best voip softswitch

This is the second topic in the NAT traversal series of VoIP communications, Nat is a technology that overwrites the source IP address or/or destination IP address when an IP group passes through a router or firewall, this technology is widely used in private networks with multiple hosts but only one public IP address accessing the Internet. In the middle of 1990s, Nat emerged as a solution to address IPv4 address shortage to avoid difficulties in re

The Calculation Method of VoIP commonly used encoding bandwidth is as follows, which manufacturer has nothing to do with it:Bandwidth = package length × packets per second= Package length × (1/package cycle)= (Ethernet header + IP header + UDP header + RTP Header + payload) × (1/packaging cycle)= (208bit + 160bit + 64bit + 96bit + payload) × (1/package cycle)= (528bit + (package cycle (seconds) × number of bits per second) × (1/package cycle)= (528/pa

This article from csdn ucser, http://blog.csdn.net/perfectpdl reprinted to indicate the source, thank you! DTMF is called multi-tone dual-join, also called secondary dialing. There are three methods for VoIP to carry DTMF: inband, RFC 2833 (the latest RFC is 4733, which is referenced in IMS), and SIP info. The inband mode transfers the buffer generated by keys to the audio RTP stream, instead of defining special RTP events similar to RFC 2833. Eac

"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

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

Things can become more complex in scenarios other than those outlined above. for example, you might use prack (defined in RFC 3262), which is a provisional response acknowledgement. there are some cases in which we wowould like to guarantee the

Document directory For users of hash tables, this is an instant. The hash table operation is very fast. In computer programs, if you need to search for thousands of records within one second, a hash table (such as the spelling checker) is usually

Some time ago, I was dragged to. Net for two months and worked overtime every day. Alas, This Is What outsourcing companies do. Now you don't have to work overtime. You can study the audio and video communication based on SIP. After studying blogs

Google is testing a more practical function for Gmail Chat: users can directly call or receive Google Voice calls in the pop-up dialog box of Gmail. A new phone window with a digital keyboard will appear in the Gmail Chat Window, where you can also

I found a very good sip Forum: Huisi Communication Technology Forum Http://www.citiy.com A series of slides for SIP This handout is copyrighted by Zheng Yu. Allows copying, distributing, and customization under "GNU Free

**************************************** **************************************** **************************************** ***Author: EasyWave time: 2013.01.26 Category: Linux application LinPhone component speex Declaration: reprinted, please keep

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

required to process IN business requests under the control of SCF ◆ Business interaction management 4) protocol functions　 As an open, multi-protocol entity, standard protocols must be used to communicate with various media gateways, terminals, and networks. These protocols include: h.248, SCTP, ISUP, TUP, INAP, H.323, RADIUS, SNMP, SIP, M3UA, MGCP, BICC, PRO, BRI, etc. 5) interconnection functions　 The signaling gateway can be used to achieve intercommunication between the branch network and t