sip proxy

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Next-generation Internet protocols led by WLCP of SIP--BEA Systems (1)

to build and maintain through servlet and JSP models-View-control architecture, but they are basically text-based, it is rarely integrated with a variety of collaborations used by some Windows-based applications, such as various VB controls that can be integrated with audio and video-based media applications. Figure 1 user experience More and more people expect Web applications to integrate traditional HTML interfaces with functions such as instant messaging, voice over

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 are interested, please contact me. Registration process (Java --> C ++ --> C) Register (ngnsipservice. Java) | Register (ngnregistrationsession. Java) | Register _ (sipsession. cxx) | Tsip_action_register

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 technology relies on WebRTC. Like Demo Video demo on the home page, you can easily implement real-time video/audio calls between Chrome and IOS/Android mobile devices. This client is a technology that can

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 response. It is stipulated in the SIP standard that the definition of the reliable transmission of temporary messages during the call establishment process can be

SIP Protocol Resolution and implementation (C and C ++ use Osip) 6

, there are no other restrictions on requests that do not contain the Route Header domain. A simple and selectable method is provided to set the route table in advance, just like specifying an external proxy. However, it is not recommended to configure an external route. A URI should specify a pre-existing route set. If a request contains a route header field, it should be sent to the position indicated by the value at the top of the Route Header fiel

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 elcapitan system upgrade, enabling a higher security protection mechanism: System Integrity protection systems Integritypro tection (SIP), is by Design? Or is it a coincidence? for System Integrity Protection systems Integrity Protection (SIP), you canAppleDownload the website to study, fro

The SIP Protocol became the focus of the VoiceCon IP Phone Exhibition

Session Initiation Protocol (SIP) is the biggest winner in the VoiceCon IP Phone exhibition held this week. Some vendors have announced the launch of SIP-based upgrade products or added support for this security protocol to their existing products.SIP is a single signaling and Event Notification protocol that combines voice, video, and message communication. This Protocol also provides a standard-based appr

[Android Application Development]-(23) Android SIP support

1. What is sip? Step-by-Step sip Ii. How to Use sip in Android? Sipdroid, Demo: Android-SDK-Windows \ samples \ Android-9 \ sipdemo 3. Does the system support sip? Android provides the SIP function since 2.3. The SIP-related APIs

SIP (7)

response retransmits Timer E Initially T1 Non-invite request retransmit interval, UDP only Timer F 64 * T1 Non-invite transaction timeout Timer Timer g Initially T1 Invite response retransmit Interval Timer H 64 * T1 Wait time for ACK receept Timer I T4 for UDP Wait time for ACK retransmits Timer J 64 * T1 for ud Wait time for non-invite request retransmits Timer K T4 for UDP0 s for TCP/sctp

Sip rfc 4538 authorization request through dialog

Background: Generally, when a UA receives a request that creates a dialog (invite/Subscribe/refer), it must decide whether to authorize the request, in some cases, UA determines whether to authenticate the request by determining whether the request is in a created dialog, For example, after invite creates a dialog, UA does not need to authenticate other requests (prack, Act, etc.) sent in this dialog, but the problem is that for refer, message, and SUBSCRIBE requests, A new dialog will be crea

Static proxy and dynamic proxy, static proxy dynamic proxy

Static proxy and dynamic proxy, static proxy dynamic proxy The so-called proxy Design refers to the operation of a real subject by a proxy subject, and the actual subject executes specific business operations, while the

Py2exe ImportError: No module named sip

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 follows:Another Solution to the same problem: from distutils.core import setupimport py2exesetup(windows=[{"script":"main.py"}], options={"py2exe":{"includes":["sip"]}})

SIP using SDP with offer/Answer Model

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 first reliable non-Failed response. Note: At that time, the reliability Effect of rfc3261 was only 2 **. Next we will talk about 1 ** (except 100. 2.If the initial offer is in the invite message, the answer m

Open source sip Client for BlackBerry

Linphone BlackBerry and AMR-NB supportHttp://www.linphone.org/eng/linphone/news/linphone-for-blackberry.html Download Source Code here:http://www.linphone.org/eng/download/git.html Download install git for Windows http://msysgit.github.com/command line execute git clone git://git.linphone.org/linphone-blackberry.git-- Recursive Copy directory Linphone-blackberry to the new workspace. Import existing project (copy not selected) Modify the JRE in Java build path from 7 to 7.1 Modify Java

Research results of SIP audio calls in Android

First, we will briefly introduce the SIP protocol, which is a Session Initiation Protocol and mainly used for network multimedia calls. The sip api can be called only in android2.3 or later versions, and the device must support the sip before making a SIP call. The APIs used by SIP

Solution for Brekeke SIP Server "Database error: Connection error: jdbc: hsqldb: hsql" startup error

From: http://brekeke-sip.com/bbs/viewtopic.php? P = 11824 SID = 1337c4d609517c9d1f0fcc5167d7d5a1 1) Please go to Ondo SIP Server admintool> [config] menu> [system].Set [Java VM arguments] =-xrs 2) If you are also using Ondo PBX, please go to Ondo PBX admintool> [Options] menu. Please find two [Java VM arguments] fields in the page.One in PBX system settings and one in media server system settings. Please set[Java VM arguments] =-xrs 3) please go to

Sip-learn about prack

OverviewSIP defines two types of responses: temporary (Provisional) and final (final ).The final response transmits the request processing result, which is reliable ). The temporary response transmits the information of the processing process, which is unreliable by rfc3261.However, from the current situation, especially during the interaction with the PSTN, it is found that temporary responses should also be reliable.Rfc3262 defines an optional Extension Method for

WebRTC SIP Trickle Ice

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 the establishment time is very slow,Because the WEBRTC client needs to communicate with the Stun/turn server to get these candidate, the entire process is block, in order to increase the speed of call establishment, WEBRTC introduced trickle ice, in in

Preliminary Research on the SIP protocol

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 slow to read. Later, I found it was said that it was Huawei's internal sip training materials, or the Chinese language was good, and I had a general understandi

Open-Source SIP Phone Linphone

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 support was x264 (a sub-project of VLC ). Haha, I had a lot to do with what I used to do. Basically, I used to develop the SIP protocol stack based on Osip, exo

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