SIP Phone service which lets you use your SIP account anywhere in the world via their web based SIP service. The good thing is they already have a few popular SIP services pre-configured, so you don’t even have to configure their SIP.
SIP (Session Initiation Protocol) is the most popular Voice over IP (VoIP) standard. SIP enables two or more people to make phone calls to each other using the Internet to carry the call.
Using Flex, Java and Red5 Server we can develop Web Sip Application. The main advantage of these phone is no need to install at client side. Simply we can browse application and we can register and make call.
Asterisk. Digium’s open source communications engine powers voice and video communication solutions worldwide. Discover the power of Asterisk, let us help you choose the right Asterisk solution.
Click2Call is a service that provides users of your websites to make calls from your website and leave voicemails for you by simply clicking a button.
TringMe also provides a complete ecosystem wherein VoIP providers, Service providers, Enterprises, Developers and End-users can play. An ecosystem which can cater to developing innovative applications that integrate Voice and Telephony without worrying about the underlying details of call routing and signalling.
Build applications that use the phone network to interact with people on landlines and cell phones all over the world. In just a few lines of code, you’ll have phones ringing.
Build voice enabled applications directly in PHP.
Make VoIP calls anywhere in the world via Doddle web based SIP phone directly from your webpage.
Public Internet Telephone: With the free, online, no registration Doddle phone service, VoIP is as easy as accessing a webpage: just start using!
It’s a Doddle.
Linphone is an internet phone or Voice Over IP phone (VoIP).
* Linphone can be used for communicating freely with people over the internet, with voice, video, and text instant messaging.
* Linphone makes use of the SIP protocol , an open standard for internet telephony. You can use Linphone with any SIP VoIP operator, including our free SIP audio/video service.
* Linphone is free-software (or open-source), you can download and redistribute it freely.
* Linphone is available for desktop computers: Linux, Windows, MacOSX, and for mobile phones: Android, iPhone, Blackberry.
OpenSIPS (Open SIP Server) is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. OpenSIPS, as a SIP server, is the core component of any SIP-based VoIP solution. With a very flexible and customizable routing engine, OpenSIPS ‘unifies voice, video, IM and presence services in a highly efficient way, thanks to its scalable (modular) design.
* SIP registrar server
* SIP router / proxy (lcr, dynamic routing, dialplan features)
* SIP redirect server
* SIP presence agent
* SIP back-to-back User Agent
* SIP IM server (chat and end-2-end IM)
* SIP to SMS gateway (bidirectional)
* SIP to XMPP gateway for presence and IM (bidirectional)
* SIP load-balancer or dispatcher
* SIP front end for gateways/asterisk
* SIP NAT traversal unit
* SIP application server
Cipango is a SIP Servlets extension to the popular Jetty HTTP Servlet engine. Cipango/Jetty is then a convergent SIP/HTTP Application Server compliant with both SIP Servlets 1.1 and HTTP Servlets 2.5 standards. It also features a Diameter extension to develop IMS applications.
OpenVBX allows developers to build voice and SMS applications for business, such as toll free phone numbers, call forwarding, voicemail, visual voicemail, voicemail transcriptions, and auto-attendants. It’s like Google Voice, but open source and for business.
FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools.