sip proxy

1xx = notification response 100 trying 180 dialing in progress 181 being transferred 182 queuing 183 call progress 2XX = successful response 200 OK 202 accepted: used for referral 3xx = Transfer Response Over 300 options 301 permanent migration 302 temporarily migrated 305 use Proxy Server 380 alternative services 4xx = call failed 400 improper call 401 unauthorized: only for use by the Registry. 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

The SIP Protocol provides a standard-based IP communication method for multiple devices and applications. This White Paper describes the support for the SIP protocol in the Cisco communications system. Cisco's Unified Communication System includes IP voice, data and video communication products and applications, which can help organizations communicate more effectively, simplify business processes, and achi

SIP Reply Message Status codeand featureType Status Code status descriptionTemporary response (1XX) Trying is in processRinging ringing181 call being forwarder is forward182 Queue Queue181* Session Progress SessionsSession success (2XX) OK session succeededRedirect (3XX) multiple multiple options301 Moved Permanently permanent mobile302 moved temporaily temporary movement305 Use Proxy User agent380 Alternat

exists, asterisk considers this request to be re-invite (! P-> owner). Otherwise, it is considered as a new invite. There are many stories about re-invite, involving whether Asterisk is b2bua or proxy. Next we will discuss non-re-invite requests. See from printed information Ast_verbose ("using invite request as basis request-% s/n", p-> callid ); Using invite request as basis request-zjriyjzkyzyzzdnjndrmmjhmmmnlnzdmode4ntyzzme. If you enable

because it does not support the URI scheme in request-Uri. 4.15 bad ExtensionThe server does not know the Protocol extensions specified by the proxy-require (20.29) or require (20.32) header in the request. The server must list unsupported extensions in the unsupported header field. For how UAC handles this response, see 8.1.3.5 4.16 421 extension requiredUAS requires a specific extension to process this request, but this extension is not listed in t

client should try again in the sip uri mode. If you receive a 420 (incorrect extension) Response (rfc3261 section 21.4.15 ), the request lists an extension option not supported by the proxy or uas in the require or proxy-require header domain. UAC should ignore the extensions listed in the response's unsupported header and try again. In all the above cases, the

, it can be faster. For more information about these problems, see rfc3489. If it is a server that supports the rport mechanism, it needs to check whether the via header contains an rport parameter with no value in the received request. If yes, it needs to include the rport value in the response, which is similar to the processing of received. To traverse symmetric Nat, the response must be sent to the same IP address and port. When a server listens for a request from multiple ports or inter

The Cisco SIP VoIP architecture solution provides users with many services. The table lists various IP telephone services that can be implemented by Cisco SIP VoIP Architecture solutions.Cisco SIP VoIP architecture solution ComponentsThe Cisco SIP VoIP architecture solution consists of the following elements:Sip ip tel

Things can become more complex in scenarios other than those outlined above. for example, you might use prack (defined in RFC 3262), which is a provisional response acknowledgement. there are some cases in which we wowould like to guarantee the delivery of a provisional response different than 100, such as 180 ringing (You may recall that TCP isn't enough to know the message reached the other end, as a proxy along the path may send the response over U

HTTP Authentication SIP provides a stateless, trial-and-error mechanism for the authentication system. This mechanism is based on HTTP authentication. At any time, the proxy server or UA receives a request (except in section 22.1), which attempts to check the identity confirmation provided by the request initiator. When the sender confirms the identity, the request recipient should confirm whether the user

message to the URI. Louse Routing, lr ): This routing mechanism is flexible and is also the soul of the SIP routing mechanism, which is defined in RFC 3261. The following describes the route decision-making process of a loose routing Proxy: 1. The Proxy first checks whether the request-URI of the message belongs to the domain in which it is responsible. If yes,

services that might is able to satisfy the CAL L. Multiple choices-the address at the request resolved to several choices, and the, and the own specific, and th e User (or UA) can select a preferred communication end point and redirect it request to that location. The moved permanently-the user can no longer is found at the address in the Request-uri, and the requesting client Shoul D retry at the "new address" given by the Contact Header field (section 20.10). The requester SHOULD update any l

Chapter 1 SIPP IntroductionSIPP is a tool software used to test the performance of the SIP protocol. This is a GPL open source software. It contains some basic sipstone user proxy workflows (UAC and UAS) and can be used to create and release multiple calls using invite and B ye. It can also read the XML scenario file, that is, the configuration file that describes any performance tests. It dynamically displ

this method is not allowed for addresses in this request-uri. The answer must include a Allow header field that contains a list of methods allowed for the specified address. 4.7 Not acceptable The resource in the request will only cause an error that the content cannot receive outside the accept header in the request. 4.8 407 Proxy Authentication Required This return code and 401 (unauthorized) are class four, but flag the client should first pass au

Chapter 9 dialogue A key concept for user proxy is dialog. A dialog indicates a point-to-point sip connection between two user proxies at some time. The dialog ensures that messages between user proxies are ordered and correctly routed. A dialog indicates the context of a SIP message. Rfc3261 the UA processing discussed in section 8th is irrelevant to the method.

Section 3 redirect servers In some frameworks, relying on the proxy server can reduce the load on the proxy server, which is beneficial for forwarding requests and enhancing signals. Redirection allows the server to send route information to the client through the response to the request, so it frees itself from the subsequent message loop of the transaction, at the same time, it can continue to accurately

Let's talk about the "things" Of The osx sip mechanism"I. Preface OSX is an exclusive operating system developed by Apple for Mac products. It is the first FreeBSD-based system to adopt an object-oriented operating system. From the OSX V10.0-V10.8 version, OSX systems are all codenamed big cats, maybe old Joe is also a fan of cats (I am also a loyal fan of cats, haha ), the OSX system is evolved as follows: 2001Mac OSX 10.0 Cheeta Cheetah 2001Mac O

Session Initiation Protocol,sip is a signaling protocol that establishes, modifies, and terminates a session between two endpoints. SIP can be used to establish a multicast session for two-party calls, multiparty calls, or even Internet calls, multimedia calls, and multimedia distribution. The JSR 116:sip Servlet API is a server-side interface that describes cont

This article, the original connection: http://blog.csdn.net/freewebsys/article/details/46546205, reprint please indicate the source!1, about FreeSWITCHFreeSWITCH is a soft-switching solution for telephony, including a softphone and soft switch to provide voice and chat product drivers. The FreeSWITCH can be used as a switch engine, PBX, multimedia gateway, and multimedia server.FreeSWITCH supports a variety of communication technology standards, including SI