Why does Sublime Text (2.0.2) connect to the Internet (and on port 80)? - sublimetext2

Why is my text editor, trying to make an insecure connection to the Internet? It happens when the application is opened. This is a screen shot from Little Snitch.

I just checked using Fiddler and as #Timeout suspected, it is checking for updates by submitting the following GET to www.sublimetext.com:
GET http://www.sublimetext.com/updates/2/stable/updatecheck?version=2221&platform=windows&arch=x64&r=1 HTTP/1.1
It responds with the latest version that is available:
{
"latest_version": 2221,
"update_url": "http://www.sublimetext.com/2"
}

Related

How to Fix ERR_HTTP2_INADEQUATE_TRANSPORT_SECURITY in IIS on Windows Server 2016

I'am recently upgraded an IIS web server to Windows Server 2016, and i have this error ERR_SPDY_INADEQUATE_TRANSPORT_SECURITYin Google Chrome.
This problem is happening because of the HTTP/2. This basically means that the site started a HTTP/2 connection but there was a blacklisted cypher negotiated. SO the browser has prevented the access to the website. So, the usual solution for this is to reorder the cypher suites to meet the requirements of the HTTP/2.
Another solution is to disable HTTP/2 and only use HTTP/1.1. This can be addressed on the server-side by setting the following registry keys and the restarting the host Windows server:
[HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\HTTP\Parameters]
"EnableHttp2Tls"=dword:00000000
"EnableHttp2Cleartext"=dword:00000000
You can downgrade kestrel to start with Http1. There seems to be some problem with new versions which starts kestrel on Http2.
"Kestrel": {
"EndpointDefaults": {
"Protocols": "Http1"
}
}
add above option in appsettings.json for temporary resolution
Below is the workaround to disable http2 in windows 2016 server.
Run Regedit -> Navigate to the HKEY_LOCAL_MACHINE\System\CurrentControlSet\Services\HTTP\Parameters
Under the Parameters folder, right-click the white-space, add 2 new DWORD (32-bit) values:
EnableHttp2Tls
EnableHttp2Cleartext
Ensure both new values have been set to 0(disabled)
Restart the OS.
the solution provided by Ding Peng works for me by updating the regedit
[HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\HTTP\Parameters]
"EnableHttp2Tls"=dword:00000000
"EnableHttp2Cleartext"=dword:00000000
#Anand Shukla have the correct solution, worked for me. I had to add additional braces to get it work.
{
"Kestrel": {
"EndpointDefaults": {
"Protocols": "Http1"
}
}
}

How do I access a github.com/lucas-clemente/quic-go server from Chrome?

when I ran quic-go-0.18.1(https://github.com/lucas-clemente/quic-go) with command: go run main.go -bind=127.0.0.1:443 -qlog -tcp in dir /example.
I use the client to visit https://127.0.0.1/demo/tile, I could get response:(Proto:"HTTP/3"),
when I use chrome and chrome beta to visit https://127.0.0.1/demo/tile, I find chrome only use http/1.1 to visit my server.
My Question is : How I can use chrome to visit https://127.0.0.1/demo/tile through HTTP/3?
My environment: Ubuntu 20.04 LTS
Chrome switch on enable-quic.
GO version: 15
reading https://www.chromium.org/quic/quic-faq it says:
How do I aim Chrome at the test server? If you have an HTTP server,
you'll need it to emit a response header that looks like:
Alternate-Protocol: quic:< QUIC server port >
Then you can just run chrome as usual and it will automatically start
using QUIC.
If you're testing only with the toy quic server, you can do something
like:
% chrome --disable-setuid-sandbox --enable-quic
--origin-to-force-quic-on=localhost:6121 http://localhost:6121/
If you need help troubleshooting, try running the QUIC server with
--v=1 or check out playing-with-quic
does it solve your issue ? Please report for others.

How to use self signed certificat with Postman

I'm using Postman to test my API.
It works fine in dev env as I'm calling my server in http.
However on the acceptance testing platform, the server is in https and we are using self signed certificat.
When I try to make the call via Postman I get "Could not get any response" and it's written underneath that SSL connections are being blocked, with a link to this page.
However as notified in the comment, this does not work anymore for self signed ceritifcate and the latest chrome release.
For info, I'm using Version 59.0.3071.115 (Build officiel) (64 bits)
When I go into google settings, then certificates : chrome://settings/certificates
And I try to import an authority certificat, I get an error popin that says that my authority is not a certificat authority and it won't allow me to import it.
As I'm on linux, I tried to do what is detailed there : https://superuser.com/questions/1201552/not-a-certification-authority-while-importing-self-signed-certificate
But even though I see my certificat when I call
certutil -d sql:$HOME/.pki/nssdb -L
It still won't allow me to import it in google.
So how can one use self signed certificat under linux with Postman ?
EDIT : I was using the version from Chrome web store which is lacking SSL support. Downloaded the standalone and as #Pratik Mandrekar wrote it works fine!
You can disable SSL certificate verification in Postman. Settings->SSL certificate verification as shown below.

WebRtc2SIP: No video is been received/transmitted when made call between chrome and a SIP client

I am a newbie to webrtc2sip. I have setup my webrtc2sip gateway and registered to sip2sip.info as my domain. The problem is when I make video calls from chrome to any SIP client(ekiga/jitsi) the call gets connected but I am unable to see videos on both the sides.
==================================================================================
Case 1: Chrome calls SIP client
Result: No video shown on both transmit and receive side
==================================================================================
On the chrome JS console it says that :
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx SIPml-api.js?svn=179:1
==session event = m_stream_video_local_added SIPml-api.js?svn=179:1
==session event = m_stream_video_remote_added SIPml-api.js?svn=179:1
==session event = m_stream_audio_local_added SIPml-api.js?svn=179:1
==session event = m_stream_audio_remote_added SIPml-api.js?svn=179:1
I have attached the JS console logs(case1_web2SIPClient_JSLogs.txt), wireshark trace(case1_web2SIPClient_WStrace.pcap) , webrtc2sip gateway console logs(case1_web2SIPClient_gatewayLogs.txt), sipml5 expert settings (Expert_settings.png) and config.xml (config.xml) for this case. I did not change anything in the config.xml that was generated after i built the source as mentioned in the instructions of this page (http://linux.autostatic.com/installing-webrtc2sip-on-ubuntu-1204).
I gave a try making calls between chrome and a android SIP client (CSipSimple) and the problem remains the same.
==================================================================================
case 2: SIP client calling chrome.
Result: as soon as I click answer button on chrome, the calls gets rejected.
==================================================================================
The JS console logs states that:
State machine: tsip_transac_ist_Proceeding_2_Completed_X_300_to_699 SIPml-api.js?svn=179:1
SEND: SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 172.21.128.118:10060;rport=10060;branch=z9hG4bK-1441398960
From: <sip:tata#172.21.229.127>;tag=300647977
To: <sip:amshyam320#sip2sip.info>;tag=ZxQFfM7fIIP3rT1HINzb
Call-ID: fbdf5a11-ff9e-0072-fa8b-09525220cec6
CSeq: 1670757835 INVITE
Content-Length: 0
Reason: SIP; cause=603; text="Failed to get local SDP"
For this case I am attaching JS logs(case2_SIPClient2WebJSLogs.txt), wireshark dump(case2_jitsiToWeb_WStrace.pcap)
Configuration:
Chrome Version: checked on 30.0.1599.114 and even on Latest chrome version
Webrtc2sip version: 2.6.0
sipml5 Version: svn=203
ubuntu version: 12.04 (checked on both desktop and server editions)
Am I missing something in my setup or configuration please guide and help in moving further.
Thanks,
Shyam
Case2:
You're using RTCWeb-capable browser(Chrome) and trying to call a SIP client which may not be implementing some mandatory features like ICE,SRTP. Chrome uses SRTP-SDES and Firefox uses SRTP-DTLS.
Enable RTCWeb Breaker in sipml5 expert settings and check.
The RTCWeb Breaker is used to enable audio and video transcoding when the endpoints do not support the same codecs or the remote server is not RTCWeb-compliant.
Case:1:
Is audio working? and I can't see your logs.

Is it possible to create a browser extension to allow for incoming requests?

I am looking to figure out a way to make incoming request to a browser. Installing an extension in the browser is OK. The goal of this is to allow another machine to connect to the extension to control a game without needing an intermediary server.
Is this feasible? Is it possible to make a Chrome or Firefox extension open a port to allow for incoming request?
What you are asking for are server sockets. For Chrome the answer is "no", Chrome extensions can only open client connections. Firefox extensions on the other hand can use nsIServerSocket interface to listen for incoming TCP connections on a port. If you use the Add-on SDK you would need to use the chrome package. Something like this:
var {Cc, Ci} = require("chrome");
var socket = Cc["#mozilla.org/network/server-socket;1"]
.createInstance(Ci.nsIServerSocket);
socket.init(12345, false, -1);
socket.asyncListen({
onSocketAccepted: function(socket, transport)
{
...
},
onStopListening: function(socket, status)
{
}
});