sip to isup

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

Background: Generally, when a UA receives a request that creates a dialog (invite/Subscribe/refer), it must decide whether to authorize the request, in some cases, UA determines whether to authenticate the request by determining whether the request is in a created dialog, For example, after invite creates a dialog, UA does not need to authenticate other requests (prack, Act, etc.) sent in this dialog, but the problem is that for refer, message, and SUBSCRIBE requests, A new dialog will be crea

Learn some of the FreeSWITCH core commands, and then learn more about FS in detail.To see if it was not previously suspected, two times when programming changes the configuration file, or Java injects some parameters into the configuration file, learn more about the following configuration file.This should be difficult, not clear.Ask Mr. Baidu.Learn a new knowledge of how FS adds recording functions to configureThe general telephone system can record voice calls in the system, and voice recordin

In sip, both re-invite and update are used to modify the session parameter. The difference is that update does not affect the status of the dialog, and re-invite changes the status of the dialog. Therefore, update can be sent before the first invite is responded (that is, before 200ok of invite is received ). That is to say, update can be used to control early media. The re-invite can only be sent after the first invite cup responds (that is, after th

InProgramWhen the MessageBox pop-up error message is applied, when you click the OK button, it is found that the keyboard icon on the mainmenu has disappeared. You must click it once to appear. Although it does not affect the use of normal functions, it may be confusing for PDA cainiao, and many methods to refresh the interface cannot be solved. Baidu cannot find the answer to this strange problem. You can directly query the problem on Google and find that foreigners also encounter this proble

A transaction is a request transaction sent by the customer (through the communication layer) to a server transaction, together with all the responses to the request of the server transaction, sent back to the client transaction. The transaction layer processes the re-sending of the application service layer, matches the response of the request, and the timeout of the application service layer. All tasks completed by a user agent client UAC are composed of a group of transactions. Generally, a

Based on the practices in the past few days, we have found an Optimal Configuration: 1. The SIP server uses trixbox. If you are familiar with Linux, we recommend that you use asterisk directly. 2 If the client is used directly, it is recommended that ekiga. By the way, how do I feel when using several clients: 1 Linphone: It seems to be well known. However, the latest version 3.1.2 crashes after being installed. I installed a general XP-SP3, compu

, has been completely attracted by WxPython , can not wait to practice, hereby record the individual learning process (Windows system):First, Wxpython environment installationThe most realistic and practical way to install Pip, and must be used in a manner that will synchronize the installation of dependent packages: pip install-u wxPythonsecond, Wxpython tastedAs a beginner, must not blindly directly into the subject, it is necessary to go through this stage, can be very good to help understand

versatile and most like relational database in a non-relational database. His support for the data structure is very loose, is similar to JSON Bson format, so can store more complex data types, he is mainly used to solve the massive data access efficiency problem. His storage seems to have a larger demand for disk space. The new version starts to support distributed. 4, Hypertable Hypertable and similar hbase are developed from Google's BigTable model, which is good for distributed support, bu

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

Kamailio is an open-source SIP server, formerly known as OpenSER. It runs C Programs on Linux/Unix platforms. It has good performance, flexibility and security. Weblinks · Homepagewithnewprojectname: http://www.kamailio.org · Home Kamailio is an open-source SIP server, formerly known as OpenSER. It runs C Programs on Linux/Unix platforms. It has good performance, flexibility and security. Web links · Home

From: http://brekeke-sip.com/bbs/viewtopic.php? P = 11824 SID = 1337c4d609517c9d1f0fcc5167d7d5a1 1) Please go to Ondo SIP Server admintool> [config] menu> [system].Set [Java VM arguments] =-xrs 2) If you are also using Ondo PBX, please go to Ondo PBX admintool> [Options] menu. Please find two [Java VM arguments] fields in the page.One in PBX system settings and one in media server system settings. Please set[Java VM arguments] =-xrs 3) please go to

response retransmits Timer E Initially T1 Non-invite request retransmit interval, UDP only Timer F 64 * T1 Non-invite transaction timeout Timer Timer g Initially T1 Invite response retransmit Interval Timer H 64 * T1 Wait time for ACK receept Timer I T4 for UDP Wait time for ACK retransmits Timer J 64 * T1 for ud Wait time for non-invite request retransmits Timer K T4 for UDP0 s for TCP/sctp

SDP application in the SIP protocol and SDPSIP Application The SDP is used to construct the message bodies of INVITE, 200OK, and ACK messages for the master and called users to exchange media information. 1. Media Stream Configuration (1) The description of the primary called media must correspond to the nth media stream (m =) of the primary called, and both contain a = rtpmap. this aims to adapt to the conversion from static Net Load types to dynamic

When you see this title, you can ask what is SIP (I have read my kids shoes from Windows Phone 7 tips series). Sip is called soft Input Panel, that is, the Input Keyboard In the touch screen. Windows Phone applicationProgramIn, you may encounter this situation, that is, after logging on to the interface, you need to automatically focus on the user name input box and pop up the keyboard to provide a good us

OverviewSIP defines two types of responses: temporary (Provisional) and final (final ).The final response transmits the request processing result, which is reliable ). The temporary response transmits the information of the processing process, which is unreliable by rfc3261.However, from the current situation, especially during the interaction with the PSTN, it is found that temporary responses should also be reliable.Rfc3262 defines an optional Extension Method for

============ Problem Description ============Not involved in audio, video send, as long as the implementation of registration, and chat function on the line, the online sipdroid source, but the configuration of the XML ============ Solution 1============9 is the Android 2.3 version, it should be very few machines are less than 2.3 of the bar, so this program can be installedAndroid platform based on the SIP protocol for registration, chat function

Release date: 2011-12-08Updated on: 2011-12-09 Affected Systems:Asterisk 1.xDescription:--------------------------------------------------------------------------------Bugtraq id: 50990 Asterisk is a free and open-source software that enables the Telephone User Switch (PBX) function. Asterisk has a security vulnerability. Attackers can exploit this vulnerability to obtain valid user names. When the regular, user/peer NAT sets different ports for responding to the request source port or the p