sip paging

Chapter 7 Registration Section 1 Overview SIP provides a capability to discover users. If a user wants to enable a session with another user, the SIP must be able to discover the current host of the target user. This discovery process is often used by SIP network elements, such as proxy servers or redirection servers. These servers can receive a request and detec

Chapter 3 SIP messages SIP is a text-based protocol that uses the UTF-8 character set (rfc2279 ). A sip message can be a request sent from a client to a server, or a response sent from a server to a client. Both requests (rfc3261 section 7.1) and responses (rfc3261 section 7.2) are in the format described in rfc2822. However, there are some differences between ch

What is VoIP? The complete name of VoIP is voice over Internet protocol, which is translated as Internet voice. It can be simply understood as a technology that uses an Internet system instead of a traditional telephone communication system for voice calls. In fact, the biggest difference between the two is that traditional voice calls use analog signal technology, which is prone to interference and difficult to avoid signal distortion, the capacity of traditional analog signal communication tec

SIP: Session Initiation ProtocolDocument StatusThis document has developed an Internet standard tracking protocol for Internet communications that asks for discussion and elevation of recommendations. Please refer to the "Official Internet Protocol Standard" for the standardized status of this agreement. Forward Unlimited!Copyright noticeCopyright (c) Internet community (2002). All rights Reserved!ProfileThis document describes the Session Initiation

What are the problems with SIP from private network to public network? Address translation of the package. SIP address Translation inside the SIP message. The RTP address translation in the SDP inside the SIP message. The existing structure of the network is complex, SIP s

Transferred from Http://www.ring180.com/sip/28-pstn-interworking/59-sipThursday January 2009 03:46 Sharp Voice Communication 1 , early media Whether in the PSTN or VoIP network, the final purpose of a call is to have two users talk (conversation). Here we will be generated by conversations between the users of the media called the regular media ("regular medium"). Early media ("Early media") is compared to conventional media. Typically, the user's co

Introduction to Open-Source SIP Server OpenSIPS and open-source sipopensips **************************************** **************************************** **************************************** ***Author: EasyWave time: 2014.09.14 Category: Linux application-open source server OpenSIPS introduction statement: reprinted, please keep the link NOTE: If any error occurs, please correct it. These are my Learning Log articles ...... *******************

We know that the SIP session process is very important in the SIP Protocol process. Let's take a closer look at the content of requests and replies. We will explain the meanings of some fields sent through the sip invite. SIP INVITE The caller Tesla first initiates the INVITE message to the called party Marconi. The I

----------------------------------------------------------------------------RFC list:3550 (RTP/RTCP)-3711 (SRTP)-5245 (ICE)-3261 (SIP)-4575 (sip-conference)-4566 (SDP)5389 (Stun)-5766 (Turn)-6455 (WebSocket)-6865 (FEC)-2616 (HTTP)----------------------------------------------------------------------------One, RFC 5245 ICE Study Notes---------------------------------------------------------------------------

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

Open source of SIP applications (proxy, PBX ,...) SIP Express Router (SER): Highspeed GNU sip proxy with a lot of features and a lot of ongoing development. homepage:Http://www.iptel.org/ser/. A really cool sip proxy-I like it! You can also take a look atDevelopment HomepageWith Web CVS. At the beginning you shoshoul

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

1. Sip1.1. overview 1.1.1. Basic components of the SIP System (1) User AgentIn sip, the user proxy (UA) is the endpoint entity. The User Agent initiates and terminates a session by exchanging requests and responses. As an application, UA includes the user proxy client and user proxy server, As follows:· User proxy client (UAC): the initial SIP request of the clie

With the development of VoIP and NGN technology, the Openh323 era is about to transition to the SIP era, in the open-source protocol stack, which occupies a dominant position, it puts a complex and advanced protocol stack presented in the eyes of ordinary programmers, which has made a contribution to the popularization of the system. However, when in the SIP era, there has been a state of separatist,

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

1 IntroductionVoIP is the most representative of the development of the new generation of Internet age.One of the application technologies. As a signaling control protocol in VoIPThere is great potential for growth. Therefore, in order to better promote the development of VoIP servicesIt will be a major research topic to solve the problem of SIP traversing NAT. This article mainlyBased on the stun method, it aims at its inability to traverse symmetric

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