Infinity Years Left in Google Chrome

[mazkopolo]

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]

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]

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]

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]

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]

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]

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]

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]

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]

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]

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]

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:["mcd@50.112.158.192:3478"],
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]

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]

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]

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]

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