sip

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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

SIP-to-HTTP connection

People often ask if SIP uses HTTP as the underlying protocol. The answer is in the negative. SIP is a protocol that works with HTTP on the same layer (that is, the application layer), which uses TCP, UDP, or SCTP as the underlying protocol. However,

Authentication Method in the SIP protocol

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

Doubango Sip/IMS registration process

I have developed custom Sip/IMS video clients, and supported voice, video, and instant communication functions. The video formats support h263, h264, and MPEG4 soft decoding, and provide hardware coding/decoding interfaces, provide servers. If you

Google released the world's first open-source HTML5 SIP client

The HTML5 SIP client is an open-source client that fully utilizes JavaScript to integrate social networking (Facebook, Twitter, Google +), online games, and e-commerce applications. No extensions, no plug-ins, or necessary gateways. The video stack

SIP rport Mechanism

3 Rport 3.1 solution description The obtained IP address is carried with the received parameter in the via header. In order to obtain port information, we also refer to this method, that is, to specify port information with the rport attribute in

Reliability of temporary response in SIP protocol

In the process of this test, the most disturbing problem is that the gateway is not receiving the alerting event during the call, causing the state machine to be disturbed. In fact, the SIP protocol already defines the reliability of the temporary

VoIP bookmarks from Klaus Darilion

Document directory RTP Stacks (mainly open source C/C ++ stacks) SIP Stacks RTP Applications SIP Phones (SIP User Agents) SIP Test Utility SIP Applications (Proxy, Location Server) Sip Express Router (ser) Ser Media Server (sems) STUN

Describe the concept of SIP Routing Mechanism

SIP is one of the most important protocols in VoIP services. For this protocol, we have discussed some basic content related to it in some previous articles. We will not go into detail here. The focus today is to explain the knowledge about the SIP

SIP Video Phone Based on HTML5 Technology

Developed for SIP/IMS video clients, supports access to sip Softswitch, IMS core network, andVoice, video, and instant messaging functions. The video formats support h263, h264, and MPEG4 soft encoding solutions. The hardware coding/decoding

Instances for communication between SIP and IAX Intranet and Internet and PSTN lines and mobile phones

At the request of the dynaguy brothers, I have attached a relatively complete experiment to colleagues who are still exploring: (There are some problems with the 2.0beta experiment, so I will demonstrate it with the most stable 1.2.3) In this test,

Troubleshooting Mac OS X 10.11 Installation SIP does not have permissions

In the process of building PYQT I met a very disgusting problem, in the installation of SIP after compiling the source of the installation process has been prompted me: Operation not permitted , I even reinstall the system is useless, and finally

SIP REDIRECT _voip technical data

SIP Redirect The SIP Redirect feature allows the IMG to respond to the 3xx class of SIP messages returned from a Redirect server. The 3XX responses provide information about a user ' s new location, or alternative services that is able to satisfy

Mac OS X El Capitan System Integrity Protection systems Integrity Protection (SIP)

introduction: The previous time experienced the Xcode compiler code was injected into the event, this time  mac  os X el capitan system upgrade, enabling a higher security protection mechanism: System Integrity protection systems Integrity pro

Source Code address of the VoIP open-source project

VoIP bookmarks from Klaus darilion Below you will find descriptions and links to sip and RTP stacks, applications, test utilities, SIP proxies, SIP pbxs and stun server and clients. most of them are open source :-), but not all of them If you have

Jain API for call control and Wireless Networks

Document directory Jain proposal SIP, ISUP, call control system, and Jain Interface Application of Jain APIs to Mobile Networks Mobile Station No-wire access to network (RAN) Network and Enterprise Internal capacity and service End-to-end

Yealink SIP-T20P IP Phone hide page Security Bypass Vulnerability

Release date:Updated on: Affected Systems:Yealink Yealink SIP-T20P IP Phone Description:--------------------------------------------------------------------------------Bugtraq id: 57029 Yealink SIP-T20P is an IP Phone. YeaLink IP Phone SIP-TxxP The

Application of STUN/TURN/ICE protocol in P2P SIP (I)

1 Description This article describes in detail the P2P SIP telephone process based on the STUN series protocol, which involves the interaction of SIP signaling, the principles of P2P, and Protocol interaction of STUN, TURN, and ICE. The interaction

SIP vs XMPP

Both SIP and XMPP are application-layer protocols that are used primarily to send voice and instant messaging over the Internet im,rfc3521 defines the sip,rfc3920 definition of XMPP. XMPP comes from instant messaging systems, SIP-like voice and

Asterisk source code parsing-SIP call

Is the call flowchart of Asterisk: We use the call process of SIP as an example to describe the call process of other channels. The call process (incoming) is as follows: Do_monitor-> sipsock_read-> handle_request-> handle_request_invite->

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