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A Preliminary Study of IMS service provision Method Based on SIP Application Server

provide services, as can work in the following four modes: Terminate UA-In this mode, as acts as ue. For example, in a message service, assume that the receiver of a message sets a filter criterion. When the criteria are met, as may indicate that the receiver generates a final response when as is a sip ua. 2. Redirect server-as notifies the initiator of a new location or session substitution service that may be satisfied. For example, when a certain

Learning notes-SIP message receiving and processing flow based on sipdroid SIP audio and video telephony

/success (actual Sipprovider callback), the Fail/sucess interface of the transactionclient is called Public classTransactionclientextendsTransaction {Transactionclientlistener transaction_listener; Publictransactionclient (Sipprovider sip_provider, Message req, Transactionclientlistener listener) {Super(Sip_provider); Request=NewMessage (req); Init (Listener, Request.gettransactionid ()); //This.transaction_listener = listener;} //The actual Sipprovider callback Public voidOnreceivedmessage (Si

SIP Message Type and Message format

is described using another protocol, is the Session Description Protocol SDP .) Request Message Request Line = method + space + request address + sip version number + blank line A request line is used as the starting line. The request line includes the method name, request URL, Protocol version number, and separated by spaces. Six request methods: Invite sends a call Session Request Ack invite request final request Bye releases a call session Cancel

VoIP in-depth: An Introduction to the SIP protocol, part 2

, the response may come back with a contact address that is specific for the device, at least for the lifetime of the dialog. ack, and all subsequent requests, wocould place in its request URI the dialog's remote target. therefore, if a proxy previusly forked a request to an AOR, it wocould not do that for subsequent requests, as this time the request URI is different. Dialogs also hold a route-set. this route-set is a list of

Introduction to Open-Source SIP Server OpenSIPS and open-source sipopensips

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 ...... *******************

SIP Server Platform

RADVISION's SIP server platform is a framework for building all SIP servers and supporting fast and effective application software development for SIP servers, as RADVISION's SIP toolkit is in the leading position in the market, therefore, the platform implements the functions of the

SIP protocol stack parameter settings

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 yo

SIP protocol Learning 1

client, they start to communicate. The SIP protocol uses a client/server in text format and is a request response protocol, which defines multiple network entities that execute the corresponding functions. These network entities usually include user proxy UA and network server NS. UA is divided into user proxy client UAC and user

Several open source SIP stack comparison Opal,vocal,sipx,resiprocate,osip

support the newest rfc3261,resiprocate was born, but now, Resiprocate has become an independent SIP stack, it is very stable, and many commercial programs are in use.SVN: http://scm.sipfoundry.org/viewsvn/resiprocate/main/sip/Language:c++VxWorks Port:noWin32 Port:yesLinux Port:yesSupports RFC 3261:yesSupports RFC 2327:yesSupports RFC 3264:yesSupports RFC 3263:partialSupports RFC 3515:yesSupports RFC 3262:n

Research on Nat penetration Based on SIP

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

SIP Response Codes_server

fulfilled at this server; 5xx:server error¡ªthe Server failed to fulfill a apparently valid request; 6xx:global failure¡ªthe Request cannot be fulfilled to any server. SIP response codes, class 1:provisional messagesThese are sent within a SIP dialogue MB trying180 ringing181 call is being forwardedQueued183 Session Progress SIP Response Codes:2xx ClassThe 2x

SIP Learning (example)

prevent cyclic jumping= per proxy server, the integer minus oneTo:g. Marconi From:nikola Tesla = Indicates the sender and target party of the request message= If there is a user name tag in it, the address needs to be wrapped in angle brackets.= for INVITE messages, you can include tag in the from field , which is also a random codeCall-id: [email protected]The Identifier (dialog indentifier) on both sides of the call.Cseq:1 INVITEcommand seqence, ea

SIP protocol parsing and implementation (C and C ++ use Osip) 11

Chapter 8 query capability The SIP Options method allows one UA to query the capabilities of another UA or proxy server. This allows the client to detect information about the methods, content types, extensions, and encoding they support, rather than the other end of "calling. For example, before the client inserts a require header field into invite and lists the capabilities that are not supported by the t

SIP protocol overview audio and video tutorial

SIP is an application control signaling protocol proposed by IETF. As the name implies, it is used to initiate a session. It can be used to create, modify, and end multimedia session processes attended by multiple participants. Participants can communicate with each other through multicast, unicast, or network connection. There are clients and servers in the SIP. A client is an application that establishes

Detailed description of the SIP response status code Function

Status Codes and type status codes of SIP response messagesTemporary response (1xx) 100 trying in process180 ringing181 call being forwarder call forward182 queue181 * session progress session Successful SESSION (2XX) 200 OK session successfulRedirection (3xx) 300 multiple multi-Choice301 moved permanently permanent movement302 moved temporaily temporary movement305 use proxy user proxy380 alternative servi

Erlang SIP business introduction

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

Introduction to Open-Source SIP Server opensips

**************************************** **************************************** **************************************** ***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 ...... **************************************** **************************************** **************************************** *** I. Introduction to op

SIP (Session Initiation Protocol, Conversation Initiation Protocol)

Via:sip/2.0/udp lab.high-voltage.org:5060;branch=z9hg4bkfw19b = SIP version number ( 2.0 Udp), call address, = Branch transport identity = Via = transport type can be udp tcp tls, SCTP Max-forwards:70= Maximum number of hops, is the number of hops through the SIP server, mainly to prevent cyclic jumping= per proxy server, the integer minus

SIP Response Status Code control detailed

SIP Reply Message Status codeand function Type Status Code status descriptionTemporary response (1XX) trying is in process180 ringing ringing181 Call being forwarder calls are forwardLine queue181* Session Progress Sessions Session succeeded (2XX) OK session succeeded Redirect (3XX) multiple multiple selectionMoved Permanently permanent move302 moved temporaily temporary movement305 Use Proxy User agent380

Proxy and Back-to-Back User proxy (b2bua)

are their advantages and disadvantages? The most common Proxy Server only connects two UA instances. The b2bua Server is an intelligent entity and more powerful. It has some functions that proxy cannot do. It is more flexible and gradually replaces the general proxy server and becomes the mainstream of the SIP server.

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