polycom sip

Reprint: http://www.cnblogs.com/ishangs/p/3816689.htmlApplication of Stun/turn/ice protocol in peer-to sip (II.)1 description2 dozen holes and the concept of crossing ... 13 hitting holes and crossing ... 24 using the STUN Series Protocol traversal features ... 25 Stun/turn/ice The relationship of the agreement ... 36 Stun protocol (RFC 5389) 36.1 Why the STUN protocol is used ... 3How the 6.2 stun protocol works ... 47 Turn protocol ... 47.1 Why the

These days use to FreeSWITCH docking other equipment knowledge, here to tidy up, also convenient I later check. Operating system: debian8.5_x64 FreeSWITCH version: 1.6.8 First, FreeSWITCH as the called deviceFreeSWITCH as a device and other devices docking the situation is relatively simple, you can directly through the 5080 port inbound.FreeSWITCH default configuration turns on port 5080 docking (for public in conf/dialplan/public.xml):extensionname= "Public_extensions"> co

Basic settings of SIP Trunk in trixbox Http://www.voclub.net/zone? Action-viewthread-tid-1065The basic settings of the SIP Trunk in trixbox are as follows: the extension can call the phone number through a SIP Trunk, and then ring the trunk number to the extension. Create a new SIP Trunk, provided that you have obtain

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

"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

Csdn lidp http://blog.csdn.net/perfectpdl The SIP response to the invite request may be final or temporary. The final response is always sent reliably, but not the temporary response. You can use the prack (temporary response confirmation) method to reliably send a temporary response.To develop applications that support prack, the following conditions must be met: The client sending the invite request must put a 100rel tag in the supported or requ

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