Re: Firefox does not accept valid srflx candidate

I'm not sure what you are reproing. Can you provide more detail about exactly what you are seeing? -Ekr On Sat, Feb 24, 2018 at 10:00 PM, wrote: > I can reproduce this on: https://webrtc.github.io/samples/src/content/ > peerconnection/trickle-ice/ > > With google's stun

Re: Firefox does not accept valid srflx candidate

I don't know if this is the issue, but the priority value of 0 is bogus. -Ekr On Wed, Jun 7, 2017 at 9:09 AM, Michael Zak wrote: > Hi > > This candidate is accepted by Chrome. > > I placed below the Firefox connectivity log where the srflx candidate was > discarded > > "

Re: audio stream with unlisted in SDP ssrc is audible

No, it's not a bug. a=ssrc is no longer necessary or advisable in JSEP. -Ekr On Fri, Mar 17, 2017 at 8:17 AM, Alexander Abagian wrote: > here's complete log > > https://yadi.sk/d/ehbdA8Vy3G6nNr > ___ > dev-media mailing list >

Re: Plans to make obsolete the Hangouts plugin?

This is a question for Google, not for Mozilla. Firefox has WebRTC support, so it's a matter of making Hangouts work on Firefox (and of course fixing the inevitable bugs in Firefox), not of adding new support to Firefox. -Ekr On Fri, Jul 8, 2016 at 11:20 AM, wrote: > Hi

Re: DTLS Encryption Alert and Firefox

On Tue, Jan 19, 2016 at 3:27 AM, Alexander Abagian wrote: > This question is mostly intended for Eric as being DTLS guru. > > I have a case when in the end of DTLS handshaking between our server and > Firefox, the server sends Encryption Alert (this name I see in Wireshark),

Re: Firefox encoding bitrate control

I'm not sure what you're looking for here. Nobody's arguing that we shouldn't implement these features, they're just not at the top of the priority list. If you'd like to submit a patch, it would certainly be welcome and one of us will be happy to review it. Other than that, I don't know what

Re: Firefox encoding bitrate control

On Fri, Dec 25, 2015 at 6:04 AM, Alexander Abagian wrote: > That means that there's not manual API controls for setting the needed > bitrates in FireFox, right ? Only an automatic one. That's not good. > We will eventually process the SDP attributes, as well as RtpSender. We

Re: Firefox encoding bitrate control

Hmm.. That's not what I see on calls between Nightly and Chrome on apprtc. This is more Jesup's area than mine, but I suggest that we're probably a the point where we're going to need logs from video engine. On Thu, Dec 24, 2015 at 11:05 AM, Alexander Abagian wrote: > I

Re: Firefox encoding bitrate control

Jesup is the expert on this code, but assuming I am reading the code correctly, the bitrate range is hardwired to 200-2000 (start = 300). kbps and can't be changed via SDP. However, this means that if your network is up to it, it should negotiate upward to 500kbps no problem. Can you describe

Re: Wrong SDP: Offer and answer have different number of m-lines

OK, then as far as I can tell, we're complying with the specification, so I'm not sure what you're looking for Firefox to do. -Ekr On Wed, Dec 9, 2015 at 5:46 AM, Alexander Abagian wrote: > Thanks, Eric, > > > If Chrome is allowing the answerer to add a third m= line, then

Re: Wrong SDP: Offer and answer have different number of m-lines

On Tue, Dec 8, 2015 at 2:28 PM, Alexander Abagian wrote: > If I add an extra m-line to remote descrition, I'm getting a "Wrong SDP: > Offer and answer have different number of m-lines (2 vs 3)" error. I don't > understand why must they be the same size. If I'm sending a

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

On Sun, Oct 18, 2015 at 3:37 AM, wrote: > Dear Mr. Roach, > > the answer below was regarding the recording of the video stream from a > webRTC peerconnection. I am currently building a webRTC-Tool where a user > should be able to start a recording using webRTC. This is part

Re: SDP offer with IP 0.0.0.0 from Firefox 41

It's for Trickle ICE: http://rtcweb-wg.github.io/jsep/#sec.initial-offers On Fri, Oct 2, 2015 at 9:21 AM, wrote: > Hi, > > Can you tell me what is the reasoning behind this? > Thanks. > ___ > dev-media mailing list >

Re: Forever reminded to hit ESC to exit fullscreen

On Sun, Aug 16, 2015 at 5:49 PM, Gavin Sharp ga...@gavinsharp.com wrote: But a 2-3 second box for each fullscreen transition seems like a small price. Seems like a pretty large price to me, given a combination of factors: - significant added friction to a common user action (start watching

Re: Forever reminded to hit ESC to exit fullscreen

that tradeoff, so having stated my opinion I'll defer to them. Gavin On Sun, Aug 16, 2015 at 6:16 PM, Chris Hofmann chofm...@mozilla.com wrote: On Sun, Aug 16, 2015 at 5:52 PM, Eric Rescorla e...@rtfm.com wrote: On Sun, Aug 16, 2015 at 5:49 PM, Gavin Sharp ga...@gavinsharp.com wrote

Re: Format of SDP

This isn't going to work very well, with any version of Firefox. The semantics of this SDP is that you are offering a bunch of audio streams, and that each of them is (for some unknown reason) with a different codec. Firefox prior to 37 only supports one audio stream so it will try to accept the

Re: webrtc screensharing questions

On Tue, Dec 9, 2014 at 11:29 PM, bryandonno...@gmail.com wrote: I've been working on screensharing using Firefox and I now have the basic case working for window sharing. That is: navigator.mozGetUserMedia( { video: { mediaSource : window } } ,function(stream){ },function(err){

Re: WebRTC connection success or failure dependant on who makes the offer?

://www.sharefest.me/ for example. On Fri, Nov 14, 2014 at 7:35 PM, Eric Rescorla e...@rtfm.com wrote: It shouldn't matter. I wonder if something is going wrong with your signaling system. Which calling site are you using? -Ekr On Fri, Nov 14, 2014 at 10:56 AM, useru...@gmail.com wrote: I

Re: Insertion of Custom Codecs into Firefox WebRTC Implementation

, 2014 8:59:34 AM UTC-7, Randell Jesup wrote: On 7/21/2014 11:04 AM, Eric Rescorla wrote: On Sun, Jul 20, 2014 at 5:06 PM, aam...@gmail.com wrote: Hi everyone, I was interested in modifying the Firefox implementation of WebRTC (for my own purposes), specifically

Re: Insertion of Custom Codecs into Firefox WebRTC Implementation

On Sun, Jul 20, 2014 at 5:06 PM, aam...@gmail.com wrote: Hi everyone, I was interested in modifying the Firefox implementation of WebRTC (for my own purposes), specifically, adding the MPEG-4 FBA codec to the implementation. I noticed that in the /media directory or Firefox source, there is

Re: GMP crash reporting update (schedule slip likely)

On Tue, Jul 15, 2014 at 1:19 PM, Benjamin Smedberg benja...@smedbergs.us wrote: The bug for hooking up crash reporting for GMP plugins is likely to slip past this week, and I'd like to talk about how to reduce risk there. Part of the problem is that I may not have broken that bug down

Re: Loop Server MVP: Protocol and Architectural Enhancements

On Mon, Jun 16, 2014 at 12:50 PM, Tarek Ziadé ta...@mozilla.com wrote: Le 16/06/14 11:54, ale...@mozilla.com a écrit : .. Maximum Simultaneous Connections 11,656,238 Current max sockets connections on an EC2 host is 250K. This means we'll need approx. 46 servers to handle the long term

Re: TURN server support in FF?

I don't understand what the problem is. As I said, Firefox has trickle ICE, so you need to see what is going into the ice candidates callback. Does that have TURN? -Ekr On Wed, Apr 16, 2014 at 12:02 AM, alexander.ro...@gmail.com wrote: Any updates on this? I am having the same problem here.

Re: a question about H.264-RTP of Firefox-webRTC on a OS which supports H.264 codecs, Thanks!

Firefox has never supported H.264 for WebRTC, though we do support it for the video tag on some platforms. We are working on H.264 support for WebRTC via Cisco's OpenH264, but that will be cross-platform and be distributed as a downloadable module. -Ekr On Tue, Apr 15, 2014 at 10:46 AM, Bo Xu

Re: Build web RTC native client

This mailing list is for Firefox media development, not development of the Google WebRTC code. You want discuss-webrtc. -Ekr On Fri, Apr 4, 2014 at 2:01 AM, Kamal Palei palei.ka...@gmail.com wrote: I am finding bit difficult to start with for a native web RTC client. Can somebody please

Re: No USE-CANDIDATE in stun request for WebRTC

it might be a real issue. Andreas On Thursday, March 20, 2014 8:09:28 PM UTC+1, Randell Jesup wrote: On 3/20/2014 2:18 PM, Eric Rescorla wrote: On Thu, Mar 20, 2014 at 10:40 AM, agra...@gmail.com wrote: Calling from FF 32 (nightly) to Chrome works fine, however calling from

Re: No USE-CANDIDATE in stun request for WebRTC

On Thu, Mar 20, 2014 at 10:40 AM, agra...@gmail.com wrote: Hi, Calling from FF 32 (nightly) to Chrome works fine, however calling from Chrome to FF fails. What I can see is FF sending a couple of STUN bind requests to the server, but it never puts the USE-CANDIDATE into any of them. Any

Re: TURN server support in FF?

On Sun, Mar 16, 2014 at 12:25 PM, Rajarshi Chaudhuri ccrajar...@gmail.comwrote: Hello - I'm using FireFox 27.0.1 (stable). Looks like though TURN server is specified while creating the PeerConnection, the ICE candidates in the SDP doesn't contain relay candidates. Is it because FF doesn't

Re: interim proposal for Loop code home process

Comments below: 1. WRT to the client code, why not simply git clone gecko-dev and make a branch? Git is good for this stuff. 2. I don't think we want a fork as a long-term prospect. Just for the next week or so while we work out the right thing. 3. I very much do not want to use a versioned hg

Re: Architecture Decisions and Action Items for User-Facing WebRTC Project

Perriault, Dan Mosedale, Eric Rescorla, Alexis Metaireau, and myself. Client Architecture Decisions * The panel JavaScript will ship with the browser and ride the trains, at least through MVP. * Local data storage will make use of the IndexedDB API. We may choose to wrap

Re: Firefox TURN auth problems

: https://gist.github.com/anonymous/71d5c804a196038f8d25 On Monday, February 10, 2014 11:13:15 PM UTC-6, Eric Rescorla wrote: These seem to be missing a lot of data. This is only covering ICE checks, not candidate gathering. Firefox will do

Re: mozilla, webrtc and TURN server

You appear not to have dialed the logging level up with R_LOG_LEVEL, thus we are only seeing errors and not debugging. Please re-run with R_LOG_LEVEL=9 -Ekr On Fri, Jan 31, 2014 at 1:07 AM, ijatsucast...@gmail.com wrote: Again thanks for answering. I tried on both windows and linux with

Re: Firefox support for a=ice-lite or STUN 487 Role Conflict error

Both of these are supposed to work. Please file a bug with a test case if possible. -Ekr On Fri, Nov 22, 2013 at 4:28 AM, mmr...@gmail.com wrote: Hi, Our testing indicate that Firefox (Nightly or Auroroa) do not support a=ice-lite in received SDP offer (on terminating side of the call) and

Re: Is there any way to get SSRC in firefox SDP

Not currently. It's on the list. -Ekr On Wed, Oct 16, 2013 at 2:35 AM, teih...@gmail.com wrote: Hi all! Chrome sends it's ssrc in attribute a=ssrc:21312312423 cname blah-blah. Is there any option to get ssrc in firefox? For example some js api, or any other way? Thank you in advance!

Re: TURN Server problems

Please file a bug that contains the SDP and wireshark traces. -Ekr On Tue, Sep 24, 2013 at 2:19 AM, rista...@gmail.com wrote: I just tried it with FF Nightly 27.0a1 (2013-09-23). Same problem. Please help. Thank you. On Tuesday, September 24, 2013 2:31:07 PM UTC+8, rist...@gmail.com

Re: TURN Server problems

It should work as of Firefox 26 (currently Aurora). But there may still be some glitches so please file a bug with wireshark traces and SDP logs if Aurora doesn't work and Chrome does. Thanks, -Ekr On Sun, Sep 22, 2013 at 8:42 PM, rista...@gmail.com wrote: Hello, Does FireFox support the

Re: [webrtc] Firefox doesn't send SRTP / SRTCP packets

You might also want to try running with NSPR_LOG_MODULES=mtransport:9,mediapipeline:9 On Tue, Sep 3, 2013 at 9:50 AM, Ethan Hugg ethanh...@gmail.com wrote: How can I get more logs / traces from firefox in order to debug this (firebug / firefox console doesn't print any errors) ? I don't

Re: STUN issue. 401 response

This should be fixed in Beta. https://bugzilla.mozilla.org/show_bug.cgi?id=886120 On Mon, Sep 2, 2013 at 1:16 AM, Jairo Canales Alfonso zyl1n3.mosto...@gmail.com wrote: Hello, I am working on a WebRTC client that uses ice4j to process STUN/TURN messages, which are in the WebRTC standard.

Re: about omxcodec for webrtc

Just to clarify this a bit: We're currently attempting to get a handle on the potential performance of HW video codecs in WebRTC via OMX. Since the current phones we have are H.264, that means prototyping H.264 via OMX but we expect the results to be fairly extensible to any HW codec, including

Re: WebRTC TURN support for Firefox ESR 24

Firefox 23 ships with TURN. -Ekr On Thu, Aug 8, 2013 at 6:25 PM, jenny...@gmail.com wrote: Hi, Does anyone know if webRTC TURN server support will be in Firefox ESR 24? If not, will there a webRTC plugin update to support this after the release? Thanks. Jenn

Re: TCP ICE candidates

TCP candidates are not currently supported. Minimally, it's behind TURN TCP on the current priority list. Of course if you wanted to submit a patch -Ekr On Tue, Jul 9, 2013 at 1:40 PM, Tom Hughes t...@vline.com wrote: I quickly looked through the Firefox code to see if it supports TCP

Re: I want my codecs ….

On Fri, Jun 21, 2013 at 9:27 AM, Ralph Giles gi...@mozilla.com wrote: On 13-06-21 9:14 AM, Eric Rescorla wrote: I believe you and cullen are thinking along different lines. You're thinking of static files but he's thinking of WebRTC. Since WebRTC has codec negotiation, the issues you

Re: peerConnection.getStats support in firefox

It is not supported yet. -Ekr On Thu, Jun 20, 2013 at 10:28 AM, saurabhs.srivast...@gmail.com wrote: I would like to know if peerConnection extension getStats API http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCStatsReport is supported in Firefox or not. Is there a way to

Re: WebRTC weekly meeting - Tues, June 4 at 9am, Pacific

On Tue, Jun 4, 2013 at 9:43 AM, bryandonno...@gmail.com wrote: The document at ( https://webrtc.etherpad.mozilla.org/weekly ) makes reference to SDES for telco. Does this mean that you may be adding SDES key exchange ? This is a reference to IETF activity, namely an upcoming IETF discussion

Re: Camera and mic control scenario of WebRTC

FWIW, Facetime's current behavior, is to freeze the current camera image but maintain the microphone. It would be easy for us to do the same. -Ekr On Sun, May 5, 2013 at 9:48 PM, s...@mozilla.com wrote: Hi all, As for the behaviour of backgrounding a real-time streaming video/audio

Re: DTLS-SRTP roles

On Tue, Apr 30, 2013 at 8:40 AM, lmini...@gmail.com wrote: Hi Eric, apologies for this very late reply to a post you sent a couple of months ago, but I only recently started trying to get DTLS working in Asterisk as well. I'm still stuck with a few other issues, but I'd like to focus on

Re: Nightly build test page

apprtc should work. if it doesn't, please report back. -Ekr On Wed, Apr 24, 2013 at 6:32 AM, Enda Mannion emann...@gmail.com wrote: Can anyone point me to a WebRTC test site or demo code I can set up to make a two way call. Something like https://apprtc.appspot.com/ Is this still

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

You may want the recording API: https://dvcs.w3.org/hg/dap/raw-file/tip/media-stream-capture/MediaRecorder.html On Thu, Mar 28, 2013 at 8:21 PM, Michael Heuberger michael.heuber...@binarykitchen.com wrote: thanks so much for your good advice. look, i already code with node.js and have an

Re: Quick Questions on Peer Connection for Testing

On Tue, Mar 5, 2013 at 12:08 PM, Jason Smith jsm...@mozilla.com wrote: Hi Everyone, Didn't get the chance to ask this in today's meeting, but I have some questions on the Peer Connection API: * What's the purpose of the onopen callback on Peer Connection? turns out this is an open

Re: Does Firefox support stun yet?

be repeatable :-). On Sunday, March 3, 2013 7:51:23 AM UTC-8, Eric Rescorla wrote: On Fri, Mar 1, 2013 at 8:14 PM, Eric Davies ericthecycl...@gmail.com wrote: We were running the current version of Firefox Nightly. We tried the apprtc.appspot.com example and the multi

Re: Does Firefox support stun yet?

On Fri, Mar 1, 2013 at 8:14 PM, Eric Davies ericthecycl...@gmail.comwrote: We were running the current version of Firefox Nightly. We tried the apprtc.appspot.com example and the multi-person video chat on http://mozilla.github.com/webrtc-landing/, and our own demo code. They worked fine as

Re: DTLS-SRTP roles

Firefox currently treats the caller as the server and the callee as the client. (This is the recommended configuration from 5763). Eventually we will do RFC 4572 roles as defined in RFC 5763/5764. -Ekr On Fri, Feb 22, 2013 at 5:17 AM, Mamadou Diop boss...@yahoo.fr wrote: Hello, I'm using

Re: Getting automated tests for WebRTC

On Wed, Nov 21, 2012 at 12:28 AM, jsmith.mozi...@gmail.com wrote: From what I'm getting from Randell and Ekr's feedback, it sounds like getting the smoke tests for the full stack of webrtc is the priority, not the crashtests. I think that makes logical sense as the primary focus area,

Re: MediaStreamGraph latency issues

There does not appear to be a link in this page. -Ekr On Tue, Nov 13, 2012 at 5:46 PM, Robert O'Callahan rob...@ocallahan.orgwrote: Randell, Maire and I talked about how to minimize latency for WebRTC and Web Audio with the MediaStreamGraph. I then talked about it some more with Ehsan and

Re: Q3 media goals

+1 On Mon, Jul 9, 2012 at 2:43 PM, Timothy B. Terriberry tterribe...@mozilla.com wrote: Maire Reavy wrote: I think we should remove the full getUserMedia backend working on Android goal for Q3. (I've quoted the current goal list below for reference.) I think this makes sense.