sip proxy

Reprint: http://www.cnblogs.com/ishangs/p/3816689.htmlApplication of Stun/turn/ice protocol in peer-to sip (II.)1 description2 dozen holes and the concept of crossing ... 13 hitting holes and crossing ... 24 using the STUN Series Protocol traversal features ... 25 Stun/turn/ice The relationship of the agreement ... 36 Stun protocol (RFC 5389) 36.1 Why the STUN protocol is used ... 3How the 6.2 stun protocol works ... 47 Turn protocol ... 47.1 Why the

Basic settings of SIP Trunk in trixbox Http://www.voclub.net/zone? Action-viewthread-tid-1065The basic settings of the SIP Trunk in trixbox are as follows: the extension can call the phone number through a SIP Trunk, and then ring the trunk number to the extension. Create a new SIP Trunk, provided that you have obtain

IntroducedThis article is a reference guide for SIP4.18. SIP is a python tool used to automatically generate Python bindings to C/C + + libraries. SIP was originally developed in 1998 with PYQT for Python and QT GUI Toolkit, but is suitable for generating bindings for any C or C + + libraries.This version of the SIP-generated bindings is available for Python vers

global. This means that the URI contained in the Contact header field is the UA that can receive the request. Supported and Require,proxy-require If UAC supports the SIP extension that the service side responds to the corresponding request, UAC should include a supported header domain description in the request when the options tags describe those sip extensio

I created a freeswitch kernel research and exchange group,45211986. welcome to join us. In addition, we accept the SIP-based communication server and client solutions. Freeswitch supports cross-platform, and the underlying library uses Apache Portable Runtime lib (APR). The author has been familiar with Apache in his previous work.In swith_apr.c, all the tool library interface functions provided by APR are referenced by freeswitch for the purpose

Tags: blog http os ar using file data on artOriginal: Asterisk real-time add SIP number--sqliteAsterisk real-time add SIP number--sqliteToday, I tried to use Asterisk's real-time mode to add a SIP account to SQLite without restartingAsterisk, no need to reload, you can successfully register a SIP account, the following

In many popular network technologies, we can see the content of the SIP protocol. So many people will doubt what the SIP protocol is? Let's take a look at it. SIPSession Initiation Protocol) is an application layer control Protocol defined by IETF Based on IP. because SIP is a plain text-based signaling protocol, it can manage meetings on different access network

Translated from:Http://developer.android.com/guide/topics/connectivity/sip.html Android provides APIs that support the Session Initiation Protocol (SIP. This allows you to add the SIP-based Internet phone function to your application.Program. Android contains a complete SIP protocol stack and integrates the call management service, so that applications can easi

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

variety of digital content and video services, thus increasing revenue. The WebLogic SIP server provides required call establishment and session management. However, carriers still need advanced media processing capabilities and standard interfaces that can interwork with BEA solutions, this allows carriers to provide scalable voice, video, and multimedia services. They also need a media gateway for cross-network connection. Cantata works with BEA Sy

Strict routing and loose Routing 1. The address list of a loose route does not list a complete and strict path, but only provides some key points in the path..You can use the automatic route selection function of the vro to route data between key points. data packets must also be copied during data packet sharding. In a SIP message, if the parameter in the first Route Header field contains the LR parameter, It is a loose route. 2. Strict routing re

We have already learned about the SIP protocol, which we will always see in the Unified Communication Platform and VOIP business. So here we will give you a brief discussion on the development of the SIP protocol and the SIP protocol stack. 1. About the SIP protocol Currently, there are three basic communication protoc

During our understanding of the SIP protocol stack, we found that the application of the SIP Protocol can be implemented on many platforms and devices. The following software is required for the establishment of the SIP Soft Phone development environment to develop a SIP-based soft phone on windows or linux platforms:

The development example of this blog post must be based on the previous SIP servlet development environment configuration, and if the SIP servlet development environment is not properly configured, refer to the SIP servlet development environment Configuration chapter.First, we create a dynamic WEB Project project called "Siptest", based on the method mentioned i

The pioneers in the converged communication field are currently sending the Session Initiation Protocol (SIP ). SIP has become the main protocol for VoIP and other real-time media communications on the packet network. There are several reasons for the popularity and success of SIP. First, SIP is regarded as a protocol

Http://www.christec.co.nz/blog/archives/42 Manage soft Input Panel (SIP) One final finishing touch to Windows Mobile applications is the proper handling of the software-based Input Panel (SIP), this can help differentiate your application, and thus your company as one which truely understands the Windows Mobile platform. The software-based Input Panel (also known as the popup or software keyboard) allows to

In asterisk, there are three types of peer: Peer, user, and friend.Let's take a look at the three types of VoIP-info. Peer: a sip entity to which asterisk sends CALS (a sip provider for example ). if you want a user (Extension) to have multiple phones, define an extension that CILS two sip peers. the peer authenticates at registration.User: a

Next Generation Network (NGN) and SIP protocol With the rapid development of mobile communication technology, we are brought into the colorful 3G multimedia Information age. In particular, the rapid development of the Internet, more and more users can use faster, cheaper Internet connection, which makes such as chat applications, video voice, online games, such as the need to continue online applications to achieve the possibility. The traditional te

Today, wireless service providers have been seeking cheaper voice services to maintain their market share. Obviously, the secret to surpassing competitors is to launch some attractive services with additional charges. However, existing technologies such as SMS need the same cost model as voice. Therefore, wireless service providers focus on the commercial technology of Session Initiation Protocol (SIP. SIP

People often ask if SIP uses HTTP as the underlying protocol. The answer is in the negative. SIP is a protocol that works with HTTP on the same layer (that is, the application layer), which uses TCP, UDP, or SCTP as the underlying protocol. However, there are many similarities between SIP and HTTP. For example, like HTTP, SIP