sip ss7

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 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

Release date: 2011-12-08Updated on: 2011-12-09 Affected Systems:Asterisk 1.xDescription:--------------------------------------------------------------------------------Bugtraq id: 50989 Asterisk is a free and open-source software that enables the Telephone User Switch (PBX) function. Asterisk has a security vulnerability in implementation. Attackers can exploit this vulnerability to cause invalid memory locations for server reference and DOS. When the "automon" feature in features. conf is e

Modification Purpose: If user-agent with GIT version information, it is easy to be caught by a version of the vulnerability targeted attack.Examples are as follows:sip/2.0 tryingvia:sip/2.0/udp 192.168.5.218:5060;rport=5060;branch= Z9hg4bk--106273027814628634511462861243from: Modification Method:In the corresponding Sofia profileAdd to It can be achieved.　　Modify the User-agent name in the SIP protocol