The Basic Conditions for the Functioning of VoIP-system

VoIP is a voice transmission via dedicated digital channels. This type of connection allows you to save on international and long-distance calls, pay for them as local ones, as well as provide other communication functions facilitating the work of the office — sending and receiving SMS/email, database maintenance, call hold and more.

Many of today’s users believe that VoIP is the best solution for the business. Data transfer is carried out at high speed and cheaply via by Internet protocols. This type of connection provides a computer-phone-internet system.

However, whatever the high-quality technology may be, it cannot exist without the appropriate equipment and competent service. Many beginners in this area ask the question: what is a voip gateway? This device allows you to convert calls from landline and cell phones in the VOIP-signals and vice versa. Accordingly, it becomes possible to transmit voice traffic over long distances at the lowest price.

The equipment facilitates communication office functions and relieves staff. Caller ID, data transmission via the voice mail, call forwarding to mobile personnel’s devices can save their time and make more money.

There are single-channel and multi-channel VoIP gateways. For home use and small communication it is better to give preference to the single-channel equipment, while for intensive office communication the solution is multi-channel equipment. Among the proposals from a variety of manufacturers, GoIP voip-gsm gateways are considered the best in terms of price and quality.

The equipment should be connected to a computer and set up by a technician. By the way, having read the info on forums, some users try connecting the gateway independently. In such cases there are problems such as no connection, system crashes, undetectable channels, etc. And if you add the wasted time and money to this, it becomes clear that trying to set up a voip gateway independently is not the best solution when connecting VoIP telephony.

  • SBO Multipath with Integrated SyncSwitch- Linux based SIP Solution.
  • Baresip Portable SIP useragent with Video support
  • Blink: It supports wideband VoIP (Opus codec), Chat, File Transfer and Multiparty conferencing based on MSRP protocol
  • Cockatoo
  • Ekiga || SIP, H.323 audio and video softphone for various linux, solaris, windows, and various unix systems. Formerly GnomeMeeting
  • FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk
  • Jitsi (formerly SIP Communicator) Audio/Video phone and messenger with end-to-end encryption through ZRTP — Multiplatform — Open Source (also supports XMPP, MSN, AIM, Yahoo! and others).
  • Kphone
  • Homer — live conferencing and more: free cross-platform video conferencing software, supporting unlimited amount of participants in a video/audio conference
  • Linphone audio and video SIP softphone for Linux and Windows XP
  • minisip cross-platform SIP softphone, Linux, Windows XP and soon Windows Mobile 2003 SE
  • MUVConf cross-platform SIP multi-user video conference. See demo video. Download from code.google.com
  • MjUA: simple cross-platform SIP softphone, written in java, based on MjSip stack
  • Open IP Phone Business IP Phone sdk support, ims compliant, good interoperability.
  • OpenSIPStack MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
  • OpenSoftphone: A simple Java based SIP softphone using the PjSip-jni wrapper.
  • OpenZoep: GPL telephone and IM messaging client engine
  • Peers Minimalist SIP softphone written in java (tested on linux and windows)
  • PhoneGaim
  • PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
  • QuteCom ex-OpenWengo: a fully SIP compliant multiplatform softphone with many features
  • SFLphone, open-source multiplatform multi-protocol VoIP client
  • Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux
  • SipToSis from mhspot.com Skype SIP UA — Multiplatform — Open Source
  • sipXezPhone («sipX easy phone») from SIPfoundry based on sipXtapi
  • sipXphone from SIPfoundry, previously known as the Pingtel phone
  • Twinkle
  • YateClient is multiprotocol and multiplatform softphone with H.323, SIP, Jingle and IAX support.
  • YeaPhone: A SIP softphone for the Yealink USB-P1K handset based on the libLinphone backend
  • CRM Integration Client Open Source program writen on java. based on MJ SIP and SIP-Communicator for Call-Centers solutions

 

MacOS X clients:

  • Blink: It supports wideband VoIP, Instant Messaging, File Transfer and Desktop Sharing based on MSRP
  • FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk
  • Jitsi (formerly SIP Communicator) Audio/Video phone and messenger with end-to-end encryption through ZRTP — Multiplatform — Open Source (also supports XMPP, MSN, AIM, Yahoo! and others).
  • PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
  • QuteCom ex-OpenWengo: a fully SIP compliant multiplatform softphone with many features
  • SFLphone, open-source multiplatform multi-protocol VoIP client
  • Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux
  • Telephone: A SIP softphone designed for the Mac (written in Objective-C/Cocoa). Very good integration with Mac OSX : Dial from Addressbook, dial tel: URIs from Safari, notifications with Growl.
  • YateClient skinnable VoIP client based on QT library which supports H.323, SIP, Jingle and IAX protocols
  • REMWAVE Communicator OS X Open source SIP phone for OS X. Based on PJSIP library, scriptable with Apple Script and address book integration.

 

Windows clients

  • Blink: It supports wideband VoIP (Opus codec), Chat, File Transfer and Multiparty conferencing based on MSRP protocol
  • Brief Msg is simple SIP messenger.
  • Ekiga || SIP, H.323 audio and video softphone for various linux, solaris, windows, and various unix systems. Formerly GnomeMeeting
  • FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk
  • Homer — live conferencing and more: free cross-platform video conferencing software, supporting unlimited amount of participants in a video/audio conference
  • Jitsi (formerly SIP Communicator) Audio/Video phone and messenger with end-to-end encryption through ZRTP — Multiplatform — Open Source (also supports XMPP, MSN, AIM, Yahoo! and others).
  • JPhone Rich software SDK support softphone development, Windows, Linux, ThreadX, Vxworks etc.
  • Linphone audio and video SIP softphone for Linux and Windows XP
  • MicroSIP: lightweight SIP softphone based on PJSIP stack for Windows OS written in C++. SIMPLE IM and Presense.
  • minisip cross-platform SIP softphone, Linux, Windows XP and soon Windows Mobile 2003 SE
  • MUVConf cross-platform SIP multi-user video conference. See demo video. Download from code.google.com
  • MjUA: simple cross-platform SIP softphone, written in java, based on MjSip stack
  • OfficeSIP Messenger is audio-video softphone and instant messenger, open source alternative to MS Office Communicator.
  • OfficeSIP Softphone GPL audio-video softphone.
  • OpenSIPStack MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
  • OpenSoftphone: A simple Java based SIP softphone using the PjSip-jni wrapper
  • OpenZoep: GPL telephone and IM messaging client engine
  • Peers Minimalist SIP softphone written in java (tested on linux and windows)
  • PhoneGaim
  • PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
  • QuteCom ex-OpenWengo: a fully SIP compliant multiplatform softphone with many features
  • REMWAVE Communicator Win Open source soft phone for Windows. Written in C# and based on the PJSIP library. Including branding engine.
  • Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux
  • SipToSis from mhspot.com Skype SIP UA — Multiplatform — Open Source
  • sipXezPhone («sipX easy phone») from SIPfoundry based on sipXtapi
  • sipXphone from SIPfoundry, previously known as the Pingtel phone
  • tSIP Portable, BSD-licensed softphone with BLF, call recording, customizable keypad and shortcuts, browser integration. Based on re/rem/baresip.
  • VMukti (formerly 1videoConference) alpha: a web2.0 VoIP video conferencing software for Asterisk.
  • wxCommunicator Windows softphone based on sipXtapi and wxWidgets 2.8.x, multi-account, conferencing, NAT support
  • YateClient is multiprotocol and multiplatform softphone with H.323, SIP,Jingle and IAX support.

 

  • SBO SIP Proxy Bypass All types of Internet Firewall
  • JAIN-SIP Proxy
  • Mini-SIP-Proxy A very tiny perl POE based SIP proxy
  • MjServer cross-platform SIP proxy/registrar/redirect, written in java, based on MjSip stack
  • MySIPSwitch SIP Proxy server which allows using multiple SIP accounts with a single SIP login
  • NethidPro3.0.6 Opensource Sip Encryption Bridge: www.vonets.com
  • Net-SIP A Perl SIP framework that includes a stateless proxy
  • OpenJSIP Opensource distributed standalone SIP proxy, SIP registrar, SIP location service run by Java VM. Based on NIST SIP and derived from JAIN-SIP Proxy.
  • OpenSBC: MPL licensed SIP proxy/registrar/B2BUA with NAT traversal and ENUM
  • OpenSER: GPL SIP Server with TLS support — renamed to Kamailio
  • OpenSIPS forked from OpenSER.
  • partysip SIP proxy server
  • repro from the reSIProcate project fully implements Federated VoIP and has a built-in web UI for quick setup
  • REMWAVE Calamar Cross-platform high performance SIP proxy written in Java
  • SaRP SIP and RTP Proxy in Perl
  • SIP Express Router (SER): the SIP router/proxy/jack-in-all-trades from IPtel.org
  • Siproxd SIP and RTP Proxy
  • SIPVicious tool suite: tools for auditing sip devices
  • sipX The SIP PBX for Linux: Complete, native SIP PBX solution for business
  • Vocal SIP softswitch with H.323 and MGCP translators for non-SIP endpoints
  • Yxa Written in the Erlang programming language
  • CRM INtegration Proxy Open Source program writen on java. based on MJ SIP lib Proxy for Call-Centers solutions
  • Clearwater — open source IMS (IP Multimedia Subsystem) implementation designed for massively scalable deployment in the Cloud — SIP routing components built on PJSIP