[needs-packaging] FreeSwitch - telephony exchange media gateway

URL:
http://www.freeswitch.org/

Description:
FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a simple switching engine, a PBX, a media gateway or a media server to host IVR applications using simple scripts or XML to control the callflow.

We support various communication technologies such as SIP, H.323, IAX2 and GoogleTalk making it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk.

FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.

FreeSWITCH supports both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future. The voice channels and the conference bridge module all can operate at 8, 16 or 32 kilohertz and can bridge channels of different rates.

FreeSWITCH builds natively and runs standalone on several operating systems including Windows, Max OS X, Linux, BSD and Solaris on both 32 and 64 bit platforms.

Our developers are heavily involved in open source and have donated code and other resources to other telephony projects including openSER, sipX, The Asterisk Open Source PBX and Call Weaver.

Possible Uses ;
Rating & Routing Server, Transcoding B2BUA, IVR & Announcement Server, Conference Server, Voicemail Server, SBC (Session Border Controller), Basic Topology Hiding Session Border Controller, Zaptel and Sangoma Hardware Support (Analog and PRI)

Features ;
Centralized User/Domain Directory (directory.xml), Nano Second CDR granularity, Call recording (In Stereo caller/callee left/right), High Performance Multi-Threaded Core engine, Configuration via CURL to your http server (xml_curl), XML Config files for easy parsing, Protocol Agnostic, Configurable RFC2833 Payload type, Inband DTMF generation and detection, Software based Conference (no hardware requirement), Wideband Conferencing, Media / No Media modes, Proper ENUM/ISN dialing built in, Detailed CDR in XML, Radius CDR, Subscription server, Shared Line Appearances, Bridged Line Appearances, Enterprise/Carrier grade Eventing Engine. (XML Events, Name Value Events, Multicast Events), Loadable File formats and streaming, Stream to Shoutcast, Multi-lingual Speech Phrase Interface, ASR/TTS support (native and via MRCP), Basic IP/PBX features, Automated Attendant, Custom Ring Back Tones, XML RPC support, Multiple format CDR's supported, SQL Engine provides session persistence, Thread Isolation, Parallel Hunting, Serial Hunting, Mozilla Public License, Support, Paid support available, Free support via IRC & e-mail

Many supported codecs ;
G.722 (wideband), G.711, G.726 (16k,24k,32k,48k) AAL2 and RFC3551, G.723.1 (passthru), G.729 (passthru), AMR (passthru), iLBC, speex (narrow and wideband), lpc10, DVI4 (ADPCM) 8khz and 16khz

Applications ;
Voicemail
   Multitenancy - Enterprise/Carrier configuration, Time of Day Greetings, Urgent Message Tagging, EMail Delivery, Playback and Rerecord messages before delivery. Keys are templates so you can rearrange to fit your needs, Callback support from inside voicemail, Podcast of Voicemail (RSS), Message Waiting Indicator (MWI) , Support for Queues (via mod_fifo), Parking (via mod_fifo), Conference, Software based Conferencing without any hardware requirements, o Wideband conferences. Multiple on-demand or scheduled conferences with entry/exit announcements, Play files into the conference or a single member, Relationships, TTS integration, Transfers, Outbound Calling, Configurable Key Lay, Volume, Gain and Energy level per call, Bridge to Conference transition, Multi Party outbound dialing. , RSS Reader

Protocols
SIP
UDP, TCP and TLS transports for full sip compliance, SIP Session timers, RTP Timers, SRTP via SDES (works with polycom, snom and grandstream), Blind SIP Registration, STUN Support, Jitter buffer, NAT Support, Distributed sip registrations, Late Codec Negotiation, Multiple sip registrations per user account., Multitenancy - Multiple sip UAs, o SIP Reinvites. Can act as an SBC (session border controller), Manage Presence, SIP/SIMPLE (can gateway to other chat protocols), SIP Multicast Paging support for Linksys and Snom, Intercom/AutoAnswer support, Call features like Call Hold(Re-INVITE), Blind Transfer(REFER), Call Forward(302) etc.

IAX (Via a modified libiax2.)

Jingle
Interop with GoogleTalk and Telepathy

h.323 (Currently only supports H.323 via the Woomera protocol. This should change soon.)

License: Open Source