freeswitch voip

VoIP is a blockbuster in the communication market. Why? Because the price of VoIP is very low, it is very convenient to use, because this network-based technology is very expensive and has many advantages. Do you want to know more about VoIP phones? Go to the following link. Since its first launch in 1995, VoIP has bec

the Sofia of FreeSWITCH kernel research call flow involves content 1, received the nua_i_invite of a, returned 407, as follows: Sofia_handle_sip_i_invite =>sofia_reg_handle_register => sofia_reg_auth_challenge => 407 2, received the nua_i_invite of a, returned 180, as follows: Sofia_handle_sip_i_invite =>sofia_reg_handle_register => Sofia_reg_parse_auth Processing Nua_i_state messagesSofia.c:sofia_handle_sip_i_state...Mod_dialplan_xml.c:dialplan_hunt

It runs as follows: Gullible users receive an e-mail (or a phone call), are told that his credit card information is being stolen, and then let him quickly dial a phone number. This phone will have computer-made voice prompts, let him enter credit card number and other authentication information and so on. Security Policy for VoIP Have you ever heard of the latest VoIP fishing trick-vishing's notoriety?

VoIP has many security risks and faces many security threats, but this does not mean that the security of VoIP is irretrievable. In fact, with the frequent occurrence of security events, many VoIP manufacturers are also accumulating experience in practice and using some measures to guarantee the security of VoIP to a g

Absrtact: In recent years, with the development of technology VoIP gradually replaces the traditional long-distance business of the corresponding demand for VoIP business monitoring is also applied. At present, more equipment is often through the router firewall and other modified but to meet the OC192 wire speed requirements of the equipment is often very high for the entire network of comprehensive monit

Background information One feature of VoIP-type apps is the need to keep running in the background to be able to receive incoming calls at any time. Because of this special mechanism that the system provides to the VoIP process, we are unable to completely kill the VoIP process by killing command directly. For more detailed information on this point, refer to th

Some people think that IP phones are just another application running on the data network. However, imagine this situation: the quality of VoIP calls is getting worse and worse, and even the entire enterprise IP phone system stops running. Terrible? Pay attention to VoIP and add specialized VoIP tools for your network management platform. Recently, Network World

XMLXML is widely used in FS (FreeSwitch). dialplan is a very important and fast content. The following describes the statements in XML execution: How to associate it with its implementation? Where can I see its implementation? What is its implementation? Next let's take a look at how FS is done! The preceding XML statement is used as an example. 1. How to associate with its implementation When the channel of FS is in the EXCUTE state: Execute the cod

the packet analysis, the process is far from my expectations Ah!A flowchart is required here (the new version of Wireshark is too small!). ）It took nearly 2 days to grab packages, test, and even download the ITU t.38 Standard to view!Finally, we need to add a few simple parameters to achieve the standard T.38 protocol transceiver.Iv. Fax Transmission of t.381 Originate {fax_enable_t38=true, fax_enable_t38_request=true}sofia/gateway/tomx8/ 010xxxxxxxx txfax (/home/123. TIFF)V. t.38 's FAX recept

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 for similar click2call or online video communi

module_support.h phone. h text. h To the festival folder 6. Go back to the Ekho installation directory: Run make: Appears: G ++:./lib/libfestival. A: No such file or directory G ++:./lib/libestools. A: No such file or directory G ++:./lib/libestbase. A: No such file or directory G ++:./lib/libeststring. A: No such file or directory Make [1]: *** [test_ekho] Error 1 Solution: Run the following command in the Ekho installation directory: CP lib32 lib Under the installation directory of festival a

Processing of FreeSWITCH SIP signaling in Mod_sofiaFirst, a thread that handles SIP messages is created inside the module's load (mod_sofia_load): /* Start one message thread /switch_log_printf (Switch_channel_log, Switch_log_info, "starting initial message Thread.\n "); Sofia_msg_thread_start (0); The Config_sofia function is then called. if (Config_sofia (Sofia_config_load, NULL)!= switch_status_success) { mod_sofia_globals.

FreeSWITCH supports audio and video recording functions, recording function is mainly through Mod_sndfile, mod_shout and other modules to achieve, video module is through the MOD_MP4V2 and Mod_av modules. Mod_sndfile Recording wav Mod_shout Recording mp3 Mod_mp4v2 recording MP4 file Mod_av is a new module available in the 1.6.x release I also put the Mod_av in the 1.5.x version, the use of this module to play video files, connect the camera and

recording of the FreeSWITCH kernel research First, register callback function Application Call Switch_ivr_record_session-> Switch_core_media_bug_add registering the Mediabug interface callback function with the FS kernel Record_callback listening for streaming media (read-write mode) second, the kernel calls the switch_core_session_read_frame to obtain the utterance tone 1. Call endpoint's Read_frame function to get endpoint's voice. 2, call the me

With the decrease in the cost of using VoIP, family and individual users are receiving more and more requests for using Vonage (or other similar products). As VoIP Communication continues to grow in the area of home calls, in addition, open source code projects are becoming more and more powerful. Based on this background and environment, Asterisk is a new product that can replace traditional PBX and is sui

VoIP is the future direction, but it is not always better. At the VoIP Summit held by the Association of higher education and communication experts (ACUTA) recently, the participating expert groups agreed that saving money is no longer the purpose of deploying VoIP in colleges and universities, the new demand is to explore integrated communication and recognize t

The earliest VoIP technology service was created to solve the long-distance telephone bill problem. However, with the continuous improvement of technology, the VoIP technology service has gradually expanded its scope, video, fax and other services, and it can also achieve low-cost and high-quality transmission functions. In itself, this has already greatly impacted the market ......

Security is often one of the main reasons for enterprises not to use VoIP. Network administrators need to overcome such exaggerated publicity and apply correct security measures to maximize the reliability of voice networks and successfully launch such services. Changing to VoIP requires many changes. In the old-fashioned TDM (time-division multiplexing) Speech Communication, PBX (switch) is a separate clos

Pjsip is an open source SIP protocol stack. It supports a wide range of SIP extensions and is arguably one of the most popular sip stacks. It implements SIP, SDP, RTP, STUN, turn, and ice. PJSIP provides a very clear API as well as NAT traversal capabilities as a SIP-based multimedia communication framework. Pjsip is very well-ported and supports almost all today's systems: from desktop systems to embedded systems to smartphones. Pjsip also supports voice, video, status rendering, and instant me

Relationship Management) system closely with, In order to maximize customer satisfaction and loyalty to enterprises to bring more business opportunities. Whether it is internal communication, or communication between enterprises and customers, it will be a huge application field of fusion communication. Converged Communications Service is based on the ability to provide voice, data, video and multimedia business information and communication technology to carry out a full range of business ser