> But you're saying that I have no way of knowing up front if it's one or
the other.
I think there is, but it is not pretty - calling
peerConnection.createOffer() will generate that SDP message and you can
parse it a bit to see if the data channel (m=application) shows up first -
that means the boolean worked.
Of course, parsing the SDP is a pretty big hammer just for knowing whether
the boolean is supported...
Looks like new RTCPeerConnection({alwaysNegotiateDataChannels: true}) does
not fail to create a peer connection at the moment, so unsupported
constraints cannot easily be feature-tested.
Perhaps using a getter rather than a simple true ({get
alwaysNegotiateDataChannels()
{ isConstraintSupported = true; return true; }}) would also work.


☆*PhistucK*


On Wed, Feb 4, 2026 at 4:20 AM Yoav Weiss (@Shopify) <[email protected]>
wrote:

> A potentially silly question, due to my complete ignorance of WebRTC
> protocol negotiations:
> If I'm planning to use this new boolean, that means that I may or may not
> want to negotiate a data channel.
> If the other side doesn't support the boolean, I need to create a dummy
> data channel. If it does, I don't need to create one.
> But you're saying that I have no way of knowing up front if it's one or
> the other.
>
> So how would developers know if they need to create a dummy channel? Is
> there some response to this boolean that indicates support or lack-thereof?
> If so, presumably it's not too late to create the dummy channel at that
> point?
>
>
>
> On Tuesday, February 3, 2026 at 1:57:19 PM UTC+1 [email protected]
> wrote:
>
>> Alex,
>>
>> this boils down to WebRTCs notion that the "SDP" is a " blob".  Which
>> makes the API look easy but from the amount of fighting over the plan-b
>> deprecation, "sdp munging" or even the question when inbound-rtp stats
>> appear, this is causing trouble since this "blob" controls a ton of
>> behavior.
>>
>> Developers can parse the SDP, e.g. like
>>   https://jsfiddle.net/fippo/kct3dy46/
>> which with the new option set should return ["application", "video"]
>> instead of ["video", "application"].
>>
>> The opt-in boolean avoids changing the default behavior so no surprises
>> unless you use it.
>> If you opt in you also (implicitly) agree that you are ready to deal with
>> the feature not being supported.
>> So the other end that is parsing and processing the SDP you generate is
>> today receiving
>>   audio-video-data
>> without the opt-in and you ensure it works with
>>   data-audio-video
>> but you will still need to support the old way.
>> I am not worried about browsers where we deal with only two SDP parsers
>> and both are well written robust enough to deal with.
>>
>> cheers
>>
>> Philipp
>>
>>
>> Am Mo., 2. Feb. 2026 um 21:04 Uhr schrieb Alex Russell <
>> [email protected]>:
>>
>>> It's a little concerning that there's no explainer and no (reasonable?)
>>> example code. How can developers debug that they're getting the intended
>>> behaviour? Will they be surprised by the change(s) and/or across browsers?
>>>
>>> Best,
>>>
>>> Alex
>>>
>>> On Friday, January 30, 2026 at 2:16:03 AM UTC-8
>>> [email protected] wrote:
>>>
>>>> Am Di., 27. Jan. 2026 um 09:27 Uhr schrieb Henrik Boström <
>>>> [email protected]>:
>>>>
>>>>> This feature is only needed by the endpoint that creates the initial
>>>>> offer and decides the m= section order, i.e. the other endpoint - whether
>>>>> they support this feature or not - will accept the m= section order that 
>>>>> is
>>>>> on offer. Once the order is decided, it is "locked in" by both endpoints,
>>>>> as per old SDP rules. So it should be cross-browser compatible and
>>>>> backwards-compatible, here's me setting SDP with m= application line first
>>>>> and the offer is accepted: https://jsfiddle.net/henbos/oweu42yz/
>>>>>
>>>>> As for if the browser that is creating this offer supports the
>>>>> feature, you can feature detect by testing it on a throwaway
>>>>> RTCPeerConnection object that you immediately close afterwards.
>>>>>
>>>>
>>>> Throwaway peerconnections are an antipattern too ;-)
>>>> Since RTCConfiguration is not exposed on window one can not feature
>>>> detect upfront but one can call pc.getConfiguration() to see if this had an
>>>> effect after creating the connection and then adapt if necessary (i.e.
>>>> create the not-quite-throwaway channel)
>>>> Not sure I would bother with feature detection, the other side should
>>>> support the resulting SDP both with the new behavior of datachannels being
>>>> first or the current behavior so opt-in should be fine with fallback to the
>>>> current status quo.
>>>>
>>>>
>>>>> On Monday, January 26, 2026 at 4:46:05 PM UTC+1 Robert Flack wrote:
>>>>>
>>>>>> On Mon, Jan 26, 2026 at 10:25 AM Robert Flack <[email protected]>
>>>>>> wrote:
>>>>>>
>>>>>>> Sorry, quick followup, keeping always in the name would make sense
>>>>>>> if it was explained that because this is negotiated first it ensures 
>>>>>>> that
>>>>>>> we at least will always negotiate a data channel even if audio and video
>>>>>>> fail.
>>>>>>>
>>>>>>> On Mon, Jan 26, 2026 at 10:23 AM Robert Flack <[email protected]>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Thank you for the explanation. I definitely agree that this sounds
>>>>>>>> very important. My concern remains that the parameter name and
>>>>>>>> developer documentation does not convey this and without this context 
>>>>>>>> it's
>>>>>>>> unclear why I should use this. As an example, the config parameter 
>>>>>>>> could
>>>>>>>> just be called negotiateDataChannels, as a cue that this will ensure 
>>>>>>>> your
>>>>>>>> connection is configured to support creating data channels. It would be
>>>>>>>> worth also explaining why you want to do this, with a short snippet 
>>>>>>>> showing
>>>>>>>> how a data channel can be added later.
>>>>>>>>
>>>>>>>
>>>> Negotiating datachannels is already an overloaded term - the SDP
>>>> negotiates support and datachannels themselves are negotiated in-band but
>>>> can also be negotiated out-of-band.
>>>>
>>>>
>>>>>>>> The spec says that it is negotiated before audio or video but
>>>>>>>> doesn't explain why you might want to do this - i.e. to ensure that 
>>>>>>>> you can
>>>>>>>> negotiate a connection even if the audio or video channels fail to
>>>>>>>> negotiate a compatible codec.
>>>>>>>>
>>>>>>>> TLDR; I'm not opposed to this feature - after learning what it does
>>>>>>>> and why, I agree we need it. I just think it would be very hard for 
>>>>>>>> someone
>>>>>>>> reading about this to understand that they should use it to resolve 
>>>>>>>> these
>>>>>>>> issues.
>>>>>>>>
>>>>>>>
>>>> The WebRTC samples <https://github.com/webrtc/samples> have always
>>>> been great for showing how a feature works but in this case even I have a
>>>> hard time coming up with a good sample.
>>>> Adoption is more likely from opensource projects currently working
>>>> around this like MediaSoup and then it trickles down from there. Who knows,
>>>> maybe Google Meet might drive adoption just to get rid of the "ignored"
>>>> datachannel ;-)
>>>>
>>>> I have updated the description accordingly.
>>>>
>>>> cheers
>>>>
>>>> Philipp
>>>>
>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Mon, Jan 26, 2026 at 7:31 AM Henrik Boström <[email protected]>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> This simple API plugs a missing piece of functionality: there is
>>>>>>>>> no way to tell WebRTC that you may want to use data channels at some 
>>>>>>>>> point
>>>>>>>>> in the future without having to re-negotiate, so the only way to 
>>>>>>>>> ensure you
>>>>>>>>> negotiate data channels unconditionally is to create these dummy data
>>>>>>>>> channels not to be intended for use that you have to manually ignore 
>>>>>>>>> at the
>>>>>>>>> other endpoint, both with regards to events firing and when parsing
>>>>>>>>> statistics.
>>>>>>>>>
>>>>>>>>> Creating dummy data channels is a workaround to a mistake in the
>>>>>>>>> API in my opinion, and I would support it even if all that it 
>>>>>>>>> achieved was
>>>>>>>>> API ergonomics.
>>>>>>>>>
>>>>>>>>> But it does achieve something more than the workaround is able to:
>>>>>>>>> placing the m= application line at the top. The fact that the order 
>>>>>>>>> of m=
>>>>>>>>> sections is more predictable reduces the risk of application bugs. I
>>>>>>>>> believe Google Meet had a serious regression a few years ago where 
>>>>>>>>> the m=
>>>>>>>>> section order was something not what we expected causing the SDP 
>>>>>>>>> parser to
>>>>>>>>> fail and the call to be disconnected. It's also a good idea that the 
>>>>>>>>> m=
>>>>>>>>> application line gets tagged, avoiding risk of rejecting the whole 
>>>>>>>>> bundle
>>>>>>>>> if the wrong m= section is rejected.
>>>>>>>>> On Thursday, January 22, 2026 at 3:20:18 PM UTC+1 Philipp Hancke
>>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>>> bear with me while I try to explain 15 years of history...
>>>>>>>>>>
>>>>>>>>>> Am Mi., 21. Jan. 2026 um 16:39 Uhr schrieb Rick Byers <
>>>>>>>>>> [email protected]>:
>>>>>>>>>>
>>>>>>>>>>> This is generally why we require explainers. Part of the point
>>>>>>>>>>> of this launch process is ensuring we've got the public material 
>>>>>>>>>>> for web
>>>>>>>>>>> developers to understand the feature. Harald can you post an 
>>>>>>>>>>> explainer
>>>>>>>>>>> somewhere which includes answers to Rob's questions?
>>>>>>>>>>>
>>>>>>>>>> Thanks,
>>>>>>>>>>>    Rick
>>>>>>>>>>>
>>>>>>>>>>> On Wed, Jan 21, 2026 at 9:30 AM Robert Flack <
>>>>>>>>>>> [email protected]> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> The specification just says "alwaysNegotiateDataChannels
>>>>>>>>>>>> defines a way for the application to negotiate data channels in 
>>>>>>>>>>>> the SDP
>>>>>>>>>>>> offer before creating a datachannel".
>>>>>>>>>>>>
>>>>>>>>>>> I'm finding it a bit hard to reason about what this means for me
>>>>>>>>>>>> as a developer. I.e. when should I set it and how does this change 
>>>>>>>>>>>> the
>>>>>>>>>>>> connection?
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>> Lets say you create a RTCPeerConnection. Create an offer and call
>>>>>>>>>> setLocalDescription. Send it to your peer. You get an answer from 
>>>>>>>>>> the peer
>>>>>>>>>> and feed it into setRemoteDescription.
>>>>>>>>>> You might wonder why it does not result in a connection.
>>>>>>>>>>
>>>>>>>>>> Next, you create a datachannel. Your datachannel never opens
>>>>>>>>>> because after adding the datachannel (only the first!) you need to 
>>>>>>>>>> call
>>>>>>>>>> createOffer/setLocalDescription and call setRemoteDescription with 
>>>>>>>>>> the
>>>>>>>>>> answer.
>>>>>>>>>> "web developers" are notably confused:
>>>>>>>>>>
>>>>>>>>>> https://stackoverflow.com/questions/79737908/how-do-i-create-a-data-channel-in-webrtc-js-firefox-115-6-esr
>>>>>>>>>> Now your connection gets established. If you add another data
>>>>>>>>>> channel you do *not* need to do this negotiation again (thankfully; 
>>>>>>>>>> adding
>>>>>>>>>> audio or video tracks requires negotiation).
>>>>>>>>>>
>>>>>>>>>> In this case using the boolean flag is saving you one round of
>>>>>>>>>> negotiation.
>>>>>>>>>>
>>>>>>>>>> In many applications you do not know if you want to use a
>>>>>>>>>> datachannel but since you might, you preemptively create one and 
>>>>>>>>>> throw it
>>>>>>>>>> away. You can see that e.g. in Google Meet which calls
>>>>>>>>>>    pc.createDataChannel("ignored", {reliable: true})
>>>>>>>>>> It never uses that channel but it shows up in the getStats API
>>>>>>>>>> every time.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> Does this mean that a data channel will be negotiated without
>>>>>>>>>>>> calling createDataChannel?
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>> The underlying SCTP association will be negotiated and can be
>>>>>>>>>> used by subsequent calls to createDataChannel without renegotiation.
>>>>>>>>>> This currently means exchanging four packets over the network
>>>>>>>>>> (but that upfront cost might go away later this year).
>>>>>>>>>> Everything is set up to just work.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> Will that requested datachannel result in a datachannel event or
>>>>>>>>>>>> does my code still need to call createDataChannel? If the latter, 
>>>>>>>>>>>> is this a
>>>>>>>>>>>> performance optimization?
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>> Part of it is a performance / ergonomics optimization. But
>>>>>>>>>> behold, there comes the 2011 backward compat angle below.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> Would I choose this when data is more important to connect first
>>>>>>>>>>>> rather than e.g. voice / video?
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>> Lets say you have a stream from getUserMedia with audio and video
>>>>>>>>>> and would also like to have a datachannel.
>>>>>>>>>> You call pc.createDataChannel and then iterate over the
>>>>>>>>>> MediaStream's getTracks and call pc.addTrack with each.
>>>>>>>>>> Because data channels were new and fancy in 2011 and backward
>>>>>>>>>> compat was a concern this results in SDP (due to rules in RFC 8829) 
>>>>>>>>>> which
>>>>>>>>>> puts the datachannel bits in the SDP after the audio and video bits 
>>>>>>>>>> (and in
>>>>>>>>>> Safari the order of audio and video bits depend on the lexical 
>>>>>>>>>> ordering of
>>>>>>>>>> the random uuid of the MediaStreamTrack) so the SDP looks roughly 
>>>>>>>>>> like this:
>>>>>>>>>>   v=0 (6 more lines)
>>>>>>>>>>   m=audio 9 UDP/TLS/RTP/SAVPF 63 111 (23 more lines)
>>>>>>>>>> direction=sendonly mid=0
>>>>>>>>>>   m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 45 46 (63 more
>>>>>>>>>> lines) direction=sendonly mid=1
>>>>>>>>>>   m=application 9 UDP/DTLS/SCTP webrtc-datachannel (9 more lines)
>>>>>>>>>> direction=sendrecv mid=2
>>>>>>>>>>
>>>>>>>>>> Which is ok but you are probably scratching your head why this
>>>>>>>>>> order does not match the order of calls (or why in Safari you get
>>>>>>>>>> video-audio-application half the time)
>>>>>>>>>> https://www.rfc-editor.org/rfc/rfc8829.html#name-initial-offers
>>>>>>>>>> specifies this (search for "Lastly, ") but does not provide much of a
>>>>>>>>>> rationale. https://github.com/w3c/webrtc-pc/issues/735 has some
>>>>>>>>>> 2016 rationale.
>>>>>>>>>> Now that seems like an odd choice but had little consequence
>>>>>>>>>> until...
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>>> The linked tracking bug <https://issues.chromium.org/433898678>
>>>>>>>>>>>> appears to be a chrome-only bug report with a particular demo that 
>>>>>>>>>>>> doesn't
>>>>>>>>>>>> use the alwaysNegotiateDataChannels attribute and works on 
>>>>>>>>>>>> Firefox. I don't
>>>>>>>>>>>> quite understand what the new configuration extension has to do 
>>>>>>>>>>>> with the
>>>>>>>>>>>> original bug report.
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>> WebRTC added APIs that allow you to negotiate codecs at some
>>>>>>>>>> point. This can result in a situation where you end up with a 
>>>>>>>>>> incompatible
>>>>>>>>>> set of video codecs (see the bug) and if those happen to be first in 
>>>>>>>>>> the
>>>>>>>>>> SDP this "m=" line will be marked as "rejected" and can not be used 
>>>>>>>>>> to
>>>>>>>>>> convey the transport information anymore.
>>>>>>>>>> At best this results in another round of negotiation to satisfy
>>>>>>>>>>
>>>>>>>>>> https://datatracker.ietf.org/doc/html/rfc8843#name-rejecting-a-media-descripti
>>>>>>>>>>
>>>>>>>>>> Now
>>>>>>>>>>   https://datatracker.ietf.org/doc/html/rfc8843#section-7.2.1
>>>>>>>>>> actually provides a recommendation on how to avoid this: you pick
>>>>>>>>>> something that is "unlikely to be rejected by the remote side".
>>>>>>>>>> The browser does not have an API that would lead to it rejecting
>>>>>>>>>> a datachannel m= line (which can happen for audio and video)
>>>>>>>>>> So it would be great to pick this datachannel m= line as "offerer
>>>>>>>>>> tagged" which avoids this headache.
>>>>>>>>>> But that requires (unless you want to risk more breakage) putting
>>>>>>>>>> this m= line first which is conflicts with the rules of RFC 8829.
>>>>>>>>>> The alternative is going through at least two rounds of
>>>>>>>>>> negotiation (which involves the remote end), the first datachannel 
>>>>>>>>>> only and
>>>>>>>>>> then adding media.
>>>>>>>>>> Hence you need a way to opt-in which is the boolean in the
>>>>>>>>>> RTCConfiguration.
>>>>>>>>>>
>>>>>>>>>
>>>>>> It sounds like you're saying here that this may not be backwards
>>>>>> compatible and if the other end doesn't support it you may not be able to
>>>>>> establish a connection. If so, is it possible for the developer to 
>>>>>> feature
>>>>>> detect whether the browser supports this so that they may determine 
>>>>>> whether
>>>>>> both ends support this or not before setting it?
>>>>>>
>>>>>> Would you like to hear more about why WebRTC is so weird? (a lot of
>>>>>>>>>> this boils down to "SDP is not an opaque blob)
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>>> On Wed, Jan 21, 2026 at 6:31 AM 'Philipp Hancke' via blink-dev <
>>>>>>>>>>>> [email protected]> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> *tContact emails*
>>>>>>>>>>>>> [email protected], [email protected]
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Specification*
>>>>>>>>>>>>>
>>>>>>>>>>>>> https://w3c.github.io/webrtc-extensions/#always-negotiating-datachannels
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Summary*
>>>>>>>>>>>>> Implement
>>>>>>>>>>>>> https://w3c.github.io/webrtc-extensions/#always-negotiating-datachannels
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Blink component*
>>>>>>>>>>>>> Blink>WebRTC
>>>>>>>>>>>>> <https://issues.chromium.org/issues?q=customfield1222907:%22Blink%3EWebRTC%22>
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Web Feature ID*
>>>>>>>>>>>>> Missing feature
>>>>>>>>>>>>>
>>>>>>>>>>>>> *TAG review status*
>>>>>>>>>>>>> Not applicable (incremental addition to WebRTC)
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Risks*
>>>>>>>>>>>>> *Interoperability and Compatibility*
>>>>>>>>>>>>> *No information provided*
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Gecko*: No signal (
>>>>>>>>>>>>> https://github.com/mozilla/standards-positions/issues/1335)
>>>>>>>>>>>>>
>>>>>>>>>>>>> *WebKit*: No signal (
>>>>>>>>>>>>> https://github.com/WebKit/standards-positions/issues/599)
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Web developers*: No signals
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Will this feature be supported on all six Blink platforms
>>>>>>>>>>>>> (Windows, Mac, Linux, ChromeOS, Android, and Android WebView)?*
>>>>>>>>>>>>> Yes
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Is this feature fully tested by web-platform-tests
>>>>>>>>>>>>> <https://chromium.googlesource.com/chromium/src/+/main/docs/testing/web_platform_tests.md>?*
>>>>>>>>>>>>> Yes, a WPT is part of
>>>>>>>>>>>>> https://chromium-review.googlesource.com/c/chromium/src/+/7080041
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Rollout plan*
>>>>>>>>>>>>> Will ship enabled for all users
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Requires code in //chrome?*
>>>>>>>>>>>>> False
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Tracking bug*
>>>>>>>>>>>>> https://issues.chromium.org/issues/433898678
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Estimated milestones*
>>>>>>>>>>>>> Shipping on desktop 146
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Link to entry on the Chrome Platform Status*
>>>>>>>>>>>>>
>>>>>>>>>>>>> https://chromestatus.com/feature/5113419982307328?gate=6516338099093504
>>>>>>>>>>>>>
>>>>>>>>>>>>> This intent message was generated by Chrome Platform Status
>>>>>>>>>>>>> <https://chromestatus.com/>.
>>>>>>>>>>>>>
>>>>>>>>>>>>> --
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>>>>>>>>>>>>> Google Groups "blink-dev" group.
>>>>>>>>>>>>> To unsubscribe from this group and stop receiving emails from
>>>>>>>>>>>>> it, send an email to [email protected].
>>>>>>>>>>>>> To view this discussion visit
>>>>>>>>>>>>> https://groups.google.com/a/chromium.org/d/msgid/blink-dev/CADxkKiKuE79-xWUFiO9NCy%2Big9Pvn1oi6fVbkdomFLo4--bJQw%40mail.gmail.com
>>>>>>>>>>>>> <https://groups.google.com/a/chromium.org/d/msgid/blink-dev/CADxkKiKuE79-xWUFiO9NCy%2Big9Pvn1oi6fVbkdomFLo4--bJQw%40mail.gmail.com?utm_medium=email&utm_source=footer>
>>>>>>>>>>>>> .
>>>>>>>>>>>>>
>>>>>>>>>>>> --
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>>>>>>>>>>>> it, send an email to [email protected].
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>>>>>>>>>>>> https://groups.google.com/a/chromium.org/d/msgid/blink-dev/CAJh39TPTxV5evdFUMnmfQE3aP2siNXNSZVpBr8wBitT56-nFCQ%40mail.gmail.com
>>>>>>>>>>>> <https://groups.google.com/a/chromium.org/d/msgid/blink-dev/CAJh39TPTxV5evdFUMnmfQE3aP2siNXNSZVpBr8wBitT56-nFCQ%40mail.gmail.com?utm_medium=email&utm_source=footer>
>>>>>>>>>>>> .
>>>>>>>>>>>>
>>>>>>>>>>> --
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