sip proxy

This article original from the http://blog.csdn.net/voipmaker reprint indicate the source. Dual-stream is the concept in video conferencing. It generally means that the client can display two video streams at the same time, one is the main video (main), and the other is usually the content sharing (slides ), it is also called share content. Content is usually screen, PPT, document, and other content. The SIP protocol implements dual-stream. The SDP c

Learn some of the FreeSWITCH core commands, and then learn more about FS in detail.To see if it was not previously suspected, two times when programming changes the configuration file, or Java injects some parameters into the configuration file, learn more about the following configuration file.This should be difficult, not clear.Ask Mr. Baidu.Learn a new knowledge of how FS adds recording functions to configureThe general telephone system can record voice calls in the system, and voice recordin

In sip, both re-invite and update are used to modify the session parameter. The difference is that update does not affect the status of the dialog, and re-invite changes the status of the dialog. Therefore, update can be sent before the first invite is responded (that is, before 200ok of invite is received ). That is to say, update can be used to control early media. The re-invite can only be sent after the first invite cup responds (that is, after th

InProgramWhen the MessageBox pop-up error message is applied, when you click the OK button, it is found that the keyboard icon on the mainmenu has disappeared. You must click it once to appear. Although it does not affect the use of normal functions, it may be confusing for PDA cainiao, and many methods to refresh the interface cannot be solved. Baidu cannot find the answer to this strange problem. You can directly query the problem on Google and find that foreigners also encounter this proble

A transaction is a request transaction sent by the customer (through the communication layer) to a server transaction, together with all the responses to the request of the server transaction, sent back to the client transaction. The transaction layer processes the re-sending of the application service layer, matches the response of the request, and the timeout of the application service layer. All tasks completed by a user agent client UAC are composed of a group of transactions. Generally, a

Based on the practices in the past few days, we have found an Optimal Configuration: 1. The SIP server uses trixbox. If you are familiar with Linux, we recommend that you use asterisk directly. 2 If the client is used directly, it is recommended that ekiga. By the way, how do I feel when using several clients: 1 Linphone: It seems to be well known. However, the latest version 3.1.2 crashes after being installed. I installed a general XP-SP3, compu

, has been completely attracted by WxPython , can not wait to practice, hereby record the individual learning process (Windows system):First, Wxpython environment installationThe most realistic and practical way to install Pip, and must be used in a manner that will synchronize the installation of dependent packages: pip install-u wxPythonsecond, Wxpython tastedAs a beginner, must not blindly directly into the subject, it is necessary to go through this stage, can be very good to help understand

versatile and most like relational database in a non-relational database. His support for the data structure is very loose, is similar to JSON Bson format, so can store more complex data types, he is mainly used to solve the massive data access efficiency problem. His storage seems to have a larger demand for disk space. The new version starts to support distributed. 4, Hypertable Hypertable and similar hbase are developed from Google's BigTable model, which is good for distributed support, bu

Dynamic Proxy Case 1: use proxy dynamic Proxy to enhance the method. Use Case proxy Dynamic proxy Case 1:/* Requirement: use Proxy dynamic Proxy to enhance the MethodQuestion:1. Define

"Python/c++ Interface Library comparison" (Swig,boost.python, Pycxx, py++, sip, Weave, Pyrex) http://blog.csdn.net/lainegates/article/details /19565823There are many open-source python/c++ binding tools, search a lot of 岾, here a little summary.SWIG Supports Python 2 and 3 Properly configured, the package can be fully automated (*.i files need to be written by themselves) When it is not fully automatic, it will mostly repeat your. h f

novice?The discovery directory has a lot of configuration files, is it necessary to change these configuration files two times? Let's take one to learn.FreeSWITCH is configured by default to 1000 to 1019 (ext.) A total of 20 users. Let's not start by downloading a SIP phone client on our own phone and try to talk.First Ipconfig/all know your LAN address. The password default appears to be 1234.My own IP is 192.168.0.113, and then I correspond with th

FreePBX SIP TrunkDockingbackground: PBX1 is a virtual machine running FreePBX, whichnow needs to be connected via SIP TRUNK docking , PBX2, using PBX2 E1 The line calls out the phone. PBX1 192.168.100.1PBX2 192.168.100.2PBX1on the configurationOneConfigurationTrunkNew SIP TRUNK650) this.width=650; "src=" http://s3.51cto.com/wyfs02/M01/54/0B/wKioL1R2mLHDypJCAACZ

You can use the method provided by. Net CF itself to enumerate all the SIP messages on the device. See: http://msdn.microsoft.com/en-us/library/ms172538.aspx Code highlighting produced by Actipro CodeHighlighter (freeware)http://www.CodeHighlighter.com/--> // Define an inputpanel Private Inputpanel m_inputpanel = New Inputpanel (); // Enumerative sip Foreach (Inputmethod Method

1. The following is the test matrix 1 (the problem is not resolved ): CalleeCaller Jain-Sip-UA X-Lite IP-video Jain-Sip-UA It can communicate normally, but the number of frames is not enough and it is not smooth. When the call is received, the xlite is dropped. After the IP address is answered, "keep" is displayed, and UA audio and video can be received. X-Lite

The network structure is as follows:Asterisk (192.168.1.99) That is to say, both Asterisk and SIP terminals are behind Nat. The solution is as follows:1. Modify the SIP Extension settings in the SIP _. conf file.Nat = YesQualify = yes; it seems this item is not requiredExternip = 55.66.77.88; change to match our external IP addressLocalnet = 192.168.1.0/255.255.

This article from csdn ucser, http://blog.csdn.net/perfectpdl reprinted to indicate the source, thank you! I have created a freeswitch learning and communication group, 45211986. welcome to join. Freeswtich can be used as the rtmp and SIP gateway of the Streaming Media Protocol. It can communicate with the SIP video phone through flash in a web browser. This function can be used on the browser side f

Currently, the main reason for the combination of the codec and call control of the SIP protocol stack is the reuse of larger particles. In this case, the SIP is too bloated. It is not easy to expand.What I want to consider now is to break down the SIP library into two databases, one responsible for codec, and decoded by the