I 'm working on esp8266 module these days.
The firmware on this module is nodemcu 0.9.6 and its language is lua.
I use mqtt protocol for my project and it works well.But TLS is necessary.
I don't know how to set tls on this firmware. I 've searched but documents about TLS on nodemcu website is not enough for me.
I know that I should use this instruction at connection time to broker:
m:connect(BROKER, 8883, 1 , function ( connection ) ... end )
and to put the CA file on module i should use something like this:
enter image description here
that 's what i read from nodemcu website --> https://nodemcu.readthedocs.io/en/master/en/modules/net/#netcertverify
but the problem is when i use this on my code the module doesn't understand it and gives error at upload time and runtime.
is this problem because of the version of my firmware? Upgrading will make it? Or I am just doing something wrong on my code?
I haven't seen sample code about putting CA file on module in other project so I guess my work is wrong.
if you have worked on TLS ( on nodemcu firmware ) I would be happy by your helps.
Build a recent firmware with SSL/TLS enabled.
Flash the binary to upgrade. Make sure all flashing parameters are exactly as documented and include the init data binary.
The rest of your code seems fine.
Take a look at this question and read through comments of the answer.
You may have to get esp_init_data_default.bin from latest sdk.
I used esptool.
esptool.py --port <PORT> write_flash -fm dio -fs 32m 0x00000 <BIN_PATH_AND_FILENAME> 0x3fc000 <PATH\>esp_init_data_default.bin
Please do upvote this post if you find it is useful. Thumbs up for #MarcelStör.
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Since Twitter changed their website design, I cannot get a set of tweets from any account by using built-in Zinc classes. It throws an error that says: ConnectionClosed: Connection closed while waiting for data
I am using Pharo 5, and I don't know how to tweak the ZnClient settings in order to keep the connection open or something to the purpose of getting the data.
testTwitter
| client |
self ensureSocketStreamFactory.
self isNativeSSLPluginPresent ifFalse: [ ^ self ].
(client := ZnClient new)
get: 'https://www.twitter.com/pharoproject'.
self assert: client isSuccess.
self assert: (client contents includesSubstring: 'Twitter').
client close
That's the test I have in place, it never passes, and throws the error mentioned above. What's missing here? I did a Ruby script using open-uri, openssl and Nokogiri and it fetched the tweets just fine. Perhaps it's a problem with the SSL connection itself?
The issue here is quite easy to answer, but you won't like it. Your issue is connected to the fact that the Twitter has deprecated support for TLS 1.0, TLS 1.1 on July 15/2019. Your pharo is using the deprecated TLS to connect. That is the reason why you are getting the timeout.
The solution?
You have to compile the new SSL/TLS support yourself which is not an easy task to do. You have to compile in at least TLS 1.2 to be able to connect again. There is lack of Pharo documentation how to compile support for new libraries. My guess is that you are using TLS 1.0 (see a note below) - since Pharo 6.1 (so your Pharo 5.x will have same or older libraries) has libgit2.so compiled against libssl.so.1.0.0 (which has dependency libcurl-gnutls.so.4) - If you update the libraries you can see that those support >= TLS 1.2.
Note:
This is connected to the issue which I have posted some time ago. Nobody upvoted it or answered so it got automatically deleted - you can vote to undelete it: https://stackoverflow.com/questions/51399321/getting-error-when-adding-ossubprocess-to-my-pharo-6-1-on-centos-7-4x (see the bottom of the post for the question). I don't have an answer for that as I have dedicated time to my Smalltalk/X project.
Or just switch to a newer Pharo. Adding your method to ZnHTTPSTest in Pharo 8 just works (tested on Pharo 8 build 686, Ubuntu 18.04.02 LTS with the stable vm in PharoLauncher)
I tried posting this on ask.openstack but it has been stuck in the moderator for 5 days now. I thought I'd try here.
I was trying to debug a Nova issue and wanted to decode the SSL / TLS packets being exchanged using Wireshark. Part of the changes I was making was setting Nova up to use SSL / TLS and I wanted to be sure that part of it I had set correctly. I eventually figure out my issues from the various log files but I'm somewhat assuming that being able to watch the network traffic may help in some very difficult cases.
The exchange uses TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384 at one point. According to this security stackexchannge question, there is a "pre-master secret" or various other terms. I've wrestled with this before in a previous life doing IPSec. Usually you can set debug in the application and it will spew out the secret into the log file. I tried "debug = true" under Default in nova.conf and got lots of debug but no secret. There was two items that looked interesting that were reported as **** in the log: keystone_authtoken.memcache_secret_key and neutron.metadata_proxy_shared_secret. I wasn't sure if those were the secrets I was looking for or not. In this case, I'm looking at the nova-api traffic going to port 8774.
Also, since all of openstack is Python and uses the same "request" and "certifi" packages, it may be possible to generalize this to all of the openstack components.
nova --version report 9.1.1
I'm trying to get asterisk 11.20.0 running with WebRTC (sip.js 0.72 which I believe is a fork of jssip), but I'm seeing the following (and the called party rings, but when the phone is answered the call gets hung up).
This is my setup:
What I see:
In the CLI:
[2015-11-24 01:01:53] NOTICE[43619][C-00000002]: res_rtp_asterisk.c:4441 ast_rtp_read: Unknown RTP codec 95 received from '(null)'
In Firefox:
InvalidSessionDescriptionError: Invalid description, no ice-ufrag attribute
Attachments:
SIP Dialogue (Asterisk CLI)
Webphone Log
Config Files (httpd.conf, sip.conf, rtp.conf)
Asterisk Compiled with Libuuid & Friends
What I've tried so far:
Changed webRTC implementations (tried chrome and firefox both with SIPML and SIP.JS)
Set the STUN server to null on the client side (stunServers: ['stun:null'])
Configured properly (I hope) my sip.conf and rtp.conf and httpd.conf
Made sure I have libuuid, uuid and their -devel companions and after i've recompiled asterisk.
What I've read:
http://forums.asterisk.org/viewtopic.php?p=201702
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
http://jssip.net/documentation/misc/interoperability/asterisk/
http://sipjs.com/guides/server-configuration/asterisk/
https://kunjans.wordpress.com/2015/01/09/web-sip-client-sipml5-with-asterisk-13-on-centos-6-6/
http://forums.digium.com/viewtopic.php?f=1&t=89798
Please, if you can, give me a hand. I'm about to smash my box with a sledge hammer.
Faced same issue and followed instruction in http://forums.digium.com/viewtopic.php?f=1&t=90167 realise that:
This issue is caused because you asterisk don't have ICE support, you can solve that by installing the uuid/libuuid and uuid-devel/libuuid-devel packages on your system. Then recompile asterisk(be sure to rerun the configure script before the make command).
I did recompile my Astersik 11.16.0 with patch for ECDH support and fallback to prime256v1 https://issues.asterisk.org/jira/browse/ASTERISK-25265 and looks like lost uuid support at that time. Reverting back to non-patched version (with uuid support, use to be compiled before) resolved my issue with "no ice-ufrag attribute" error in Firefox console and calls are going well now from WebRTC client SIPML5 based to asterisk, but not in opposit direction
I am trying to fetch a file over HTTPS in Io language:
url := URL with("https://api.example.com")
url fetch println
And I get this:
Error_0x7f97e1509a80:
location = "/opt/local/lib/io/addons/Socket/io/URL.io:232"
message = "Protocol 'https' unsupported"
I was trying to find something on the net, but, as everybody knows, it's not easy because of the name. I only found this thread http://tech.groups.yahoo.com/group/iolanguage/message/10898 but that's quite old.
How can I get the HTTPS support in Io?
EDIT
I've found that there is a SecureSocket addon, a wrapper over OpenSSL, in Io's source. It wasn't installed when I did sudo port io install on my MacBook with Mountain Lion, though. I tried building it from source, but no luck. It didn't build for me on a Linux machine, either.
EDIT2
I just tried to build Io from source (git clone https://github.com/stevedekorte/io.git) again (using the included script build.sh) and it turned out that cmake did detect OpenSSL:
-- Found OpenSSL: /usr/lib/libssl.dylib;/usr/lib/libcrypto.dylib
But then the SecureSocket addon is not built. Its readme file: https://github.com/stevedekorte/io/tree/master/addons/SecureSocket says:
The DTLS1 bindings are not usable unless the patches in this file are
applied to OpenSSL 0.9.8e. However, this patch includes a
deactivation of the handshake retransmission code in d1_both.c,
making it unsuitable for production environments. I take no
responsibility, etc, etc. If you want to use it anyway, apply the
patches(gathered from various newsgroups and my own experimentation)
and uncomment the commented-out block of build.io. For what it's
worth, DTLS support in OpenSSL is new as of 0.9.8 and is pretty buggy
to begin with. It's a nice idea, but it doesn't seem to be
production ready at all yet. These bindings are no exception.
If you can't get io to do it your best option would be calling an external tool like wget or curl which can and then loading the file/result locally or returning it via a pipe.
For anybody else interested in another workaround, it should be possible to put stud in front of an Io program which will do the SSL stuff. I have not tested that myself yet.
stud - The Scalable TLS Unwrapping Daemon stud is a network proxy that
terminates TLS/SSL connections and forwards the unencrypted traffic to
some backend. It's designed to handle 10s of thousands of connections
efficiently on multicore machines.
I have tried for days now trying to find the right version of red5phone, but to no avail. Also, I need red5 and asterisk to be on different servers. I have followed all instructions described in the various tutorials on web, but nothing helps. I have downloaded 2 different versions of red5phone: sip1, sip_47 from the red5phone google code site, but none of them worked!
When I use:
a) sip1 - shows the correct parameters being passed on the red5 server console, but connection is stuck up with the console displaying the following error:
[NioProcessor-1] ERROR o.r.server.service.ServiceInvoker - Method login with parameters [<sip user>, <sip user>,<sip user>,<sip user pwd>, <asterisk server i/p>, <asterisk server i/p>] not found in org.red5.server.webapp.sip.Application#2d0c94a7
b) sip47 - when I type in the values in the flex interface and check the red5 server console, I see all parameters correct(the ones I passed) except for asterisk server ip that i m trying to connect. Instead, it shows 127.0.0.1 by default and completely ignores the passed ip and regsitration fails.
I am using:
red5 server version: 0.9.1, centos: 4.8(final), red5phone used: sip_47 (tried sip1 as well)
As a desperate measure I tried debugging the source code for red5phone (java and flex files) myself. but when I try to create the environment on my local system, I get several compile errors for missing java packages such as javax.media, org.slf4j, org.red5, etc. really confused and desperate for some guidance. Any tips highly appreciated.
Sunil, I'm also new to this, I would try red5-voicebridge installed in your red5 server.
red5-voicebridge Please let me know if you got it to work.