After OS X upgrades to El Capitan, it provides a security-related pattern called SIP (System Integrity Protection), also known as rootless mode, which is a new feature that emphasizes security for OS X, which prohibits the software from being used as root in Mac running on, upgrade to OS X 10.11 Maybe you'll see some apps are disabled so that/usr/bin folders we can't read and write properly, but it also causes some programs (such as homebrew and Git)
"Python/c++ Interface Library comparison" (Swig,boost.python, Pycxx, py++, sip, Weave, Pyrex) http://blog.csdn.net/lainegates/article/details /19565823There are many open-source python/c++ binding tools, search a lot of 岾, here a little summary.SWIG
Supports Python 2 and 3
Properly configured, the package can be fully automated (*.i files need to be written by themselves)
When it is not fully automatic, it will mostly repeat your. h f
novice?The discovery directory has a lot of configuration files, is it necessary to change these configuration files two times? Let's take one to learn.FreeSWITCH is configured by default to 1000 to 1019 (ext.) A total of 20 users. Let's not start by downloading a SIP phone client on our own phone and try to talk.First Ipconfig/all know your LAN address. The password default appears to be 1234.My own IP is 192.168.0.113, and then I correspond with th
FreePBX SIP TrunkDockingbackground: PBX1 is a virtual machine running FreePBX, whichnow needs to be connected via SIP TRUNK docking , PBX2, using PBX2 E1 The line calls out the phone. PBX1 192.168.100.1PBX2 192.168.100.2PBX1on the configurationOneConfigurationTrunkNew SIP TRUNK650) this.width=650; "src=" http://s3.51cto.com/wyfs02/M01/54/0B/wKioL1R2mLHDypJCAACZ
You can use the method provided by. Net CF itself to enumerate all the SIP messages on the device. See: http://msdn.microsoft.com/en-us/library/ms172538.aspx
Code highlighting produced by Actipro CodeHighlighter (freeware)http://www.CodeHighlighter.com/-->
//
Define an inputpanel
Private
Inputpanel m_inputpanel
=
New
Inputpanel ();
//
Enumerative sip
Foreach
(Inputmethod Method
Csdn lidp http://blog.csdn.net/perfectpdl
The SIP response to the invite request may be final or temporary. The final response is always sent reliably, but not the temporary response. You can use the prack (temporary response confirmation) method to reliably send a temporary response.To develop applications that support prack, the following conditions must be met:
The client sending the invite request must put a 100rel tag in the supported or requ
1. The following is the test matrix 1 (the problem is not resolved ):
CalleeCaller
Jain-Sip-UA
X-Lite
IP-video
Jain-Sip-UA
It can communicate normally, but the number of frames is not enough and it is not smooth.
When the call is received, the xlite is dropped.
After the IP address is answered, "keep" is displayed, and UA audio and video can be received.
X-Lite
The network structure is as follows:Asterisk (192.168.1.99) That is to say, both Asterisk and SIP terminals are behind Nat.
The solution is as follows:1. Modify the SIP Extension settings in the SIP _. conf file.Nat = YesQualify = yes; it seems this item is not requiredExternip = 55.66.77.88; change to match our external IP addressLocalnet = 192.168.1.0/255.255.
This article from csdn ucser, http://blog.csdn.net/perfectpdl reprinted to indicate the source, thank you!
I have created a freeswitch learning and communication group, 45211986. welcome to join.
Freeswtich can be used as the rtmp and SIP gateway of the Streaming Media Protocol. It can communicate with the SIP video phone through flash in a web browser. This function can be used on the browser side f
Currently, the main reason for the combination of the codec and call control of the SIP protocol stack is the reuse of larger particles. In this case, the SIP is too bloated. It is not easy to expand.What I want to consider now is to break down the SIP library into two databases, one responsible for codec, and decoded by the
1. Authentication and encryptionThe role of certification (Authorization) is to show who you are, to prove to others who you are. The associated concept is MD5, which is used for authentication security. Note that MD5 is just a hash function and is not used for encryption. Because the hash function after processing the data can not reverse recovery, this way others can not steal your authentication identity password.Encryption (encryption) is the role of the data to be transferred to the process
In many cases, the SIP does not go directly to the target host, but goes through many intermediate node servers. In the request message, the Via header field indicates the nodes that have passed through (each node passes through, add a via header). In the response message, the via header field indicates the node that the message will go through next (each time the request is returned from the original path, a via header is deleted from each node ).
T
Prack English translation (the provisional Response acknowledgement), you can call IT security information! This compares the image.The final response in the SIP is understood to be reliably transmitted, such as a 200OK response to the invite, and UAC will give an ACK telling UAS that it has received 200OK. The reliability between 200 and ACK is end-to-end. Prack is a mechanism for guaranteeing the reliable transmission of temporary messages (101-199)
In PPC development, you sometimes need to hide the sip. There are many ways to hide the SIP in Windows Mobile 5.0. The following are several methods:
1. shsippreference (m_hwnd, sip_down );
2. sipinfo Si;
Memset ( Si, sizeof (SI ));
Shsipinfo (spi_getsipinfo, 0, Si, 0 );
Si. fdwflags = ~ Sipf_on;
Shsipinfo (spi_setsipinfo, 0, Si, 0 );
3. shfullscreen (hdlg, shfs_showtaskbar, shfs_hidesipbutton );
In Windows Phone, when the input box gets the focus, the soft Input Panel (SIP) is automatically displayed for the user to input. When we click the physical rollback key, the SIP will be automatically hidden. So what event is triggered? Where should we write code for other operations?
We can register the keyup event in the input box. When the input box obtains the focus and click the back button, the syste
In my previous article, I introduced how to use a SIP Soft Phone for direct calls. However, if many other users need to talk to each other, at the same time, authentication and control of User Account logon is required based on security considerations, in these cases, you need to establish a sipserver.Role of sipserver:Call control and processing functions, business provision/support functions, user management functions, protocol processing functions,
From
The From header field contains the logical flags of the requesting initiator, which may be the user's Address-of-record. Just like the To header field, the From header field also contains a URI and can contain a displayed name (SIP display info).
To
The To Header field is the first and also the "logical" receiving place that specifies the request first ("First" is because it may refer to another receiving place),
Or the Address-of-record of the
I began to study the VOIP/SIP agreement from 09, open source project also saw a few, the earliest Pjsip 05 began to push the time, began to pay attention to, also in their own winmobile project used. Later also saw Sipdroid,imsdroid (Doubango), Linphone,csipsimple (PJSIP).I think the best advantage of Linphone and Csipsimple,linphone is the full platform support, Android,ios,winphone,windows,linux,mac osx,web all support, but the quality is still unde
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