recently, the SIP protocol was used, so we looked for two open source projects to compare, Linphone and Pjsip, and finally chose Pjsip this open source protocol stack for development.The main reasons are as follows (for personal reference only):1, Linphone code structure than Pjsip clear, pjsip in Windows more convenient debugging ;2, Linphone after the update does not use Osip as a protocol stack, instead of self-written BELLE_SIP,PJSIP protocol stack is maintained, and has been stable ;3, Pjsi
to do not worry, then configure the "Call Options", set "sipdroid priority", in order to facilitate the use you can choose the last item, this ... Translation estimate deserted, meaning "always ask", we hook this in the software interface in the upper right corner of the "5" this button to dial, you can also directly in the interface of the "Phone number" box to enter the number in fact, you can ignore, directly open the phone with the dialer dial-up input the number to be dialed, press the cal
Tested by: mx60 VoIP Voice Gateway Bug: getting the administrator password to log on to control the entire gateway. Impact scope: no device test is available for users with MX and operators, haha MX60 introduction Figure 1 Brief Description: MX60 is a carrier-level Voice Gateway. The permission settings for managing users are divided into two levels: Administrator and operator. The specific permission is granted to me (figure 2 ). However, the permiss
Transfer from China VoIP Forum
On the PBX or local switch side, a small amount of power is not fully converted and returns along the original path, forming an echo. If the caller is not far from the PBX or vswitch, the echo will return very quickly and the Speaker cannot hear it. In this case, it does not matter. However, when the response time exceeds 10 ms, the human ears can hear the echo. In order to prevent echo, echo cancellation technology is
**************************************** **************************************** **************************************** ***Author: EasyWave time: 2013.03.31
Category: Linux application LinPhone Declaration: reprinted. Please keep the link
NOTE: If any error occurs, please correct it. These are my Learning Log articles ......
**************************************** **************************************** **************************************** ***
In 《Linux-based open-source
Introduction
My company wowould like to set up a call center. the call center needs a VOIP router. we choosed Cisco 2811, and we applied an E1 cable to host 30 phones in. we need to do necessary configuration in the Cisco 2811 router.
Config the Cisco 2811 router
We have Ed the router. we started to config the router. first we checked the delivery list file in the package, and found a console cable. the console cable is used to config the router. one
Http://www.lxvoip.com/thread-36596-1-1.html
3cx phone system, which is based on WindowsThe VOIP server software can replace the traditional dedicated hardware program-controlled switch. It has a Chinese operation interface and is easy to set up. It is suitable for enterprises to build a telephone network and allow free calls between extensions,Each extension can also be called to a traditional telephone network, or used as a telephone customer service
"On the side of a PBX or bureau switch, a small amount of electrical energy is not fully converted and returned along the original path to form an echo." If the caller is not far from the PBX or switch, the Echo returns quickly and the human ear
H323 open source code
: Www.openh323.org
There are protocol stacks, soft terminals (openphone, ohphone), and opengk, openmcu .....
Http://www.voxgratia.org/documents.html
Open g.729
I. Features of ECHO in Internet voice communicationCompared with traditional telephones, real-time voice transmission over the Internet has a fatal weakness. That is, the quality of voice is poor, and there are many factors that affect the quality
SIP is an application control signaling protocol proposed by IETF. As the name implies, it is used to initiate a session. It can be used to create, modify, and end multimedia session processes attended by multiple participants. Participants can
I recently found out doing a sip capture on a FreeBSD system is a little different than centos or other Linux distributions. here is a easy to use command that will grab the SIP packets from TCP dump. this will give you an easy to read text file for
Principle: InProgramAfter the program is switched to the background, an action is executed every 10 minutes to wake up the program, thus ensuring that the program runs in the background.
1. Add the uibackgroundmodes key value to info. plist.
Let
"IPv6 configured Commands"
IPv6 Install/Installation IPv6, limited to XP system use
IPv6 unicast-routing/Enable IPv6 forwarding, configuring on the router
Show IPv6 router/view IPv6 routing Table
IPv6 router RIP test/enable RIP protocol, named
To the same version to import backup VOS2009 data interrupts the current call login ssh. Terminating vos service/etc/init.d/mbx2009d Stop/etc/init.d/vos2009dall stop/etc/init.d/mysql STOPCD/VAR/LIB/MYSQLCP-DPRF Vosdb/root/vosdbdate/etc/init.d/mysql
1. VoIP1) First, modify the plist configuration of the application and add the following settings:Application does not run in Background: NoRequired background modes: VoIPNote: after these configurations are added, the application will automatically
Document directory
Registering multiple user devices
The via header, forking, loop prevention
An example using proxies
User location
Let's step out of the SIP layers and see what we have so far: using the layers, we can now create and receive
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