sip

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

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

Haha, if you have never touched network programming, don't look down.   Give a definition first:   SIP (Session Initial Protocol) is a signaling protocol that is used to set up, modify and terminate sessions, like Internet phone calland

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

Use FireBreath to develop real-time playback interfaces (Yate + SIP + FFMPEG + SDL) and firebreathyate At that time, such a blog post was really needed to guide this function module. Unfortunately, FireBreath has very little information on the

The SIP protocol is used in the national standard of the security video system. This document describes and develops a set of SIP protocol components. The exosip2 and osip2 libraries are generally used when developing such systems. This is an

Handling 4xx responses A specific 4xx response requires the processing of a specific UA instead of the Request Method. If a 401 (unauthorized) or 407 (proxy authentication required) response is received, UAC should carry the credential to retry the

Simple Solution: c: \ Python26 \ python.exe setup. py py2exe -- extends des sip   It is very convenient to use pyqt to complete the form interface, but there will be problems after packaging it into exe. The solution on the internet is as

According to the RFC3261-13.2.1, the offer/answer model used by the SIP is established in the dialog environment. The RFC also specifically imposes restrictions on offer/answer Interaction: 1.The initial offer must be in the invite message or the

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

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

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This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC clients establish a call through interactive candidate to achieve NAT traversal, if these interactions candidate in the Offer/anwser SDP will lead to through

Preliminary Exploration of OsipToday, we started to study the SIP protocol, called Session Initiation Protocol, which is a required protocol in VoIP.First, I found the RFC document, rfc3261, and more than 200 pages of English documents, which is too

Find this Linphone when looking for Open Source SIP Phone reference, download: http://www.linphone.org/eng/download/packages/linphone.html After reading about the structure, we used Osip, exosip, and ortp protocol stack for development. 264 of the

* RFC 2327 (SDP: Session Description Protocol) * RFC 2782 (dns srv Resource Record) * RFC 3261 (SIP: Session Initiation Protocol) * RFC 3263 (locating sip servers) * RFC 3264 (an offer/Answer Model with SDP) * RFC 3265 (SIP-specific event

There is a logical association between the commands and must be configured in a sequential order.1. Configure voice encoding and SIP address formatVoice class URI 2 sipHost IPv4:address of CUCMVoice Class codec 1Codec Preference 1 G729R8Codec

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)

Original: 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

The concept of unified communication needs to be understood from the combination of multiple protocols. Among them, there is more participation in the SIP protocol. The integration of unified communication involves software, hardware, mobile devices,