Infinity Years Left in Google Chrome #59

mazkopolo · 2013-09-27T00:10:07Z

When I run the sharefest, I get the following message on my browser. Any idea how I can solve this issue?

"Download Rate: 0.00KB/s , Infinity years left "

shacharz · 2013-09-28T05:57:32Z

Either you're trying to download from a swarm that has no peers currently in OR you can connect to the peers that are in the swarm.
What does it say on the "notification bar" at the top of the drop zone?
If you enter the demo swarm does it work? (on the menu bar)

mazkopolo · 2013-09-29T23:14:48Z

Thanks for the reply.
I followed the following steps:

Install nodejs
Download this repo, or git clone https://github.com/Peer5/ShareFest.git
cd ShareFest
npm install --dedupe to install dependencies.
npm start to start the server
http://localhost:13337 should work

When I want to share a file with a user, I keep getting the same error. Sorry I am new to Sharefest and have no idea what the SWARM is.
When I use Sharefest.me, everything seems to work perfectly. But when I use mine(http://webrtc.thehii.com:13337), it keeps saying the same thing. Could you please test my server and see where the problem is coming from?

shacharz · 2013-09-30T06:54:08Z

I haven't tested your server yet,
but there is a bug in the npm start,
try doing "node server.js" instead, to start the node server.
If that doesn't work, check if there are any errors in the server
update:
I just tested with your server and it worked for me.

mazkopolo · 2013-09-30T23:44:25Z

Thanks for your comment. I am now using https and I have no error in my err.log.
However, I have no "notification bar". Could you please take a look at https://webrtc.thehii.com?
When I upload a file in https://sharefest.com, it gives me "You are the only peer with this file. Please don't close this tab yet. why?" but I have never seen such thing when I drag and drop a file into the box in my server.
The same story happens on the receiver side.

Thanks much sir,
Maziar

mazkopolo · 2013-10-01T01:24:13Z

btw, it works with Firefox! But not with Chrome :( I am using the latest version of Chrome for both peers. In Firefox I can also see the Notification Bar.

shacharz · 2013-10-01T11:13:39Z

I've tested both with Firefox and Chrome they both seem to work. but the only notification you'll see is the one related to "finished the download", I've yet to push the public version to git.
I still don't understand why it wouldn't work on your Chrome, I need a more extensive log:
In clientConfig change LOG_LEVEL to 3 and do this test:
In Chrome - incognito mode, add a file to sharefest, open the url in another tab (same computer) and pastebin the logs from the 2 tabs.

mazkopolo · 2013-10-01T22:26:39Z

It works between two tabs in one browsers.
However, when I use two browsers, I got the following logs:
maziar@maziar-VirtualBox:/opt/google/chrome$ ./chrome --enable-logging --v=3
[11395:11487:1002/112440:ERROR:cert_verify_proc_nss.cc(853)] CERT_PKIXVerifyCert for webrtc.thehii.com failed err=-8179
[11395:11395:1002/112440:ERROR:browser_window_gtk.cc(1024)] Not implemented reached in virtual void BrowserWindowGtk::WebContentsFocused(content::WebContents*)
[1002/112447:VERBOSE2:webrtcvoiceengine.cc(329)] SetTraceCallback() failed, err=0
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(365)] WebRtc VoiceEngine codecs:
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(391)] ISAC/16000/1 (103)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(391)] ISAC/32000/1 (104)
[1002/112447:VERBOSE2:webrtcvoiceengine.cc(411)] Unexpected codec: ISAC/48000/1 (105)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(391)] PCMU/8000/1 (0)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(391)] PCMA/8000/1 (8)
[1002/112447:VERBOSE2:webrtcvoiceengine.cc(411)] Unexpected codec: PCMU/8000/2 (110)
[1002/112447:VERBOSE2:webrtcvoiceengine.cc(411)] Unexpected codec: PCMA/8000/2 (118)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(391)] opus/48000/2 (111)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(391)] CN/8000/1 (13)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(391)] CN/16000/1 (105)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(391)] CN/32000/1 (106)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(391)] CN/48000/1 (107)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(391)] telephone-event/8000/1 (126)
[1002/112447:VERBOSE3:webrtcvideoengine.cc(838)] WebRtcVideoEngine::WebRtcVideoEngine
[1002/112447:VERBOSE2:webrtcvideoengine.cc(858)] SetTraceCallback(0x2bcd7104a6e0) failed, err=0
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(449)] WebRtcVoiceEngine::Init
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(480)] WebRtc VoiceEngine Version:
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(145)] VoiceEngine 4.1.0
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(145)] Build: svn:Unavailable(issue687) Sep 16 2013 15:49:43 r
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(145)] External recording and playout build
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(662)] Applying audio options: AudioOptions {aec: true, agc: true, ns: true, hf: true, swap: false, typing: true, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, aec_dump: false, }
[1002/112447:VERBOSE2:webrtcvoiceengine.cc(724)] SetTypingDetectionStatus(1) failed, err=8003
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(1291)] Adjusting AGC level from default -3dB to -3dB
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(495)] WebRtc VoiceEngine codecs:
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(498)] opus/48000/2 (111)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(498)] ISAC/16000/1 (103)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(498)] ISAC/32000/1 (104)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(498)] PCMU/8000/1 (0)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(498)] PCMA/8000/1 (8)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(498)] CN/48000/1 (107)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(498)] CN/32000/1 (106)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(498)] CN/16000/1 (105)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(498)] CN/8000/1 (13)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(498)] telephone-event/8000/1 (126)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(452)] WebRtcVoiceEngine::Init Done!
[1002/112447:VERBOSE3:webrtcvideoengine.cc(905)] WebRtcVideoEngine::Init
[1002/112447:VERBOSE1:cpumonitor.cc(143)] open proc/stat failed:: [0x00000001] Operation not permitted
[1002/112447:VERBOSE1:webrtcvideoengine.cc(911)] Failed to start CPU monitor.
[1002/112447:VERBOSE3:webrtcvideoengine.cc(926)] WebRtcVideoEngine::InitVideoEngine
[1002/112447:VERBOSE3:webrtcvideoengine.cc(944)] WebRtc VideoEngine Version:
[1002/112447:VERBOSE3:webrtcvideoengine.cc(279)] VideoEngine 3.33.0
[1002/112447:VERBOSE3:webrtcvideoengine.cc(279)] Build: svn:Unavailable(issue687) Sep 16 2013 15:50:35 r
[1002/112447:VERBOSE3:webrtcvideoengine.cc(917)] VideoEngine Init done
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(662)] Applying audio options: AudioOptions {aec: true, agc: true, ns: true, hf: true, swap: false, typing: true, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, aec_dump: false, }
[1002/112447:VERBOSE2:webrtcvoiceengine.cc(724)] SetTypingDetectionStatus(1) failed, err=8003
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(1291)] Adjusting AGC level from default -3dB to -3dB
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(838)] Setting microphone to (id=0, name=Default device) and speaker to (id=0, name=Default device)
[1002/112447:VERBOSE3:webrtcvoiceengine.cc(913)] Set microphone to (id=0 name=Default device) and speaker to (id=0 name=Default device)
[11395:11411:1002/112447:ERROR:cert_verify_proc_nss.cc(853)] CERT_PKIXVerifyCert for webrtc.thehii.com failed err=-8179
[11:11:1002/112452:ERROR:rtc_data_channel_handler.cc(63)] WebRTCDataChannelHandlerClient not set.
[11:11:1002/112452:ERROR:rtc_data_channel_handler.cc(63)] WebRTCDataChannelHandlerClient not set.

And it doesn't do any thing. When I compare it with sharefest.me, I got the same log with the following added to it after a while:

[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1591)] Setting receive voice codecs:
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1625)] opus/48000/2 (111)
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1625)] ISAC/16000/1 (103)
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1625)] ISAC/32000/1 (104)
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1625)] PCMU/8000/1 (0)
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1625)] PCMA/8000/1 (8)
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1625)] CN/48000/1 (107)
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1625)] CN/32000/1 (106)
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1625)] CN/16000/1 (105)
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1625)] CN/8000/1 (13)
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1625)] telephone-event/8000/1 (126)
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(2807)] Starting playout for channel #0
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1813)] Disabling NACK for stream 0
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1821)] Selected voice codec opus/48000/1 (111), bitrate=32000
[1002/112732:VERBOSE2:webrtcvoiceengine.cc(1727)] CN frequency 48000 not supported.
WARNING: no real random source present!
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1860)] Enabling audio level header extension with ID 1
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1564)] Setting voice channel options: AudioOptions {}
[1002/112732:VERBOSE3:webrtcvoiceengine.cc(1582)] Set voice channel options. Current options: AudioOptions {}

mazkopolo · 2013-10-02T04:20:45Z

I don't know for what reason I can't see anything in my notification bar after I upload a file. I also get a warning/error as follow:

mazkopolo · 2013-10-03T03:08:36Z

I used wireshark to compare the traffic between my server and sharefest.me. Interestingly, when I used my server, I don't see any STUN packets in wireshark, but I see a lot of STUN packets while I am sending/receiving data from sharefest.me. Is there an specific settings for STUN in "./sharefest/clientConfig.js"?
Mine was set to STUN_SERVERS:['stun.l.google.com:19302'].

shacharz · 2013-10-03T18:29:49Z

Production Sharefest uses our own ICE servers, I don't believe google's are inferior.
but it's an interesting case to check, since firefox uses their own stun servers, so the symptoms might align here.
test in the same use case that didn't work https://apprtc.appspot.com and see whether the communication there is direct or relayed

mazkopolo · 2013-10-04T02:40:41Z

I created one TURN server on Amazon EC2 with the following address:
50.112.128.192:3478
username: mcd
password: mcd
I tested my turn server with TCPDUMP and stunclient and it accepts connections.

and did the following settings in /core/apiValidators/DataChannelsValidator.js:

     try {
         var RTCPeerConnection
         if (window.webkitRTCPeerConnection) {
             RTCPeerConnection = window.webkitRTCPeerConnection;
             //var SERVER = "stun:stun.l.google.com:19302";
                     var SERVER = "50.112.158.192:3478";
             servers = {iceServers:[
                 {url:"stun:" + SERVER}
             ]};

and the following in ClientConfig.js

peer5.config = {
LOG_LEVEL:5,
MAX_PENDING_CHUNKS:200, //max number of chunks pending per peer
MOZ_MAX_PENDING_CHUNKS:8, //max number of chunks pending per peer for mozilla
CHUNK_SIZE:800,
CHUNK_EXPIRATION_TIMEOUT:1500,
REPORT_INTERVAL:10000,
STAT_CALC_INTERVAL:1000,
MONITOR_INTERVAL:1000,
STUN_SERVERS:[],
TURN_SERVERS:["[email protected]:3478"],
TURN_CREDENTIALS:["mcd"],
P2P_PREFETCH_THRESHOLD:100,
PC_FAIL_TIMEOUT:15000,
PC_RESEND_INTERVAL:1000,
SOCKET_RECONNECTION_INTERVAL:2000,

ALLOWED_FILE_SIZE:250*1024*1024, //in bytes 250MB
USE_FS:true,
CACHE_SIZE:50000000, //in bytes
FS_ROOT_DIR:'peer5/',
FS_SIZE: 4294967296 //in bytes

};

When I use sharefest.me, I can see packets go to your stun/turn server (199.115.112.148:3478) in wireshark. However, I can't see any packets going to my stun/turn server.
Could you please take a look at my configurations and let me know if there is any misconfiguration?

Thanks much

shacharz · 2013-10-04T06:00:20Z

When I use sharefest.me, I can see packets go to your stun/turn server (199.115.112.148:3478) in wireshark. However, I can't see any packets going to my stun/turn server.

why would your changes apply in sharefest.me?

have you tried the apprtc demo on the same use-case?

I could reproduce your problem only between chrome M29 and chrome M32,
Is that your use case?
In this case it looks like a server problem: no matches are being send to anyone of the clients and they're not trying to connect. I suspect a user agent identification problem or something of that sort, try to debug in the server side step by step why after the second peer joins no matches are being sent

mazkopolo · 2013-10-06T23:14:51Z

I have tried apprtc demo and it works both in Chrome and Firefox.
I am using Google Chrome "Version 30.0.1599.69 m" on two machines in a local network.
I installed the shareseft on Ubuntu on another local machine in the same network.
This solutions perfectly works with Firefox but I still get the same error in Chrome.
I guess this is a server problem. It seems the webrtc in sharefest.me is different from the one in Github. Could you please email me the latest source code in sharefest.me (which is working perfectly in both Chrome and Firefox) so that I can find the difference. I will surely let you know where the problem/difference was.

my email is mazkopolo at gmail . com

Meanwhile I will work on debugging the server.
Cheers,
Maziar

mazkopolo · 2013-10-07T21:38:22Z

Thanks for your responses.
Sharefest works on two tabs in Google Chrome (I assume because it does not look for any ice,stun, or turn server). However, when I use two machines (even in a local network) it gives me the "Infinity years left" notification.
When I use sharefest.me in my local network and capture packets with wireshark, I happen to see some stun request but it never happens with my server. So, I reckon, I should check to see the reason why the Chrome doesn't generate any stun packets with my server.

Just a quick update for sharefest.me:
It seems when we use "Firefox" with Sharefest.me, all the stun packets go through firefox stun server IP address(23.21.150.121), In other words, they never happen to reach your stun server @(199.115.112.148).
But when we use chrome, I see the flow of packets going towards your server.

Cheers,
Maziar

mazkopolo · 2013-10-08T03:21:50Z

I guess I am getting closer to solve that issue. As far as I tried different webrtc solutions such as webrtc.io I should confirm that the default stunserver for google does not work (And it could be the reason why yours is working because you have different STUN server).

My main question is why there are two places to put the STUN configurations in this project. One in /core/apiValidators/DataChannelsValidator.js

if (window.webkitRTCPeerConnection) {
RTCPeerConnection = window.webkitRTCPeerConnection;
var SERVER = "stun:stun.l.google.com:19302";

                    servers = {iceServers:[
                        {url:"stun:" + SERVER}
                    ]};

and the other one in sharefest/clientConfig.js:

STUN_SERVERS:['stun.l.google.com:19302'],

As far as I know, the ClientConfig.js is the one the webrtc client uses (because I could see the configuration in client.js in Chrome logs). So, what is "/core/apiValidators/DataChannelsValidator.js"? Why there is another place to put the STUN configuration?

Thanks,
Maziar

mazkopolo · 2013-10-09T02:35:58Z

I have done the code debugging and found out that the problem comes from where the Chrome Receiver is not able to establish a STUN connection. Then I switched it into incognito mode and it WORKED!
Do you happen to know why It does not work in Normal Mode but it works in incognito mode?

shacharz · 2013-10-09T08:41:08Z

Weird, first time I've heard of such a phenomenon.
Maybe you have a kind of a script blocker, or a privacy tool that gets in the way?
Have you tried clearing cookies and such?

mazkopolo · 2013-10-10T01:23:03Z

Yes, I cleared all the cookies and such. I also tested in on two brand-new machines. I am pretty sure there is no blocker in my Chromes.
I don't understand why it simply works with Chrome Incognito mode.
I compared the client.js from sharefest.me and client.js from my server and didn't see much difference.
May I please ask you to send me the working source code from sharefest.me?
It will be a great help to find the problem.

Cheers,
Maziar

mazkopolo · 2013-10-11T02:59:57Z

I got the following from the logs in Chrome incognito mode.
Should SDP declare the 127.0.0.1 for the IP address or caller/callee?

Sending SDP message:
v=0
o=- 1142263659006405532 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio data
a=msid-semantic: WMS
m=audio 1 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
c=IN IP4 0.0.0.0
a=rtcp:1 IN IP4 0.0.0.0
a=ice-ufrag:jT/ddL7IHNPYhIyi
a=ice-pwd:wvFRrFsvXEiiMCT2nhvdUERV
a=ice-options:google-ice
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=recvonly
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:xU1VClsxPIgaqq6cjfQsr3Xql4qPAt7bfITm/PmM
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 CN/48000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
m=application 1 RTP/SAVPF 101
c=IN IP4 0.0.0.0
a=rtcp:1 IN IP4 0.0.0.0
a=ice-ufrag:jT/ddL7IHNPYhIyi
a=ice-pwd:wvFRrFsvXEiiMCT2nhvdUERV
a=ice-options:google-ice
a=mid:data
a=sendrecv
b=AS:1638400
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:xU1VClsxPIgaqq6cjfQsr3Xql4qPAt7bfITm/PmM
a=rtpmap:101 google-data/90000
a=ssrc:79792429 cname:OzdHdR1KO1Y1zGyy
a=ssrc:79792429 msid:0ebb49cb-0cb2-458e-a6b6-1889fd37e91d401fdfbd-f5a8-48e1-b99e-06e7418b8d49 0ebb49cb-0cb2-458e-a6b6-1889fd37e91d401fdfbd-f5a8-48e1-b99e-06e7418b8d49
a=ssrc:79792429 mslabel:0ebb49cb-0cb2-458e-a6b6-1889fd37e91d401fdfbd-f5a8-48e1-b99e-06e7418b8d49
a=ssrc:79792429 label:0ebb49cb-0cb2-458e-a6b6-1889fd37e91d401fdfbd-f5a8-48e1-b99e-06e7418b8d49

mazkopolo · 2013-10-12T04:11:48Z

As far as I could compare the client.js on both my server and sharefest.me server, I found out that the following functions/subclass haven't been implemented in the github version of sharefest.me:
1- peerConnectionStateValid: function()
2-peer5.core.contollers.AbstractController = peer5.core.controllers.IController.subClass

shacharz · 2013-10-12T08:46:48Z

Have you tried other app's using google's stun to see if the problem is there?
Either way I"m recommending creating your own stun server using https://code.google.com/p/rfc5766-turn-server/

mazkopolo · 2013-10-14T01:16:11Z

I have installed my own STUN server(54.200.117.227 using "gorst" as a username and "hero" as a password). The STUN server works properly (as you can see in attached images) but my server (https://webrtc.thehii.com) still works in Incognito mode only!
The following shows that sharefest does not work in normal mode and Wireshark could capture nothing.

However, it works in incognito mode and wireshark could capture STUN packets with "CreatePermission success response".

Any idea?

mazkopolo · 2013-10-15T04:00:22Z

registerEvents:function in both "core/transport/PeerConnectionImpl.js" and "core/controllers/P2PController.js" are never get called in the Normal mode.

shacharz · 2013-10-15T08:14:31Z

Are there no errors (not SDP related)?
if not:
registerEvents is not called in P2PController.js -> init() is not called in P2PController.js -> radio('resourceInit').subscribe is not called in sfClient.js ->
in updateMetadata:function() in sfClient.js there are 2 cases one when filesystem isn't used (incognito) and one when it is used (normal) maybe you're getting a silent failover there. Are you getting a warning "couldn't change resource Id".?

mazkopolo · 2013-10-15T21:37:04Z

I don't get "couldn't change resource Id" warning. "blockMap.changeResourceId(metadata.swarmId,function(succ)" is called when you upload a file (sender side). in both modes the following is executed:
if(succ){radio('resourceInit').broadcast(metadata);}

mazkopolo · 2013-10-16T01:56:54Z

I compared the client.js from sharefest.me (shown in red colour) with client.js from my server @ webrtc.thehii.com (shown in green colour). Would that help to identify the problem?

mazkopolo · 2013-10-16T03:31:48Z

Or could be because of the FileSystem issue? the last line I get from my server is: "finished initiating from filesystem"
Please take a look at attached image. Any idea?

shacharz · 2013-10-16T06:54:31Z

if(succ){radio('resourceInit').broadcast(metadata);}

So what is the reason for:

"registerEvents:function" ... "core/controllers/P2PController.js" are never get called in the Normal mode.

The screen shot indicates that there is no problem with the connection making it self. It looks like it doesn't even start.

mazkopolo · 2013-10-16T06:58:52Z

Yeah, you are right. Could it be because some FileSystem issue with Chrome?
I checked your codes and I can confirm that you have used the latest webkit filesystem instructions.

shacharz · 2013-10-16T07:03:00Z

Can't tell, I can't reproduce it and it needs to be debugged line by line...
I'd start with why the registerEvents isn't called in the P2PController

mazkopolo · 2013-10-16T07:04:49Z

Is there any way I can read the working code on sharefest.me and compare it with the one in github?

shacharz · 2013-10-16T07:06:41Z

It should be very similar, but currently I don't have time to push it.
you can deuglify it...

On Wed, Oct 16, 2013 at 10:04 AM, Maziar Janbeglou <[email protected]

wrote:

Is there any way I can read the working code on sharefest.me and compare
it with the one in github?

—
Reply to this email directly or view it on GitHubhttps://github.com//issues/59#issuecomment-26396901
.

mazkopolo · 2013-10-16T07:12:38Z

I posted a comment in here today about the differences of these codes. I am pretty sure they are very similar, however there are some small differences that make the code on github does not work properly.

Belial-public · 2013-10-24T10:35:37Z

I confirm the problem.
I'll patiently wait for a commit

cdonito · 2014-02-05T17:17:29Z

Hi,
I have the same problem as mazkopolo. does anyone find a solution ?

Thanks

dpavliuchenko · 2014-06-06T23:06:31Z

I have the same issue as mazkopolo.
I've tried with Chrome v35 on both machines as well as with Canary v37, it does not work in Incognito mode too. But works perfectly for Firefox.
Looks like the source code is not uptodate on git.
shacharz, could you please update it?

Eccenux · 2014-07-06T08:25:19Z

@mazkopolo From your screens I gather that you don't have a certificate for HTTPS. WebRTC don't work on insecure connections and HTTPS without certificate is insecure. The only exception seem to be localhost which works over HTTP.

shaharmor removed the question label Apr 21, 2016

Infinity Years Left in Google Chrome #59

Infinity Years Left in Google Chrome #59

Comments

mazkopolo commented Sep 27, 2013

shacharz commented Sep 28, 2013

mazkopolo commented Sep 29, 2013

shacharz commented Sep 30, 2013

mazkopolo commented Sep 30, 2013

mazkopolo commented Oct 1, 2013

shacharz commented Oct 1, 2013

mazkopolo commented Oct 1, 2013

mazkopolo commented Oct 2, 2013

mazkopolo commented Oct 3, 2013

shacharz commented Oct 3, 2013

mazkopolo commented Oct 4, 2013

and did the following settings in /core/apiValidators/DataChannelsValidator.js:

and the following in ClientConfig.js

};

shacharz commented Oct 4, 2013

mazkopolo commented Oct 6, 2013

mazkopolo commented Oct 7, 2013

mazkopolo commented Oct 8, 2013

My main question is why there are two places to put the STUN configurations in this project. One in /core/apiValidators/DataChannelsValidator.js

and the other one in sharefest/clientConfig.js:

STUN_SERVERS:['stun.l.google.com:19302'],

mazkopolo commented Oct 9, 2013

shacharz commented Oct 9, 2013

mazkopolo commented Oct 10, 2013

mazkopolo commented Oct 11, 2013

mazkopolo commented Oct 12, 2013

shacharz commented Oct 12, 2013

mazkopolo commented Oct 14, 2013

mazkopolo commented Oct 15, 2013

shacharz commented Oct 15, 2013

mazkopolo commented Oct 15, 2013

mazkopolo commented Oct 16, 2013

mazkopolo commented Oct 16, 2013

shacharz commented Oct 16, 2013

mazkopolo commented Oct 16, 2013

shacharz commented Oct 16, 2013

mazkopolo commented Oct 16, 2013

shacharz commented Oct 16, 2013

mazkopolo commented Oct 16, 2013

Belial-public commented Oct 24, 2013

cdonito commented Feb 5, 2014

dpavliuchenko commented Jun 6, 2014

Eccenux commented Jul 6, 2014