# The telephony/pjsua port

## pjsua-2.1 – lightweight sip client

## Description

    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.

WWW: http://www.pjsip.org/

Categories:
  telephony

## Library dependencies

- audio/gsm
- audio/opencore-amr
- audio/portaudio-svn
- audio/speex

## Build dependencies

- archivers/bzip2
- devel/autoconf/2.69
- devel/gmake
- devel/metaauto

## Run dependencies