30d0a555e3
pjsua is an open source command line SIP user agent that is used as the reference implementation for PJSIP and PJMEDIA. It has many features, such as: * Mutiple identities/account registrations * Concurrent calls and conference (unlimited number, but only up to 254 sources can be mixed to a single destination) * Call features: call hold, call transfer (attended or unattended, with or without refersub). * SIP Presence/SIMPLE with PIDF and XPIDF support. PUBLISH support. * Instant messaging and message composing indication * DTMF digits transmission/receipt (RFC 2833) * WAV file playing, streaming, and recording. * Accoustic echo cancellation (AEC). * Auto-answer, auto-play file, auto-loop RTP * Support SIP UDP, TCP, and TLS transports. Support for DNS SRV resolution. * NAT traversal with rport and STUN. * Tone generation. * Codecs: PCMA, PCMU, GSM, Speex (including wideband/16KHz and ultra-wideband/32KHz), L16 (8-48KHz, mono or stereo), and iLBC. * Adaptive jitter buffer, adaptive silence detection, and packet lost concealment audio features. With lots of testing and help from todd@, sthen@, jakemsr@, jolan@ and Benny Prijono. ok todd@ sthen@ |
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.. | ||
app_conference | ||
asterisk | ||
asterisk-native-sounds | ||
asterisk-sounds | ||
chan_unistim | ||
fobbit | ||
libosip2 | ||
ohphone | ||
p5-asterisk | ||
p5-Net-SIP | ||
pjsua | ||
siproxd | ||
Makefile |