Merge pull request #201 from micmac1/master

New PR: libsrtp2, pjproject and asterisk-13/15
This commit is contained in:
Jiri Slachta 2017-11-15 06:03:07 +02:00 committed by GitHub
commit 4f0567e197
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GPG key ID: 4AEE18F83AFDEB23
6 changed files with 13 additions and 74 deletions

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@ -55,8 +55,8 @@ endef
define Package/libsrtp2/install
$(INSTALL_DIR) $(1)/usr/lib
$(INSTALL_BIN) \
$(PKG_INSTALL_DIR)/usr/lib/libsrtp2.so.* \
$(CP) \
$(PKG_INSTALL_DIR)/usr/lib/libsrtp2.so* \
$(1)/usr/lib/
endef

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@ -10,12 +10,12 @@
include $(TOPDIR)/rules.mk
PKG_NAME:=pjproject
PKG_VERSION:=2.6
PKG_RELEASE:=3
PKG_VERSION:=2.7.1
PKG_RELEASE:=1
PKG_SOURCE:=pjproject-$(PKG_VERSION).tar.bz2
PKG_SOURCE_URL:=http://www.pjsip.org/release/$(PKG_VERSION)
PKG_HASH:=2f5a1da1c174d845871c758bd80fbb580fca7799d3cfaa0d3c4e082b5161c7b4
PKG_HASH:=59fabc62a02b2b80857297cfb10e2c68c473f4a0acc6e848cfefe8421f2c3126
PKG_INSTALL:=1
PKG_FIXUP:=autoreconf
@ -56,6 +56,7 @@ endef
CONFIGURE_ARGS+= \
$(if $(CONFIG_SOFT_FLOAT),--disable-floating-point) \
--disable-bcg729 \
--disable-ext-sound \
--disable-ffmpeg \
--disable-g711-codec \
@ -104,16 +105,16 @@ define Build/InstallDev
$(foreach m,$(PJPROJECT_LIBS),$(CP) $(PKG_INSTALL_DIR)/usr/lib/$(m)* $(1)/usr/lib;)
$(INSTALL_DIR) $(1)/usr/lib/pkgconfig
$(SED) 's|$(TARGET_CFLAGS)||' $(PKG_INSTALL_DIR)/usr/lib/pkgconfig/libpjproject.pc
$(SED) 's|$(TARGET_CFLAGS)||g' $(PKG_INSTALL_DIR)/usr/lib/pkgconfig/libpjproject.pc
$(CP) $(PKG_INSTALL_DIR)/usr/lib/pkgconfig/libpjproject.pc $(1)/usr/lib/pkgconfig
endef
$(eval $(call PJSIPpackage,libpj,libpj,+librt))
$(eval $(call PJSIPpackage,libpjlib-util,libpjlib-util,+libpj +librt))
$(eval $(call PJSIPpackage,libpjmedia,libpjmedia*,+libpj +libpjlib-util +libpjnath +librt +libsrtp))
$(eval $(call PJSIPpackage,libpjmedia,libpjmedia*,+libpj +libpjlib-util +libpjnath +librt +libsrtp2))
$(eval $(call PJSIPpackage,libpjnath,libpjnath,+libpj +libpjlib-util +librt))
$(eval $(call PJSIPpackage,libpjsip-simple,libpjsip-simple,+libpj +libpjlib-util +libpjsip +librt))
$(eval $(call PJSIPpackage,libpjsip-ua,libpjsip-ua,+libpj +libpjlib-util +libpjmedia +libpjsip-simple +libpjsip +librt))
$(eval $(call PJSIPpackage,libpjsip,libpjsip,+libpj +libpjlib-util +librt +libsrtp))
$(eval $(call PJSIPpackage,libpjsip,libpjsip,+libpj +libpjlib-util +librt +libsrtp2))
$(eval $(call PJSIPpackage,libpjsua,libpjsua,+libpj +libpjlib-util +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +librt))
$(eval $(call PJSIPpackage,libpjsua2,libpjsua2,+libpj +libpjlib-util +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +librt +libpjsua))

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@ -1,36 +0,0 @@
From f0c717463d569f87a16f9b014033c8ca8939a7b4 Mon Sep 17 00:00:00 2001
From: Mark Michelson <mmichelson@digium.com>
Date: Thu, 13 Apr 2017 16:59:40 -0500
Subject: [PATCH] Parse zero-length multipart body parts correctly.
The calculation of end_body could result in a negative length being
passed to multipart_body_parse_part().
---
pjsip/src/pjsip/sip_multipart.c | 16 +++++++++-------
1 file changed, 9 insertions(+), 7 deletions(-)
--- a/pjsip/src/pjsip/sip_multipart.c
+++ b/pjsip/src/pjsip/sip_multipart.c
@@ -646,13 +646,15 @@ PJ_DEF(pjsip_msg_body*) pjsip_multipart_
end_body = curptr;
- /* The newline preceeding the delimiter is conceptually part of
- * the delimiter, so trim it from the body.
- */
- if (*(end_body-1) == '\n')
- --end_body;
- if (*(end_body-1) == '\r')
- --end_body;
+ if (end_body > start_body) {
+ /* The newline preceeding the delimiter is conceptually part of
+ * the delimiter, so trim it from the body.
+ */
+ if (*(end_body-1) == '\n')
+ --end_body;
+ if (*(end_body-1) == '\r')
+ --end_body;
+ }
/* Now that we have determined the part's boundary, parse it
* to get the header and body part of the part.

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@ -1,24 +0,0 @@
From b5f0f8868363c482a2c4ce343e3ee6ad256b0708 Mon Sep 17 00:00:00 2001
From: Mark Michelson <mmichelson@digium.com>
Date: Thu, 13 Apr 2017 16:20:07 -0500
Subject: [PATCH] Ensure 2543 transaction key buffer is large enough.
The CSeq method length needs to be factored into the allocated buffer
length. Otherwise, the buffer may not be large enough to accommodate the
entire key.
---
pjsip/src/pjsip/sip_transaction.c | 3 ++-
1 file changed, 2 insertions(+), 1 deletion(-)
--- a/pjsip/src/pjsip/sip_transaction.c
+++ b/pjsip/src/pjsip/sip_transaction.c
@@ -288,7 +288,8 @@ static pj_status_t create_tsx_key_2543(
host = &rdata->msg_info.via->sent_by.host;
/* Calculate length required. */
- len_required = 9 + /* CSeq number */
+ len_required = method->name.slen + /* Method */
+ 9 + /* CSeq number */
rdata->msg_info.from->tag.slen + /* From tag. */
rdata->msg_info.cid->id.slen + /* Call-ID */
host->slen + /* Via host. */

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@ -228,7 +228,6 @@ CONFIGURE_ARGS+= \
--enable-xmldoc
CONFIGURE_VARS += \
ac_cv_lib_srtp2_srtp_init=no \
ac_cv_path_ac_pt_CONFIG_LIBXML2=$(STAGING_DIR)/host/bin/xml2-config
MAKE_FLAGS+= \
@ -415,14 +414,14 @@ $(eval $(call BuildAsterisk13Module,res-musiconhold,MOH,Music On Hold support,,m
$(eval $(call BuildAsterisk13Module,res-parking,Phone Parking,Phone Parking application,,res_parking.conf,res_parking,,))
$(eval $(call BuildAsterisk13Module,res-phoneprov,Phone Provisioning,Phone provisioning application for the asterisk internal http server,,phoneprov.conf,res_phoneprov,,))
$(eval $(call BuildAsterisk13Module,res-pjsip-phoneprov,PJSIP Phone Provisioning,PJSIP Phone Provisioning,+asterisk13-pjsip +asterisk13-res-phoneprov,,res_pjsip_phoneprov_provider,,))
$(eval $(call BuildAsterisk13Module,res-pjproject,Bridge PJPROJECT to Asterisk logging,,+libpj +libpjlib-util +libpjmedia +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +libpjsua +libpjsua2 +libsrtp,pjproject.conf,res_pjproject,,))
$(eval $(call BuildAsterisk13Module,res-pjproject,Bridge PJPROJECT to Asterisk logging,,+libpj +libpjlib-util +libpjmedia +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +libpjsua +libpjsua2 +libsrtp2,pjproject.conf,res_pjproject,,))
$(eval $(call BuildAsterisk13Module,res-realtime,RealTime CLI,RealTime CLI,,,res_realtime,,))
$(eval $(call BuildAsterisk13Module,res-rtp-asterisk,RTP stack,Supports RTP and RTCP with Symmetric RTP support for NAT traversal,+libpjsip +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsua +libpjsua2,rtp.conf,res_rtp_asterisk,,))
$(eval $(call BuildAsterisk13Module,res-rtp-multicast,RTP multicast engine,Multicast RTP Engine,,,res_rtp_multicast,,))
$(eval $(call BuildAsterisk13Module,res-smdi,Provide SMDI,Simple Message Desk Interface capability,,smdi.conf,res_smdi,,))
$(eval $(call BuildAsterisk13Module,res-sorcery,Sorcery data layer,Sorcery backend modules for data access intended for using realtime as backend ,,,res_sorcery_astdb res_sorcery_config res_sorcery_memory res_sorcery_realtime,,))
$(eval $(call BuildAsterisk13Module,res-speech,Speech Recognition API,Support for the Asterisk Generic Speech Recognition API,,,res_speech,,))
$(eval $(call BuildAsterisk13Module,res-srtp,SRTP Support,Secure RTP connection,+libsrtp,,res_srtp,,))
$(eval $(call BuildAsterisk13Module,res-srtp,SRTP Support,Secure RTP connection,+libsrtp2,,res_srtp,,))
$(eval $(call BuildAsterisk13Module,res-stun-monitor,STUN monitoring,resource STUN Monitor,,res_stun_monitor.conf,res_stun_monitor,,))
$(eval $(call BuildAsterisk13Module,res-timing-dahdi,DAHDI Timing Interface,DAHDI timing interface,+asterisk13-chan-dahdi,,res_timing_dahdi,,))
$(eval $(call BuildAsterisk13Module,res-timing-pthread,pthread Timing Interface,POSIX pthreads Timing Interface,,,res_timing_pthread,,))

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@ -228,7 +228,6 @@ CONFIGURE_ARGS+= \
--enable-xmldoc
CONFIGURE_VARS += \
ac_cv_lib_srtp2_srtp_init=no \
ac_cv_path_ac_pt_CONFIG_LIBXML2=$(STAGING_DIR)/host/bin/xml2-config
MAKE_FLAGS+= \
@ -415,14 +414,14 @@ $(eval $(call BuildAsterisk15Module,res-musiconhold,MOH,Music On Hold support,,m
$(eval $(call BuildAsterisk15Module,res-parking,Phone Parking,Phone Parking application,,res_parking.conf,res_parking,,))
$(eval $(call BuildAsterisk15Module,res-phoneprov,Phone Provisioning,Phone provisioning application for the asterisk internal http server,,phoneprov.conf,res_phoneprov,,))
$(eval $(call BuildAsterisk15Module,res-pjsip-phoneprov,PJSIP Phone Provisioning,PJSIP Phone Provisioning,+asterisk15-pjsip +asterisk15-res-phoneprov,,res_pjsip_phoneprov_provider,,))
$(eval $(call BuildAsterisk15Module,res-pjproject,Bridge PJPROJECT to Asterisk logging,,+libpj +libpjlib-util +libpjmedia +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +libpjsua +libpjsua2 +libsrtp,pjproject.conf,res_pjproject,,))
$(eval $(call BuildAsterisk15Module,res-pjproject,Bridge PJPROJECT to Asterisk logging,,+libpj +libpjlib-util +libpjmedia +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +libpjsua +libpjsua2 +libsrtp2,pjproject.conf,res_pjproject,,))
$(eval $(call BuildAsterisk15Module,res-realtime,RealTime CLI,RealTime CLI,,,res_realtime,,))
$(eval $(call BuildAsterisk15Module,res-rtp-asterisk,RTP stack,Supports RTP and RTCP with Symmetric RTP support for NAT traversal,+libpjsip +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsua +libpjsua2,rtp.conf,res_rtp_asterisk,,))
$(eval $(call BuildAsterisk15Module,res-rtp-multicast,RTP multicast engine,Multicast RTP Engine,,,res_rtp_multicast,,))
$(eval $(call BuildAsterisk15Module,res-smdi,Provide SMDI,Simple Message Desk Interface capability,,smdi.conf,res_smdi,,))
$(eval $(call BuildAsterisk15Module,res-sorcery,Sorcery data layer,Sorcery backend modules for data access intended for using realtime as backend ,,,res_sorcery_astdb res_sorcery_config res_sorcery_memory res_sorcery_realtime,,))
$(eval $(call BuildAsterisk15Module,res-speech,Speech Recognition API,Support for the Asterisk Generic Speech Recognition API,,,res_speech,,))
$(eval $(call BuildAsterisk15Module,res-srtp,SRTP Support,Secure RTP connection,+libsrtp,,res_srtp,,))
$(eval $(call BuildAsterisk15Module,res-srtp,SRTP Support,Secure RTP connection,+libsrtp2,,res_srtp,,))
$(eval $(call BuildAsterisk15Module,res-stun-monitor,STUN monitoring,resource STUN Monitor,,res_stun_monitor.conf,res_stun_monitor,,))
$(eval $(call BuildAsterisk15Module,res-timing-dahdi,DAHDI Timing Interface,DAHDI timing interface,+asterisk15-chan-dahdi,,res_timing_dahdi,,))
$(eval $(call BuildAsterisk15Module,res-timing-pthread,pthread Timing Interface,POSIX pthreads Timing Interface,,,res_timing_pthread,,))