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openbsd-ports/telephony/pjsua/Makefile

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use LIBCXX
2017-04-10 11:46:32 +00:00
# $OpenBSD: Makefile,v 1.40 2017/04/10 11:46:34 sthen Exp $
Update and rework the telephony/pjsua port. The upstream distribution 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@.
2015-10-09 21:28:13 +00:00
COMMENT-main= multimedia communication library (SIP, RTP, ICE, etc)
COMMENT-pjsua= lightweight sip client
V= 2.4.5
fix LIB_DEPENDS-pjsua, problem reported by naddy
2015-10-10 12:30:38 +00:00
REVISION-pjsua= 0
Update and rework the telephony/pjsua port. The upstream distribution 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@.
2015-10-09 21:28:13 +00:00
DISTNAME= pjproject-$V
PKGNAME-main= pjsip-$V
PKGNAME-pjsua= pjsua-$V
EXTRACT_SUFX= .tar.bz2
LIBVER= 0.0
SHARED_LIBS += g7221codec ${LIBVER}
SHARED_LIBS += ilbccodec ${LIBVER}
SHARED_LIBS += pj ${LIBVER}
SHARED_LIBS += pjlib-util ${LIBVER}
SHARED_LIBS += pjmedia-audiodev ${LIBVER}
SHARED_LIBS += pjmedia-codec ${LIBVER}
SHARED_LIBS += pjmedia-videodev ${LIBVER}
SHARED_LIBS += pjmedia ${LIBVER}
SHARED_LIBS += pjnath ${LIBVER}
SHARED_LIBS += pjsip-simple ${LIBVER}
SHARED_LIBS += pjsip-ua ${LIBVER}
SHARED_LIBS += pjsip ${LIBVER}
SHARED_LIBS += pjsua ${LIBVER}
SHARED_LIBS += pjsua2 ${LIBVER}
SHARED_LIBS += resample ${LIBVER}
CATEGORIES= telephony
HOMEPAGE= http://www.pjsip.org/
MASTER_SITES= ${HOMEPAGE}release/$V/
MAINTAINER= Stuart Henderson <sthen@openbsd.org>
# GPLv2+
PERMIT_PACKAGE_CDROM= Yes
WANTLIB += crypto gsm m opencore-amrnb opencore-amrwb portaudio pthread
WANTLIB += speex speexdsp ssl
WANTLIB-pjsua += ${WANTLIB}
WANTLIB-pjsua += c g7221codec ilbccodec pj pjlib-util pjmedia pjmedia-audiodev
WANTLIB-pjsua += pjmedia-codec pjmedia-videodev pjnath pjsip pjsip-simple
WANTLIB-pjsua += pjsip-ua pjsua resample
use LIBCXX
2017-04-10 11:46:32 +00:00
WANTLIB-main = ${WANTLIB} ${LIBCXX}
Update and rework the telephony/pjsua port. The upstream distribution 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@.
2015-10-09 21:28:13 +00:00
MULTI_PACKAGES= -main -pjsua
LIB_DEPENDS-main= audio/gsm \
audio/opencore-amr \
audio/portaudio-svn \
audio/speex
fix LIB_DEPENDS-pjsua, problem reported by naddy
2015-10-10 12:30:38 +00:00
LIB_DEPENDS-pjsua= telephony/pjsua,-main \
Update and rework the telephony/pjsua port. The upstream distribution 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@.
2015-10-09 21:28:13 +00:00
audio/portaudio-svn
# statically linked
BUILD_DEPENDS= security/libsrtp
CONFIGURE_STYLE= autoconf no-autoheader
AUTOCONF_VERSION= 2.69
requires --enable-shared
2016-03-10 11:50:13 +00:00
CONFIGURE_ARGS= --enable-shared \
Update and rework the telephony/pjsua port. The upstream distribution 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@.
2015-10-09 21:28:13 +00:00
--disable-oss \
--disable-video \
--with-external-gsm \
--with-external-pa \
--with-external-speex \
--with-external-srtp
CFLAGS+= -DNDEBUG -DPJ_HAS_IPV6=1 -I${LOCALBASE}/include
CONFIGURE_ENV= CFLAGS="${CFLAGS}" \
LDFLAGS="-L${LOCALBASE}/lib"
CONFIGURE_ENV+= ac_cv_header_uuid_uuid_h=no \
ac_cv_lib_uuid_uuid_generate=no
USE_GMAKE= Yes
MAKE_FLAGS= PJ_VERSION_MAJOR=${LIBVER} TARGET_NAME=openbsd
MODULES+= lang/python # for tests
MODPY_BUILDDEP= No
MODPY_RUNDEP= No
TEST_DEPENDS= ${MODPY_RUN_DEPENDS} \
pjsip-=$V:${BUILD_PKGPATH}
ALL_TARGET= dep all
TEST_TARGET= selftest
# distributed with a wrapper script that runs the real autoconf script
# in ${WRKSRC}/aconfigure; not really compatible with gnu.port.mk so
# just move it into place.
post-extract:
mv ${WRKSRC}/aconfigure ${WRKSRC}/configure
mv ${WRKSRC}/aconfigure.ac ${WRKSRC}/configure.ac
post-install:
rm ${PREFIX}/lib/*.so ${PREFIX}/lib/*-openbsd.a
Update to pjsua-1.5.5 From (new) MAINTAINER Christopher Zimmermann with some modifications by landry@, ajacoutot@, sthen@ and myself. Thanks! OK landry@, ajacoutot@, sthen@
2010-04-02 11:48:24 +00:00
${INSTALL_PROGRAM} ${WRKSRC}/pjsip-apps/bin/pjsua* ${PREFIX}/bin/pjsua
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 04:34:23 +00:00
Update and rework the telephony/pjsua port. The upstream distribution 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@.
2015-10-09 21:28:13 +00:00
pre-test:
ln -fs ${MODPY_BIN} ${WRKDIR}/bin/python
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 04:34:23 +00:00
.include <bsd.port.mk>
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