rate is adjusted to
Using the current WebRTC API, this would require an offer / answer
round trip to perform the change and thus the quality would be
updated until the answer was acknowledged, although proposals have
been made to alter the rules for offer / answer in this case and
allow for an exception. This round trip is unnecessary technically
since the bit-rate can be dynamically adjusted without remote
acknowledgment. Yet, the current offer / answer model imposes a
round trip (unless yet another exception to the SDP rules are
4.7. Video Codec Option Change Scenario
Consider the follow scenario:
Using the current WebRTC API, this would require parsing the entire
SDP, isolating the video codecs for a particular video media line,
figuring the mapping and then reconstructing the original SDP with
the newly incorporated changes. Accessors have been suggested for
these common use cases but do not exist yet. If such accessors are
created then a more involved API cannot be avoided out of necessity.
One of the main justifications given by SDP proponents for only
having an API that creates and accepts SDP is due to its supposed
simplicity, as opposed to providing a more involved API.
4.8. Video Upgrade Scenario
1. Alice and Bob are having an audio conversation
2. Alice presses the video button on her application and offers Bob
3. Bob does not wish to see Alice's video, so the application
rejects the media (e.g. using "a=inactive" or "m=video 0")
4. Alice's web application successfully parses and interprets Bob's
5. As Alice's video window of herself is independent of the SDP
negotiation, Alice's HTML5 application successfully renders
Alice's video locally
The current WebRTC implementation offers no event to indicate the
rejection, thus Alice is given no feedback of the rejection. She
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incorrectly assumes she's in a video conversation. In order to solve
this scenario, custom signaling must be added to indicate of Bob's
rejection of Alice's video. Yet this is duplication of signaling as
developer with a choice: either parse the SDP, understand the SDP and
derive meaning, or duplicate the SDP efforts by introducing custom
signaling for a common scenario when upgrading from audio to video
and providing appropriate user feedback.
The browser can expose simple object methods, properties and events
representing the various RTC components at an abstracted level and
provide a solid API for controlling how the media should be
pipelined. The properties needed to be exchanged is separated into
the appropriate object rather than meshed into an all-encompassing
provide easy SDP offer / answer capability for those who want a
similar "simple" API to the current WebRTC API for use with SIP. A
developer can chose to use this shim or not if they do not need SDP.
Likewise, the object model could be used to produce alternative
formats to SDP if the same do-everything format is needed but in an
alternative on-the-wire session description format.
The object model described in the solution is presented in a related
draft. This solution will allow for the RTCWEB Working Group to
complete its chartered mandate without starting from scratch. If
adopted, all of the drafts proposed to solve issues in expressing SDP
for WebRTC can be moved to more appropriate working groups. For
example, SDP for SIP issues can be moved to the appropriate SIP
working groups and multi-party SDP to the MMUSIC (e.g. drafts like
PlanA or PlanB).
5.2.1. Greater compatibility
outputs, properties and events can be well defined on individual
objects and each object will be designed to be a specific contract
5.2.2. Easier to extend
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New objects and methods can be added without breaking existing
compatibility. Compliance can be verified with unit tests able to
test each and every behavior across all browsers' versions on every
object contract to remain fixed to expected behaviors and not break
(unless through well planned deprecation).
current version of the API regardless of any extensions, unless
explicitly deprecated. This is unlike SDP where extensions could be
silently added into the SDP produced by the browsers at will, even in
minor browser version changes, where any component that consumes the
SDP may be unaware what those additional feature behaviors imply or
require as a result.
5.2.3. Faster Reaction Time To Issues
fixed and updated at any time regardless of the browser's release
cycle. If a SIP provider discovers their SIP is not compatible
code to their own needs dynamically without lobbying the browser
vendor and waiting for the browser to be patched and updated.
5.2.4. Decreased surface API
surface API is fixed to the agreed contract. Once agreed, a browser
vendor only has to ensure their compatibility with well defined
limited scope unit tests, and need not worry about some free-form
format that may introduce untold compatibility issues should another
vendor issue an update. This is also true of any non-browsers that
may wish to implement and be compliant to the WebRTC API for
5.2.5. Greater compatibility for SIP
While SIP is not the main RTCWEB Working Group charter responsibility
for WebRTC, SIP compatibility is highly desirable. By exclusively
identical across all platforms and all devices with every browser
version and entirely under the control of the SIP provider. This
increases compatibility for SIP providers. The SDP produced from the
shim can be custom tailored to a SIP network without affecting any
other SIP vendor or harming compatibility with other utilizing
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5.2.6. Alternative formats
information going over the wire can be transformed from the
under control of the service provider and identical regardless of the
consistent and controllable, thus ensuring maximum compatibility
within a network.
The party receiving this format can be sure the format is to an
exacting specification of their choosing rather than relying on
whatever format is produced by whatever browser vendor.
5.3. Design Goals and Considerations
5.3.1. Objects Model Kept Simple
of RTC other than understanding how to plumb the objects together.
Those who need extended properties or events for finer control can
obtain them with simple method access to an object, but those
extended attributes should not be required for simple use cases.
5.3.2. Simple to Gather Negotiation Information
The objects model should allow a simple method for collecting
information that will be needed for various alternative negotiation
models, highly focused to the object. One of the targets for
negotiation must be SDP and SIP.
5.3.3. Offer / Answer
answer state machine but must not preclude this state machine being
built in a layer above. The offer / answer state machine must be
built-in browser services needing to be implemented.
Extending the object model for the expected common extension use
possible extension use cases should include items like local mixing
and data synchronization, or extended properties, events or features.
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As any design, there may be limitations but the design should hold up
to various realistic scenarios that are likely to happen in the near
5.3.5. Well Defined Behaviors
An API must describe specific API behavior sets to the browser
vendors so they have the appropriate guidelines for implementation,
including the mapping to on-the-wire to RTC protocols. The API
presented in the related draft may be the input to a W3C efforts to
define specific and exact expected behavior sets for an object based
5.3.6. Data Channel
5.3.7. Satisfy the expectations of the RTCWEB charter
The object model must adhere to the expectations of the RTCWEB
charter either directly, via extensions that can be defined by the
shim written to utilize the functionality of the object model but it
must not preclude the RTCWEB charter from fulfilling its previously
5.3.8. SIP/SDP and current WebRTC API shim compatibility statement
provides a simple mechanism for parsing and generating SDP for basic
compatibility with SIP networks (capable of supporting the WebRTC
The goal of this object based model is not to provide working
WebRTC API as a shim, including all behaviors, features, bugs and
expectations since the definition of the current approach is not
defined enough to be able to produce that level of compatibility.
This would be an impossible goal as a result, and would add little
possible for others to fork and modify the shim to their own needs
specific to their own SIP/SDP network infrastructure.
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Compatibility with the SDP used in all SIP networks is not a stated
on a common agreed definitive standard set of RFCs and drafts.
5.3.9. Greater Separation of RTCWEB Working Group and Other Working
A JavasScript object model would remove much of the need for cross
IETF working group coordination, which has become common place with
the current movement because of utilizing SDP and its close ties to
SIP. By limiting the RTCWEB technologies used to only those required
for Real-Time Communication from the browser (e.g. RTP, ICE/STUN/
TURN, DTLS), the RTCWEB Working Group is freed from tight couplings
with other IETF working groups, each having their own charters,
schedules, agendas and interests and thus ensures more rapid progress
between RTCWEB Working Group the W3C and developers who are to use
6. Security Considerations
While RTCWEB has it's own security considerations for protocols, a
those already established for use within RTCWEB, e.g. ICE
connectivity permission check or DTLS fingerprint checks.
specific security considerations to their defined model and API,
should any exist.
as needed and warranted.
7.1. Normative References
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264, June
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
Raymond, et al. Expires January 07, 2014 [Page 30]
7.2. Informative References
Kutscher, D., Ott, J., and C. Bormann, "Session
Description and Capability Negotiation", draft-ietf-
mmusic-sdpng-08 (work in progress), February 2005.
Ivov, E., Marocco, E., and P. Thatcher, "No Plan:
Economical Use of the Offer/Answer Model in WebRTC
Sessions with Multiple Media Sources", draft-ivov-rtcweb-
noplan-01 (work in progress), June 2013.
Roach, A. and M. Thomson, "Using SDP with Large Numbers of
Media Flows", draft-roach-rtcweb-plan-a-00 (work in
progress), May 2013.
Uberti, J., "Plan B: a proposal for signaling multiple
media sources in WebRTC.", draft-uberti-rtcweb-plan-00
(work in progress), May 2013.
Burnett, D., "Media Capture and Streams", May 2013, <http:
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, April
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
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[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, January 2012.
Bergkvist, A., "WebRTC 1.0 Real-time Communication Between
Browsers", August 2012,
436, 3553 31 St. NW
Calgary, Alberta T2L 2K7
436, 3553 31 St. NW
Calgary, Alberta T2L 2K7
Inaki Baz Castillo
4905 Del Ray Ave Suite 300
Bethesda, MD 20814
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