Got Training? Opensips Summit: 05/2018 – Amsterdam Who am I? Community Liaison Founder / CEO What is FreeSWITCH? ● FreeSWITCH is an open source carrier-grade telephony platform implemented as a back-to-back user agent. ● Because of this design it can perform a great number of diferent tasks from a PBX to transit switch, text and speech conversions, audio and video conferencing, and an extensive feature-set ● Easy-to-install WebRTC implementation, eliminating need for auxiliary endpoints. ● Modular design, mods and applications augment FreeSWITCH with additional functionalities. Create your own custom modules. ● Supports many common languages, codes, file formats, and protocols. Fundamentals ● Threaded Model ● Stable Protected Core ● Dynamic Modules ● Detailed Event and Logger System ● Hooks into Everything Features ● WebRTC support ● Enterprise/Carrier grade Eventing Engine. (XML Events, Name Value Events, Multicast Events) ● Centralized User/Domain Directory (directory.xml) ● Loadable File formats and streaming ● Nano Second CDR granularity ● Stream to and play from Shoutcast and Icecast ● Call recording (In Stereo caller/callee lef/right) ● Multi-lingual Speech Phrase Interface ● High Performance Multi-Threaded Core engine ● ASR/TTS support (native and via MRCP) ● Configuration via cURL to your HTTP server (mod_xml_curl). ● Basic IP/PBX features ● XML Config files for easy parsing. ● Automated Attendant ● Protocol Agnostic ● Custom Ring Back Tones (Early Media) ● ZRTP support for transparent RTP based key exchange and ● XML-RPC support encryption ● Multiple format CDRs supported ● Configurable RFC 2833 Payload type ● SQL Engine provides session persistence ● Inband DTMF generation and detection. ● Thread Isolation ● ● Sofware based Conference (no hardware requirement) Parallel Hunting ● Serial Hunting ● Wideband Conferencing ● Mozilla Public License ● Media / No Media modes ● Support Applications ● Support for Queues (via mod_fifo or mod_callcenter) Conference ● Parking (via mod_fifo) ● RSS Reader ● Sofware based Conferencing without any hardware requirements. ● Fax endpoint, gateway and passthrough mode. ● Wideband conferences. – T.30 (G.711) Audio Fax (via mod_spandsp) formerly known as ● Multiple on-demand or scheduled conferences with entry/exit mod_fax. announcements – T.38 faxing (gateway, endpoint and passthrough) ● Play files into the conference or a single member. ● Relationships Voicemail ● TTS integration ● Transfers ● Multitenancy - Enterprise/Carrier configuration ● Outbound Calling ● Time of Day Greetings ● Configurable Key Lay ● Urgent Message Tagging ● ● E-mail Delivery Volume, Gain and Energy level per call. ● Playback and Rerecord messages before delivery. ● Bridge to Conference transition ● Keys are templates so you can rearrange to fit your needs. ● Multi Party outbound dialing. ● Callback support from inside voicemail. ● RFC 4579 SIP CC Conferencing for UAs ● Podcast of Voicemail (RSS) ● Automatic or on-demand recording ● Message Waiting Indicator (MWI) Supported Codecs ● PCMU – G.711 µ-law ● DVI4 (IMA ADPCM) ● PCMA – G.711 A-law ● BroadVoice ● G.722 ● SILK ● G.722.1 ● Speex (narrow and wideband) ● G.722.1c ● Codec2 ● G.726 ● LPC-10 ● G.726 with AAL2 packing ● AMR (passthrough) ● G.729 (passthrough) ● iSAC ● G.729 (licensed) ● VP8 ● GSM ● VP9 ● CELT ● iLBC Supported Formats mod_av mod_native_file mod_shout mod_sndfile pvf av AAL2-G726-16 mp3 adpcm r16 m4a AAL2-G726-24 mpga aiff r24 mkv AAL2-G726-32 shout al r32 mov AAL2-G726-40 shouts alaw r8 mp4 DVI4 au raw rtmp G722 mod_png avr rf64 rtsp G723 png caf sd2 webm G726-16 flac sds G726-24 mod_tone_stream gsm sf mod_dptools G726-32 silence_stream htk ul file G726-40 tone_stream iff ulaw file_string G729 mat voc GSM mod_local_stream mpc vox mod_httapi L16 local_stream oga w64 http LPC oga wav https PCMA mod_fsv ogg wve PCMU fsv paf xi Supported Protocols SIP with mod_sofia Jitter buffer Call features like Call Hold (Re- INVITE), Blind Transfer UDP, TCP, SCTP and TLS Distributed SIP registrations (REFER), Call Forward (302), transports for full SIP compliance. Late Codec Negotiation etc. SIP v.2.0 (RFC 3261) Multiple SIP registrations per user. Interop with Google Talk and IPv6 Support Multi-tenancy - Multiple SIP UAs Telepathy SIP Session timers Manage Presence H.323 with mod_opal ( opalvoip.org) RTP Timers SIP/SIMPLE (can gateway to other H.323 with mod_h323 ( RFC 3263 (SRV and NAPTR) chat protocols) www.h323plus.org) Blind SIP Registration SIP Multicast Paging support for Linksys and Snom IAX2 with mod_opal ( STUN Support opalvoip.org) Intercom/AutoAnswer support. NAT Support Supported Languages Event Socket Libary ● Java ● JavaScript (via Google V8 JavaScript engine.) ● Ruby ● Python ● Perl ● PHP ● Lua ● Golang Supported Encryption ● WSS - WebSocket Secure ● SRTP - Secure Real-time Transport Protocol ● ZRTP - Zimmerman Real-time Transport Protocol ● TLS - Transport Layer Security ● SIPS - Secure Session Initiation Protocol ● SDES - Session Description Protocol Security Descriptions ● DTLS - Datagram Transport Layer Security Reasons to use FreeSWITCH? Flexability Reliability Scalability Common Use Cases ● Rating & Routing Server ● Transcoding B2BUA ● IVR & Announcement Server ● Conference Server ● Voicemail Server ● SBC (Session Border Controller) ● Basic Topology Hiding Session Border Controller ● Fax server ● PBX Why To Integrate ● High Availability ● True Scalability ● Security How To Integrate Special Integrations ● Dispatcher and Load Balancer ● Event Socket Layer Where To Start ● How to install FreeSWITCH ● How to install Verto Communicator ● Registering and making calls between endpoints ● Administer various configuration files Where To Start ● Enable and configure modules ● Video Teleconferencing ● FS_CLI usage for logs, debug, troubleshooting ● Gateway Registration, inbound/outbound calls ● Encryption methods SRTP, ZRTP, and Certificates for SIP/TLS Community Use Cases Who? When? Where? What? Why? Contribute Sign Up Now! We still have additional training spots! Thank you :).