Flexisip!

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* Overview * Features * Downloads * Documentation Flexisip is a SIP proxy server implementation compliant to RFC 3261 The%20source%20code%20is%20licensed%20under%20GPLv2.%20ITU%20G729%20Annex%20A/B%20were%20offically%20released%20October/November%201996%20 (https:/www.itu.int/rec/T-REC-G.729
,%20hence%20all%20patents%20covering%20these%20specifications%20shall%20have%20expired%20in%20November%202016. RFC 3261 , written in C++11.

Flexisip alternatives

  • FreeSWITCH

  • FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools.

    tags: voice-chat pbx zrtp

  • kamailio

  • Kamailio® (successor of former OpenSER and SER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second. Among features: asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video); WebSocket support for WebRTC; IPv4 and IPv6; SIMPLE instant messaging and presence with embedded XCAP server and MSRP relay; asynchronous operations; IMS extensions; ENUM; DID and least cost routing; load balancing; routing fail-over; accounting, authentication and authorization; support for many backend systems such as MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, Memcached; XMLRPC control interface, SNMP monitoring. It can be used to build large VoIP servicing platforms or to scale up SIP-to-PSTN gateways, PBX systems or media servers like Asterisk™, FreeSWITCH™ or SEMS. Kamailio and the SIP Express Router (SER) teamed up for the integration of the two applications and new development. »

    tags: sip