is a telephony library (pjsip) with some sample applications; pjsua
itself is one of those samples. Previously the port built everything
but only installed the application; newer versions of Asterisk require
the libraries, so I've converted to multi-packages, providing a new
pjsip package with the libs, and retaining pjsua as a separate package
providing the CLI SIP client.
Taking maintainership with the agreement of chrisz@.
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@
