2007-11-05 21:50:28 -05:00
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# $OpenBSD: Makefile,v 1.3 2007/11/06 02:50:28 deanna Exp $
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import pjsua 0.7.0
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@
2007-10-27 00:34:23 -04:00
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COMMENT= lightweight sip client
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DISTNAME= pjproject-0.7.0
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2007-11-05 21:50:28 -05:00
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PKGNAME= pjsua-0.7.0p1
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import pjsua 0.7.0
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@
2007-10-27 00:34:23 -04:00
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CATEGORIES= telephony
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HOMEPAGE= http://www.pjsip.org
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MASTER_SITES= http://www.pjsip.org/release/0.7.0/
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MAINTAINER= Deanna Phillips <deanna@openbsd.org>
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WANTLIB= c crypto m ossaudio pthread ssl
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USE_GMAKE= Yes
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CONFIGURE_STYLE= gnu
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# GPL
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PERMIT_PACKAGE_CDROM= Yes
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PERMIT_PACKAGE_FTP= Yes
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PERMIT_DISTFILES_CDROM= Yes
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PERMIT_DISTFILES_FTP= Yes
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post-configure:
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perl -pi -e's,%%LOCALBASE%%,${LOCALBASE},' \
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${WRKSRC}/build/os-openbsd.mak
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cd ${WRKSRC} && ${GMAKE} dep
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CFLAGS+=-DNDEBUG
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NO_REGRESS=Yes
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do-install:
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cp ${WRKSRC}/pjsip-apps/bin/pjsua* ${PREFIX}/bin/pjsua
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.include <bsd.port.mk>
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