Host Communication - SIP Tools

Transit class4 softswitch(SBC) with integrated billing and routing engine and REST API.

Full-featured IPBX solution built atop Asterisk with integrated Web administration interface and REST-ful API.

Troubleshooting and monitoring VoIP calls.

VoIP troubleshooting and monitoring platform.

A lightweight sip proxy, location server, and registrar for a reliable and scalable SIP infrastructure.

OpenSIPS is an Open Source SIP proxy/server for voice, video, IM, presence and any other SIP extensions.

Modular SIP server (registrar/proxy/router/etc).

Scalable open source cross-platform telephony platform.

Web-based open source GUI that controls and manages Asterisk.

Open source project that provides a customizable and flexible web interface to the very powerful and highly scalable multi-platform voice switch called FreeSWITCH.

A complete, modular and scalable SIP server, includes a push gateway, to deliver SIP incoming calls or text messages on mobile device platforms where push notifications are required to receive information when the app is not active in the foreground.

Eqivo implements an API layer on top of FreeSWITCH facilitating integration between web applications and voice/video-enabled endpoints such as traditional phone lines (PSTN), VoIP phones, webRTC clients etc.

VoIP Billing Solution for Freeswitch. It supports prepaid and postpaid billing with call rating and credit control. It also provides many other features.

Easy to use but advanced IP PBX system, VoIP gateway and conference server.