13 projects tagged "SIP"
Updated 26 Nov 2009
Jingle Nodes
| Pop | 25.73 |
|---|---|
| Vit | 36.07 |
Jingle Nodes is a relay auto-discovery service that provides Jingle Relay Type Candidates that can be used in ICE-UDP and also on RAW-UDP Jingle sessions. Relay candidates can provide NAT traversal for users who don't have STUN/TURN Support, but also for users with STUN/TURN support for whom negotiation failed. It is designed to allow you to communicate freely with your friends without being attached to closed service providers like Skype or telecommunications carriers.
Updated 07 Feb 2010
Open Unified Recording
| Pop | 45.12 |
|---|---|
| Vit | 35.06 |
Open Unified Recording (OUR) is a full featured Linux-based VoIP/SIP call recording engine, indexing, and retrieval system. The system resides on the network and passively captures SIP sessions.
Updated 21 Sep 2011
SIP Inspector
| Pop | 74.01 |
|---|---|
| Vit | 2.34 |
SIP Inspector is a tool to simulate different SIP messages and scenarios. You can create your own SIP signaling scenarios, customize SIP messages, and monitor incoming and outgoing messages. The tool can play RTP streams from a pcap file.
Updated 21 Jun 2010
WikiPBX
| Pop | 37.55 |
|---|---|
| Vit | 1.00 |
WikiPBX is a PBX Web interface for FreeSWITCH. Multiple "accounts" are supported per server instance: each account is effectively a completely independent PBX. Configuration is layered so that XML files go on top of what is stored in the database. This allows you to use a database, but stays out of your way if you choose to use flat files. Extensions, SIP endpoints, and gateways can be configured via a Web interface. Live calls can be viewed, hanged up, and transferred. Call history (CDR records) can be viewed over the Web interface. There is a Web interface for managing IVRs. "Sound clips" can be easily recorded for use in dialplan or IVRs. Audio or text-to-speech can be injected into live calls.
Updated 13 Dec 2010
SEMS
| Pop | 50.85 |
|---|---|
| Vit | 1.46 |
SEMS is a media and application server for SIP based VoIP services. It shows good performance doing basic services like announcements and conference for combination with external application servers. Thanks to its easy-to-use and flexible application development framework and back-to-back user agent support, application logic and media serving can be combined in the same process. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox, and lots of example applications are available. Scripting can be done in Python and a simple state machine description language. Support All commonly used free codecs (including g711, gsm, iLBC, speex, adpcm, and l16) are supported. Other features include wideband, ZRTP encryption, a SIP registrar client, an XMLRPC server/client, and a DIAMETER client.
Updated 22 Dec 2010
ScopServ Communicator
| Pop | 49.75 |
|---|---|
| Vit | 1.46 |
Scopserv Communicator is an audio/video Internet phone and instant messenger that supports some of the most popular VoIP and instant messaging protocols such as SIP, Jabber (XMPP), AIM/ICQ, MSN, Yahoo! Messenger, and a whole lot of other useful features. ScopServ Communicator is based on the SIP Communicator softphone.
Updated 06 May 2013
libre
| Pop | 166.24 |
|---|---|
| Vit | 12.01 |
libre is a generic library for real-time communications with asynchronous I/O support. It is written in portable POSIX source code that conforms to the ANSI C89 and ISO C99 standards. It is robust and fast, with a low memory footprint. It also features RFC compliance and support for IPv4 and IPv6. Protocol implementations include SIP, SDP, RTP/RTCP, BFCP, DNS, and STUN/TURN/ICE.
Updated 18 May 2013
baresip
| Pop | 264.88 |
|---|---|
| Vit | 15.21 |
baresip is a bare-bones SIP user agent. It supports SIP, SDP, RTP/RTCP, and STUN/TURN/ICE, and IPv4 and IPv6, and is RFC-compliant and has portable C89 and C99 source code. A modular plugin architecture provides stdio, cons, and evdev user interfaces, celt, g711, g722, gsm, ilbc, l16, and speex audio codecs, alsa, coreaudio, gst, portaudio, oss, winwav, and mda audio drivers, speex_pp, speex_aec, speex_resamp, and sndfile audio filters, the avcodec video codec, avformat, quicktime, qtcapture, v4l, and v4l2 video sources, sdl, opengl, and x11 video display drivers, and srtp media encoding.
Updated 07 Jun 2011
Billion PBX
| Pop | 15.26 |
|---|---|
| Vit | 27.27 |
Billion PBX is IP phone system that uses the SIP protocol. In addition to voice calls it supports call forwarding, voicemail, callback, and much more, and features simple installation and configuration, reliability, low hardware requirements, and a flexible pricing policy. Its main features are subscriber registration by SIP Registrar and RAS protocols, call routing by SIP and H.323, call authorization, subscriber self-service, Call Data Records (CDR), embedded rating and billing, integration with external billing systems, NAT traversal, IVR, report generation, and a convenient administrative Web interface allowing remote management.
Updated 07 Jun 2011
Billion Softswitch
| Pop | 16.09 |
|---|---|
| Vit | 27.27 |
Billion Softswitch works as a switchboard in SIP/H.323 VoIP networks, proxying both signalling and media streams while performing cross-protocol conversion and even media stream transcoding. It integrates advanced call routing features as well as embedded compact billing system. It supports subscriber registration by SIP Registrar and RAS, call authorization, call data record (CDR) list maintenance, external billing system interfaces, NAT/firewall support, and report generation, and is easy to install and maintain.
