Audience
Companies searching for a signaling SIP server solution to manage and improve their communication
About OpenSIPS
OpenSIPS is an open-source SIP proxy/server for voice, video, IM, presence, and any other SIP extensions. OpenSIPS is a multi-functional, multi-purpose signaling SIP server used by carriers, telecoms or ITSPs for solutions like Class4/5 residential platforms, trunking/wholesale, enterprise/virtual PBX solutions, session border controllers, application servers, front-end load balancers, IMS platforms, call centers, and many others. OpenSIPS is recommended for any kind of SIP scenario/service by the high throughput, tens of thousands of CPS, millions of simultaneous calls (see official tests), the flexibility of routing and integration, routing scripts for implementing custom routing logics, several interfacing APIs. The effective application building, more than 120 modules to provide features, for SIP handling, backend operations, integration, the routing logic. OpenSIPS 3.2 release was the subject of a security audit, making everything safer and more secure for everybody.
Pricing
Starting Price:
Free
Free Version:
Free Version available.
Company Information
OpenSIPS
United States
opensips.org
Videos and Screen Captures
Signalmash
Discover Signalmash, the ultimate Communication Platform as a Service (CPaaS) solution, offering a versatile suite of communication APIs designed to cater to the evolving needs of businesses. Our offerings encompass:
SMS API: Unlock enhanced customer interaction capabilities through a range of text-based communication functions, including alerts, authentication, promotional outreach, surveys, appointment scheduling, and nurturing customer connections.
Flexible SIP Trunking: Simplify the establishment of communication channels with customizable SIP trunking solutions, empowering businesses with tailored contact center setups, personalized call routing options, and encryption for secure data transmission.
VoIP PBX Solutions: Optimize your internal communication infrastructure with Signalmash's VoIP PBX solutions, streamlining the setup and administration of private branch exchange systems, featuring intuitive call routing, voicemail management, conference calling and more.
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Kamailio
Kamailio® (successor of former OpenSER and SER) is an open-source SIP server released under GPLv2+, able to handle thousands of call setups per second. Kamailio can be used to build large platforms for VoIP and real-time communications, presence, WebRTC, Instant messaging, and other applications. Moreover, it can be easily used for scaling up SIP-to-PSTN gateways, PBX systems, or media servers like Asterisk™, FreeSWITCH™, or SEMS. Among the powerful features, are asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video, text); WebSocket support for WebRTC; IPv4 and IPv6; simple instant messaging and presence with embedded XCAP server and MSRP relay; asynchronous operations; IMS extensions for VoLTE; ENUM; DID and least cost routing; load balancing; routing fail-over; accounting, authentication and authorization; support for many backend systems such as MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, Memcached.
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FreeSWITCH
Everything you need to know to dive into the FreeSWITCH platform. FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a Raspberry PI to a multi-core server, FreeSWITCH can unlock the telecommunications potential of any device. Combined with our hosted cloud platform, SignalWire, FreeSWITCH can interconnect with the outside world and scale to any size. The Pre-built virtual machine has both a written and video how-to to help you get started. All you need is a copy of virtualbox and you can be off and running in minutes. If you are an experienced developer you can also follow the instructions below to compile the FreeSWITCH platform from source among other manual installation options. If you already have a spare Debian instance handy, you can follow the link below to get detailed instructions on how to get our packages installed.
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OpenPBX
It is an easy-to-install and configure PBX software platform that is fully supported by our telephony hardware. It is suitable for small to medium businesses (SMB) that require a sophisticated phone system usually found in larger enterprises but without the price tag! OpenPBX harnesses the ease of use and efficiency of the desktop PC and combines it with the plain old telephone handset to give you advanced features and the ultimate control and use of your communication system. Whether you need to connect to primary rate (E1/T1) trunk lines or to standard POTS lines, Voicetronix has suitable line interface hardware. Our OpenPBX Do-it-Yourself (DIY) option offers customers the ability to build their own phone system using commodity PC servers running Linux and analog telephone handsets. Configure and maintain your own phone system, eliminating the need for a technician. Use your web browser to navigate, select and listen to voicemail messages.
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Product Details
Platforms Supported
Linux
On-Premises
Training
Documentation
Webinars
Videos
Support
Online