Kamailio (formerly OpenSER) is a high-performance SIP (RFC3261) server with a flexible architecture and many extensions. The server implements proxy, registrar, redirect, and location SIP/VoIP services. It has support for UDP, TCP, TLS, and SCTP transport layers, DNSsec, ENUM, AAA via database, RADIUS, DIAMETER, gateways to SMS and XMPP, least cost routing, load balancing, NAT traversal, and call processing language. Kamailio implements SIMPLE presence and instant messaging extensions, and includes an embedded XCAP server and MSRP relay. It can be also used as a routing SIP sever for WebRTC via WebSocket.
PhoNetInfo retrieves detailed phone and network information. It runs on Symbian "Belle", S^3/"Anna", and S60 5th and 3rd edition mobile phones. Information about the following topics can be retrieved: Phone manufacturer and model, firmware version, battery level, WLAN and bluetooth MAC, bluetooth device class, IMSI, IMEI, subscriber ID, charger status, running tasks, active profile settings, network mode, network signal strength and cell ID, network name and ID, network country code and registration status, CPU speed, CPU type and architecture, size of RAM and ROM, time since last reboot, and more. All information can be saved to a file.
PhoNetInfo for WP8 retrieves detailed technical phone and network information such as firmware version, device name, battery status, power saving status, network operator, roaming status, WiFi/WAP interface names, number of CPUs, CPU architecture, CPU features, CPU page size, high performance timer frequency, time since last device reboot, RAM, etc.
Yate is a next-generation telephony engine. While currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. It supports SIP, H.323, IAX, MGCP, Jingle, Jabber, E1, T1, analogic, robbed bit, ISDN PRI, BRI, and SS7. YateClient is an Instant Messenger and Voice application for Jabber/Jingle IM, SIP, H.323, and IAX VoIP protocols.
Ip phone Scanning Made Easy (ISME) scans a VOIP environment, adapts to enterprise VOIP, and exploits the possibilities of being connected directly to an IP Phone VLAN. It seeks to get the phone's configuration file directly from a TFTP server, enable SIP/SIPS (TCP/UDP), communicate with an embedded Web server and Web server banner, identify the editor by MAC address, and identify potential default login/password combinations which should be changed.
The GNU Gatekeeper is a free H.323 gatekeeper based on the OpenH323 project. You can use it to manage a Voice-over-IP network and let endpoints (e.g., Netmeeting) communicate through symbolic names. It also has an external interface for billing and other applications. It runs on a number of Unix versions (including Linux and Solaris) and Windows.
Oreka is an enterprise telephony recording and retrieval system with a Web-based user interface. It supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP, and audio sound devices, and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems, such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc. It is being used in call centers and contact centers for quality monitoring (QM) purposes.
GNU SIP Witch is a secure peer-to-peer VoIP server. Calls can be made even behind NAT firewalls, and without requiring service providers. SIP Witch can be used on the desktop to create bottom-up secure calling networks and as a free software alternative to Skype. It can also be used as a stand-alone SIP-based office telephone server, or to create secure VoIP networks for an existing IP-PBX such as Asterisk, FreeSWITCH, or Yate.