Re: Lossless video codec for web use?

On 6/13/19 5:28 PM, Eric Shepherd (Sheppy) wrote: Yeah, I know that lossless is huge, but was curious if there were any options since it will be a question that gets asked, and therefore I want to be prepared. My experiements suggest that even where there are codecs that have a lossless or near-l

Re: screen sharing and webcam video simultaneously

On 2/16/18 2:04 PM, Alexander Abagian wrote: I want to make Firefox stream both screen sharing and common webcam video simultaneously. Is it possible at all ? Chrome lets do it with an external screen sharing plugin, but here both of the should be produced by getUserMedia and I haven't find a

Re: Webextensions WebRTC API

Conveying addresses that the ICE stack isn't listening on would only make connection setup take longer and/or fail. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Re: [webrtc] Is DTMF support in the FireFox roadmap?

. So it's more likely to ship in Fx50 --which goes to DevEdition on August 1st and to Beta on Sept 12th (and releases in November). Luis -- see also https://hacks.mozilla.org/2015/11/webrtc-sending-dtmf-in-firefox/ for a stopgap measure that may work for you. -- Adam Roach Principal Pla

Re: Partial offer and answer support

certain glare conditions arising. So far, the MMUSIC and RTCWEB working groups have not decided that this kind of glare reduction is sufficiently important to adopt the work for finalization and publication. Basically: this work is dead, and unlikely to be resumed. -- Adam Roach Principal

Re: H.264 codec and/or SDP issues with RTCPeerConnection

hanks in advance and best regards, Christian ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/

Air Mozilla Studio

equipment they would typically have available -- say, a computer and a couple of cell phones (used as remote camera/mic units). There's a bit more information as well as a mock-up of the basic interface here: https://people.mozilla.org/~rmilewski/AirMoStudio/ -- Adam Roach Principal Platf

Re: How to force Hello Firefox to chose H.264 instead of VP8?

0/1 (note: the /1 is optional) You want to be the first in the list of payload values in the m=audio line. It's all just (careful) string manipulation. Don't assume any particular number will be used for ! -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903

Re: device selection and getUserMedia in privileged code

Yes, this work is underway. See bug 1046245. /a > On Mar 20, 2015, at 07:44, "jolic...@gmail.com" wrote: > > Hello, > > It seems impossible for privileged code (chrome://) to select a device for > getUserMedia ( > https://mxr.mozilla.org/mozilla-central/source/dom/media/MediaManager.cpp#1715

Re: [webdev] Blog Post: Privacy and Google Analytics

>>> http://gu.illau.me/posts/privacy-and-google-analytics/ >>> >>> A few months ago on the Marketplace team we discussed whether Google >>> Analytics should be disabled when Do Not Track is enabled. The consensus >>> was no - but with an asteri

IETF Internet Video Codec (NETVC) BoF approved

. Feel free to reach out to me with any process questions you may have. A description of the BoF is here: http://trac.tools.ietf.org/bof/trac/wiki#NETVC The proposed working group charter that will be discussed is here: https://daala.etherpad.mozilla.org/video-codec-charter -- Adam Roach

Re: Turn Server doesn't work if PC is under Airport WiFi AP

On 11/11/14 16:58, Adam Roach wrote: So, basically, this: The picture didn't make it through. It should have been: https://dl.dropboxusercontent.com/u/53717247/aslan-network.png -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800

Re: Turn Server doesn't work if PC is under Airport WiFi AP

something like that? Best Nils ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing li

Re: [loop-services-dev] Migration to MySQL for loop-server / hello.

thing offline for a while while copying data over, preserves the utility of all data still in use (while allowing you to abandon old cruft), and avoids surprising service behavior. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 __

Re: Warning: "Mandatory/optional in createOffer options is deprecated!..." in FF 33 Beta - what does it mean?

t;optional". You just pass the parameters you want to set. It's somewhat unfortunate that we *also* changed the uppercase/lowercase properties of the parameter at the same time, because it leads to this kind of confusion, but that's what happened. Last I checked, Chrome wasn't c

Re: WebRTC: missing media type in SDP -- FF 33 beta

? In the new format, "OfferToReceiveVideo" isn't valid. You need to use " offerToReceiveVideo". If you're missing the difference, look very carefully at the first letter of those two different strings. -- Adam Roach Principal Platform Engineer a...@mo

Re: WebRTC: missing media type in SDP -- FF 33 beta

See https://bugzilla.mozilla.org/show_bug.cgi?id=1063808 /a > On Sep 10, 2014, at 19:00, Rajarshi Chaudhuri wrote: > > Hi - > > For the current FF 33 beta, I see one issue with WebRTC. The problem is that > when the user has only one media type (say audio or video, but not both), and > creat

Re: Cumbersome URLs for sharing verbally/small screens

On 7/31/14 16:51, Jared Wein wrote: - Original Message - From: "Adam Roach" In practice, to prevent abuse by marketers and similar parties, we're going to need to make this space far more sparse than would be provided by 8 digits. If we were to provide the same proper

Re: Cumbersome URLs for sharing verbally/small screens

19 or 20. I'm not sure that's much of an improvement, for the use cases you describe. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozil

Re: Using a as a *source*?

On 7/14/14 13:20, Adam Roach wrote: On 7/14/14 13:15, Benjamin Smedberg wrote: Can I use a element as a source for webrtc/peerconnection? For GMP testing, I was hoping to be able to feed a known video through webrtc, and not have to worry about actual camera stuff. Not yet... Heh. I guess

Re: Using a as a *source*?

W3C media capture group recently (cf. http://www.w3.org/2014/05/19-mediacap-minutes.html#item05) I'm not sure where this is on the priority list of things to do. Maire would be the person to speak to that. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800

Re: Loop Server MVP: Protocol and Architectural Enhancements

/call-url/{token} does -- it allows adjusting URL data after the fact. So, for automatic generation, the client uses "POST /call-url/" with information that it automatically selects (i.e., without user intervention). The user can then, if he so chooses, modify that preselected infor

Re: Loop Server MVP: Protocol and Architectural Enhancements

hing that's MLP and only released in nightly. e.g. in case it breaks, I believe we should just let it break for now and focus on getting the new APIs right rather than having backwards compatibility at this stage of the project. I suppose a large portion of this depends on how quickly the

Re: Loop Server MVP: Protocol and Architectural Enhancements

tant thing, from a mid-term perspective, is that we need to be able to scale to more than one server if we get 11M or 12M users; and the design very much takes this into account, by allowing scaling across multiple servers. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903

Loop Server MVP: Protocol and Architectural Enhancements

https://wiki.mozilla.org/Loop/Architecture/MVP Note that this URL is a temporary scratch area for this work. Once it is more baked, the intention is to move it to a more final location within the main Architecture document and/or the Loop Server API documentation. -- Adam Roach Principal Platform Eng

Re: Reporting Loop bugs - needed information

all be automated prior to launch of what is considered a viable product. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Intent to land: Voice/video client ("Loop")

34 (33 is a stretch goal for the team, 34 is a committed date). Preference behind which this will be implemented: For initial landing on Nightly, none. Will be behind "loop.enabled" before riding the trains. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903

Re: No WebRTC team meeting this week (so we're free to attend the Town Hall about the FxOS goals for 2014)

Friday and the Friday following. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Re: Plans to alter gUM?

ll will be more like one of the others, so a user only has to click allow once on your site? Starting in Firefox 30, you should see the drop-down menu next to "Share Selected Devices" expand to include new options of "Always Share" and "Never Share." --

Re: WebRTC Service: On temporary call URLs

ersions in the URL path when and if they become necessary. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Re: WebRTC Service: On temporary call URLs

would you continue to use the service if you started getting nuisance calls on "mz.la/romain" as frequently as you get email spam? So I think the important thing to note is that we don't support what you propose right now, but we do have specific architectural hooks that will

Re: WebRTC service database selection

MVP. I would also like to reinforce Martin's call for a data store abstraction. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Re: WebRTC Service: On temporary call URLs

On 2/20/14 12:08, Dan Mosedale wrote: On Thu Feb 20 09:15:15 2014, Adam Roach wrote: Keep in mind that the nominal user experience for this, once we get to MVP, will be rooted in an address book. I'll have my list of contacts, which includes phone numbers and email addresses. When an ad

Re: WebRTC Service: On temporary call URLs

On 2/20/14 04:54, Alexis Métaireau wrote: Adam, thanks for starting this discussion, Le 20/02/2014 04:52, Adam Roach a écrit : Putting this together, what we want is something that semantically evaluates to: http://authority/action/url-format-version/{serial #,caller,callee,expiration,hmac

Re: WebRTC Service: On temporary call URLs

asks for a full URL, and the server provides one. The client is in no way involved in synthesis of the URL. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://l

Re: WebRTC Service: On temporary call URLs

On 2/19/14 22:03, Timothy B. Terriberry wrote: Adam Roach wrote: Now, revocation is inherently state that we need to store; but revocation events should be infrequent. Nonetheless, we don't want to I assume you've thought about just using time-based expiration instead? The iss

WebRTC Service: On temporary call URLs

rty's contact information -- a...@nostrum.com in this example -- but for MLP, this should be sufficient) -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Architecture Decisions and Action Items for User-Facing WebRTC Project

tion of available databases. * L: *Alexis* to create repo in under mozilla github; *Adam* to send him list of team members so they can be added to the committers. * L: *Mark* to investigate state of github/bugzilla integration. [1] That's "Minimum Landable Product" (aka

Re: peerConnection getStats

at doesn't go against your morals? ;) Sure, send me a URL to look at. :) -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Re: peerConnection getStats

ormation in the stats object now. I've attached a primitive demo that just sets up a couple of peer connections, points them at each other, and dumps the raw stats objects out for you to see. -- Adam Roach Principal Platform Engineer a...@mozilla.co

Re: peerConnection getStats

the moment. For example, compare the stats that Firefox 27 exposes: http://mxr.mozilla.org/mozilla-beta/source/dom/webidl/RTCStatsReport.webidl With those from Firefox 29: http://mxr.mozilla.org/mozilla-central/source/dom/webidl/RTCStatsReport.webidl -- Adam Roach Principal Platform Engineer a

Re: mozilla, webrtc and TURN server

and will not attempt to authenticate itself to the TURN server. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Re: iceConnectionState does not go to Disconnected

ozilla.org/show_bug.cgi?id=852665 I have updated the title of that bug to be slightly more descriptive. Is there currently any way for an app to detect that it must restart ice? Unfortunately, there's not a really good way to detect this situation at the moment. -- Adam Roach Pri

Re: ICE-TCP and BUNDLE

. What you describe isn't real-time (at least, not in the way WebRTC uses that term), so WebRTC doesn't really seem like the right tool. In fact, your application seems like it would benefit greatly from the kind of multi-second buffering that we would consider to be wildly unacceptable f

Re: webRTC and cisco openH264

nH264's binary from Cisco when they install Firefox? or Mozilla will include openH264's binary inside the new version of Firefox? Thanks! Users will not need to take any additional actions, and shouldn't notice any difference in behavior. Firefox will automatically download the

Re: ICE-TCP and BUNDLE

bove, you can set up your conference server to act as a TURN-TCP server (just speak the TURN over TCP protocol, with RTP encapsulated inside it, but process and generate the RTP locally), and configure your PeerConnections to use it as their sole ICE server. -- Adam Roach

Re: Live video streaming for Firefox OS?

On 10/25/13 16:59, Felix E. Klee wrote: On Fri, Oct 25, 2013 at 11:50 PM, Adam Roach wrote: low-latency live streaming. Actually, latency isn't a big issue, as long as it's no more than say a second. Yeah, an icecast-like thing is probably the way to go, then. Once you get your

Re: Live video streaming for Firefox OS?

en WebRTC could certainly be used for low-latency live streaming. You'll probably find yourself doing quite a bit of custom development for this, at least, in the short term. But should certainly be feasible. I believe the current roadmap has WebRTC landing in FirefoxOS 1.3. -- Adam Roac

Re: PeerConnection.removeStream

/2013/09/webrtc-update-and-workarounds/ -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Re: MediaStream constructor

xpect your clone trick to work, once we get cloning working. You may wish to follow this bug to keep abreast of progress: https://bugzilla.mozilla.org/show_bug.cgi?id=910249 -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863

Re: MediaStream constructor

hing. If you were to somehow manage to construct one in a vacuum, what source would you expect it to represent? In other words: what are you trying to do? -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media ma

Re: SDP Parsing Error: End of line beyond end of buffer.

made by Firefox that the SDP will never reach this size? ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media -- Adam Roach Principal Platform Engineer a...@mozilla

Re: clone MediaStream

of MediaStream or MediaStreamTrack. This is currently being tracked in Bug 910249 (see https://bugzilla.mozilla.org/show_bug.cgi?id=910249), although cloning a large enough block of functionality that it might be broken out into its own bug at some point. -- Adam Roach Principal Platform

Re: Question on DTLS roles

"actpass" indicating that it's willing to assume either role). That is a different problem, and not one that I've seen. Can you send me a copy of the offer and the answer so I can try to figure out what's going wrong here? -- Adam Roach Principal

Re: ice-lite support

. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Re: Firefox failing to parse Chrome's SDP on basic video call setup

dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Re: How it works. Video streaming in WebRTC

__ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Re: How to turn on WebRTC debug logs on Firefox?

uot;,timestamp:1" to that list if you want each log message to include timestamps. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Re: webrtc constraints

y ought to open another bug for plumbing this into the media engine, similar to what we're doing in Bug 881935 right now. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.m

Re: webrtc constraints

ng, as are the WebRTC related specifications, and many features are not yet implemented. See: https://hacks.mozilla.org/2013/04/webrtc-update-our-first-implementation-will-be-in-release-soon-welcome-to-the-party-but-please-watch-your-head/ ...and... https://hacks.mozilla.org/2013/06/webrtc-comes-t

Re: onicecandidate plan

following this issue once work starts on it, add your name to the CC list for this bug: https://bugzilla.mozilla.org/show_bug.cgi?id=842549 -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list

Re: In onaddstream-Handler, 'this' does not equal the corresponding RTCPeerConnection-Object

ot;} If this doesn't work for you, please let us know. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Re: changing ice credentials in answer

finer point on it: I'm probably the one who will be implementing the SDP editing handling, and I did not plan on allowing these values to be changed. Can you explain why you would want to do this? -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903

Re: Does FF supports h264 RTP packetization?

daneich.com/2012/10/html5-video-update/). -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Re: In onaddstream-Handler, 'this' does not equal the corresponding RTCPeerConnection-Object

er) should refer to the corresponding RTCPeerConnection-Object. Right? Cheers Kai ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media -- Adam Roach Principal Platform Engineer a...@mozilla.com

Re: WebRTC Test Day on June 21st

esting complete state." It implies that Nightly contains code that is some substantial portion of the way -- say, more than 50% -- to being feature complete. I think that's misleading. -- Adam Roach Principal Platform Engineer a...@mozilla

Re: webrtc firefox calling chrome

you might want to insert the following lines in Firefox's video m= section: a=rtcp-fb:120 ccm fir a=rtcp-fb:120 nack Your best bet is to search for " a=rtpmap:120 VP8/9" and insert those two lines directly after it. -- Adam Roach Principal Platform Engineer a...@mo

Re: Reflection Comments from a Recent Experience of Building Three End to End WebRTC Web Applications

n. This is likely indicative of a more serious bug than just an inscrutable error message. -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media

Re: Fwd: Re: Nightly build test page

nd of NAT issue. What's your topology? Are you trying to call between two machines on the same network, or are you trying to talk to someone behind a different NAT than you are? -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 _

Re: WebRTC is not a Peer 2 Server Solution. How can I make one?

g defined in the W3C's "WebRTC" working group, and the browser-to-network interface being defined in the IETF's "RTCWEB" working group. The term "WebRTC" is used in the press to refer to both halves of the effort. You won't generally see "RTC

Re: WebRTC is not a Peer 2 Server Solution. How can I make one?

a solution yet and doubt it's possible with the current WebRTC implementation. Am I right? Nope. The browser-to-server use case is explicitly called out in the RTCWEB requirements document as something we expect to support. There should be nothing preventing you from making use of it alre

Re: Does Firefox support stun yet?

Yes. If you don't specify a server, it'll use Mozilla's STUN server by default. At least, for now. You might need to be running Aurora or Nightly for this to work properly. Also, we don't currently take DNS names for STUN servers, although we're working on it at the moment. /a On Mar 1, 20

Re: WebRTC weekly meeting postponed

On 2/18/13 15:13, Maire Reavy wrote: A number of us will be physically attending or remotely supporting SIPit-30 tomorrow (all day). So we're moving the WebRTC weekly meeting from Tuesday at 9am Pacific to Thursday or Friday morning. If anyone who is planning to attend has a preferred day or t

Re: goto in webvtt parser code

ptions, the judicious use of goto for error handling conditions is, IMHO, perfectly fine. There might be alternatives that fit the situation better, and those decisions can be made on a per-instance basis; but treating "goto" as a strictly verboten four-letter word removes a useful tool fr

Re: WebVTT vs/ WebRTC

On 1/23/13 11:57, Ralph Giles wrote: Does that help with the relative context? -r Thanks! Yes, it does. I think what I missed (but David caught) was that "mozilla/webrtc repo" was a typo for "mozilla/webvtt repo." It was your mention of WebRTC that prompted my question. /a _

Re: Current State Of WebVTT

On Jan 22, 2013, at 19:08, Ralph Giles wrote: > When > we're happy we at least understand the state of the code, we'll merge > into the mozilla/webrtc repo and work can continue there, with ongoing > review of smaller pull requests. Ralph: Thanks for looping the rest of the media team in. What

Re: CU-RTC-Web

o provide some of the counterargument to the argument Microsoft makes in the article below. Having said all that, anything that inspires the working group members to move a little faster in discussions and decision making is a good thing IMO. :-) -Maire On 1/20/2013 1:31 PM, Adam Ro

CU-RTC-Web

Not good news for market adoption... http://arstechnica.com/information-technology/2013/01/microsoft-goes-its-own-way-with-web-audiovideo-spec-despite-w3c-rebuff/ /a ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listin

Re: WebRTC on iOS

0/12 09:09, Jb Piacentino wrote: Adam, I have asked for a technical conference call with Ericsson Labs. I hope they will accept that and you will be able to ask questions and maybe cooperate. More on this soon. Jb On 29/11/12 22:00, Adam Roach wrote: On 11/29/12 12:26, Jb Piacentino wrote: I

Re: WebRTC on iOS

monolithic nature of UIWebView, how did Ericsson manage to put new, non-Apple-supported javascript APIs in an iOS web browser application? -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list

WebRTC on iOS

Apple. https://itunes.apple.com/us/app/bowser/id560478358 -- Adam Roach Principal Platform Engineer a...@mozilla.com +1 650 903 0800 x863 ___ dev-media mailing list dev-media@lists.mozilla.org https://lists.mozilla.org/listinfo/dev-media