sip ss7

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SIP Protocol Resolution and implementation (C and C ++ use Osip) 6

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

Scheme of VoIP supporting Audio and Video Based on SIP

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

SIP 100rel Extension

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

Open-source sipimp Based on the SIP protocol

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

SIP registration Spoofing

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

SIP Experience Summary

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

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

Multi-channel stress testing program for SIP proxy

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 = "

Handy Color picker Sip on Mac

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

Asterisk Configure PSTN analog card to make the SIP soft phone call out outside by PSTN fixed telephone

============================================== 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 distributed

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

SS7 Basic Signal Unit format

The following excerpt from the "China domestic telephone network No.7 signal Mode technical Specifications (Interim provisions) gf001-9001". There are three basic signal cell formats, namely the message Signal Unit (msu:message Signal unit), the

Thread sip blocking Queue __java in JAVA

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

sip:180 Ringing vs 183 Session Progress

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

Jain API for call control and Wireless Networks

specialties to a single network structure, you can quickly create and deploy new services. Jain technology is being defined as a universal expansion of Java platform. According to Sun's Java program, the article of JSpA and JCP is being developed. Figure 1 Jain API structure Overview Jain advocates the following three main components of the network:Network Layer: Email: (high-level) Intelligent Network (AIN/In) or 7-message system with ISUP, INAP, and TCAP (

Asterisk SIP Channel Driver DoS Vulnerability

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

Automatic Registration of Sip

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

sip:180 ringing vs 183 Session Progress

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

Asterisk SIP Channel Driver Remote Crash Vulnerability

Release date: 2011-10-18Updated on: 2011-10-18 Affected Systems:Asterisk Open Source 10.xAsterisk Open Source 1.8.xDescription:--------------------------------------------------------------------------------Cve id: CVE-2011-4063 Asterisk is a free

Comprehensive application of 5-rtp packet Removal Process for SIP and RTP

The RTP receiving part is relatively simple (you do not need to consider jitterbuffer and so on). Start with here. In fact, there are three steps: 1. Create a UDP listener, such as 5200. 2. After receiving the RTP package, send it to the unpacking

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