sip to isup

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

implement stateful elements according to the processing process mentioned in [4], and try each address until it is connected to the server. Each connection attempt starts a new transaction. Therefore, the values at the top of the Via header domain for each connection attempt are different, and a new branch parameter is added. In addition, the transport value in the via header domain is set to the value of the target server of the request. Process Response The response is first processed by the

Some time ago, I was dragged to. Net for two months and worked overtime every day. Alas, This Is What outsourcing companies do. Now you don't have to work overtime. You can study the audio and video communication based on SIP. After studying blogs written by others for a few days, we have a preliminary solution: Server: opensips Client audio decoding: speex Video decoding: Library H.264 in FFMPEG The above two libraries are all C libraries that

The flexibility of the SIP protocol brings about compatibility issues. Recently, we encountered the compatibility of the Client Server caused by the 100rel extension. The RFC for this extension is 3262. Http://www.ietf.org/rfc/rfc3262.txt This extension defines reliable Response Processing for temporary responses, but is not supported by some servers or clients. Although this function is performed through negotiation, however, the call fails due to

Sipimp This is a project to develop a SIP based simple compliant client for instant messaging and presence. initially there are UNIX (Yes Linux too) and Windows versions. it supports advanced security with S/MIME and TLS. Development Status: 3-alpha License: vovida software license 1.0 Operating System: 32-bit MS windows (95/98), all 32-bit MS windows (95/98/NT/2000/XP) Programming Language: C ++ Topic: chat User Interface: Win32 (MS

In the previous article, we talked about sip registration, but not everyone can register. If a normally used user alice@sip.com is already registered, an invalid user intercepts Alice's information. Modify the to domain and use the address of the physical device as the contact address. Isn't that a big error? All the information is sent to your device. Therefore, you must have an authentication mechanism in the registration server to ensure that the r

The application of any knowledge must be based on long-term accumulation to achieve true accumulation.Source codeThe research helps you complete the entire C language style.CodeIn the face of bugs, especially memory leaks, the dynamic memory generated by osip_xxx_init will definitely cause you a headache.ProgramThe final fall. Of course, errors similar to osip_cal1__set_number ( callid, "2") also introduce deep-level bugs. Selecting Osip as a client will be the right choice, simple, just simpl

Create a sip environment in Windows Step 1 software preparation:Server Software ndosip server: http://www.brekeke.com/This software is written in Java. Therefore, install JDK before installation.Take sipphone as an example. Step 2: Install1. install ondosip server first. set a port during installation. this port is used to manage an http port of the server. therefore, you can define it yourself. set it to 8080. the installation type is for education

This stress test is based on the basiccall Project Modification of resiprocate 1.5 and passes the resiprocate 1.5 proxy test. In theory, it should also be able to perform stress tests on other proxies. How to Use 1. Download resiprocate 1.5 2. Slave nodes/Resiprocate-1.5/resip/dum/test 3. Compile and run the basiccall Project Notes: 1. If you use the resiprocate proxy, pay attention to several parameters. Char * domains = "192.168.1.101 "; Char * interfaces = "

There is always a lot of things, you just look at a glance has been fascinated. Measures have such an app to do is really the heart of the very, eye-pleasing, triggering the beauty of your inner impulse. Let's leave one, memo.It's so delicate.Simple operation, the upper left corner of the heart of the click can be anywhere to take color, the color point will zoom in.The color of the heart in the upper right corner, is the system of color board. Each color finish is automatically added to the Liu

============================================== View hardware configuration # Dahdi_hardware==============================================View Dahdi Service ConfigurationMore/etc/dahdi/system.confShow the following content, obviously less my PSTN card configuration# Global DataLoadzone = usDefaultzone = usRebuilding the Dahdi service configuration#dahdi_genconfView Dahdi Service Configuration again# more/etc/dahdi/system.confShow# autogenerated by/usr/sbin/dahdi_genconf on Wed Aug 15 22:09:20 201

SIP Key-value Database (iii)--MONGODB distributedThe random Read and write performance of a single-machine MongoDB is tested, and this section is about MongoDB distribution.MongoDB is distributed into two types, one is replication, and the other is sharding. We mainly look at sharding.Put a structure first:MongoDB auto-sharding configuration is very simple, in different machines to open Shard, config server, MONGOs process can be. (assuming that the c

Document directory Jain proposal SIP, ISUP, call control system, and Jain Interface Application of Jain APIs to Mobile Networks Mobile Station No-wire access to network (RAN) Network and Enterprise Internal capacity and service End-to-end structure Jain API for call control and Wireless Networks Face-to-Face Jain API of the integrated network connects the business agility, network convergen

Comparison between H.323 and SIP Currently, 3GPP uses SIP as the core protocol of the third-generation mobile communication network. The NetMeeting protocol in Windows XP is also changed from H.323 to the SIP protocol. Considering its business flexibility, the SIP protocol will become the future development direction.

In terms of routes and switches, we have learned a lot about them. Now, let's take a look at the content about the softswitch protocol. Currently, 3GPP uses SIP as the core protocol of the third-generation mobile communication network. In Windows XP, NetMeeting's Softswitch protocol is also changed from H.323 to the SIP protocol. Considering its business flexibility, the

Recently in the study of Java with the JDK and the contract, Java.util.concurrent, Discovery is very powerful, one of which is the work of many times the use of threading tool class Blockingqueue. During the actual development work and interview

If You know this phone is ringing (an ALERT q.931 message, for instance) you send a 180 ringing. If You receive a notification indicating this call is progressing, but you don't know for sure whether the user I s being alerted or not, your send a 183

Release date: 2012-04-23Updated on: 2012-04-24 Affected Systems:Asterisk 10.xAsterisk 1.xUnaffected system:Asterisk 10.3.1Asterisk 1.8.11.1Description:--------------------------------------------------------------------------------Bugtraq id: 53205

SipManager: setautoregisterandpolicy () --> SipService: openpolicysession () --> SipSessionGroupExt: opentoreceivecballs () opentoreceivenovel () --> AutoRegistrationProcess: start () --> first, perform the anti-registration (duration = 0)