softphone sip

This digest from: http://blog.chinaunix.net/uid-20655530-id-1589483.htmlThe quickest way to learn the SIP protocol is to learn by example,Found a complete call flow, let's go!INVITEThe calling party Tesla First initiates the INVITE message to the called party Marconi. The INVITE message contains the session type and the parameters necessary for some calls. The session type may be simple speech, or it may be a multimedia video used by a web conferenci

Previously, we handled some application problems related to the SIP Session Initiation Protocol. Many solutions are summarized. I wonder if you have mastered it. See Basic Q A of the SIP Session Initiation Protocol. Let's add more content in this article. Is X standard supported? We have implemented and tested a large number of standards. We listed these standards in the Product Information Center (see ref

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

SIP Response Codes SIP/2.0 allows six values for the "the" I digit: From the SIP RFC: The response codes are consistent with, and extend, http/1.1 response. Not all http/1.1 responseCodes are appropriate, and only those that are appropriate are. Other http/1.1 response codesSHOULD is not used. Also, SIP defines a new c

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

With the rise and development of information services and applications, the data traffic on IP networks is growing rapidly. Data Services must follow certain protocols. Currently, H323, SIP, and MGCP are mostly used. These protocols have their own characteristics and have been promoted to a certain extent. This article explores how to interwork between SIP and MGCP. Logical Structure Figu

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

Kamailio is an open-source SIP server, formerly known as openser. Kamailio is an open source, gpl2, SIP Server Routing platform. It is written in C for Linux/Unix plaforms and focuses on performance, flexibility and security. On Nov 04,200 8, kamailio and SIP Express Router have started the SIP router project. Web Link

BICC vs sip-Comparison of NGN Protocol -------------------------------------------------------------------------------- One notable feature of the Next Generation Network Based on Softswitch is the separation of Call Control and bearer control. In terms of call and control of communication, from the Standard Research of the International Telecommunication Union (ITU-T) and Internet Engineering Task Group (IETF), there are two protocols that deserve

Abstract: IP MultimediaSubsystem ( IMSThe user plane, control plane, and business plane are separated to facilitate the provision of services to users. This article will focus on analysis SIPApplication Server(SIP as) Issues related to the IMS service provision process, including the general process, billing functions at the business layer, and technical difficulties provided by the Service, analytics is based on a specific service (Message Service) p

SIP Message Type and Message format SIP is a text-based protocol that uses the UTF-8 character set. There are two main types of SIP messages: One is the request sent from the client to the server ); One is the response message sent from the server to the client ). A basic SIP message includes the starting line, one or

When talking about the SIP Protocol, many people first think of the VoIP service. So what else is there? Today we will introduce another embodiment of the SIP business-Erlang project called yxa official website http://www.stacken.kth.se/project/yxa/), can be said to be the Erlang on the SIP protocol stack and server implementation. In addition to the basic

SIP protocol Learning 1The SIP protocol is an application layer control protocol proposed by IETF for multimedia communication over an IP network. A hierarchical method is used to create a service. A control protocol on the application layer is used to create, modify, and terminate multimedia session processes with multiple participants. Members participating in a session can communicate through multicast,

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

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

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

In The XML configuration file of jain sip proxy, The proxy needs to initialize it through The XML file. Therefore, we need to know a lot about this part of content. Let's take a look at the parameters of the SIP protocol stack you configured. So we will give a detailed explanation of this Part, and hope it will help you. SIP_STACK tag: this parameter is required. It defines the core parameters of the

In the previous article, we used the description of the SIP Routing Mechanism to understand the definition and concept of the SIP routing mechanism. Next, let's explain these abstract concepts to help you understand them. Next, we will use two SIP routing instances to help you understand these concepts. SIP route Examp

ubuntu9.10 Install OpenSER and use RTP Proxy to realize turn, solve the problem that symmetric NAT brings to SIP voice communication Reprinted from Link http://hi.baidu.com/zj8la8la/blog/item/d700d8b2c11a41abd9335af9.html Leave a note, convenient other people, online resources are not complete, I do a complete bar, at least I test through:My goal is very simple, just realize the SIP network MySQL database

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