pjproject: update config_site.h

Sync asterisk settings, see [0].

[0] https://github.com/asterisk/asterisk/blob/master/third-party/pjproject/patches/config_site.h

Signed-off-by: Andre Heider <a.heider@gmail.com>

diff --git a/libs/pjproject/Makefile b/libs/pjproject/Makefile
index 2f35d721847c1438296eb8ac3cf5e4b28cf32060..c9dffee81f49e301245fad4e6897915cdb07c513 100644 (file)
--- a/libs/pjproject/Makefile
+++ b/libs/pjproject/Makefile
@@ -11,7 +11,7 @@ include $(TOPDIR)/rules.mk
 
 PKG_NAME:=pjproject
 PKG_VERSION:=2.7.2
-PKG_RELEASE:=2
+PKG_RELEASE:=3
 
 PKG_SOURCE:=pjproject-$(PKG_VERSION).tar.bz2
 PKG_SOURCE_URL:=http://www.pjsip.org/release/$(PKG_VERSION)

diff --git a/libs/pjproject/patches/150-config_site.patch b/libs/pjproject/patches/150-config_site.patch
index 58051374b0dcf068b0159cab1fd5503ff89a0167..7e1c897062be56c9031c223a435d4acda68cfdd4 100644 (file)
--- a/libs/pjproject/patches/150-config_site.patch
+++ b/libs/pjproject/patches/150-config_site.patch
@@ -1,6 +1,6 @@
 --- /dev/null
 +++ b/pjlib/include/pj/config_site.h
-@@ -0,0 +1,92 @@
+@@ -0,0 +1,79 @@
 +/*
 + * Asterisk config_site.h
 + */
@@ -8,37 +8,16 @@
 +#include <sys/select.h>
 +
 +/*
-+ * Since both pjproject and asterisk source files will include config_site.h,
-+ * we need to make sure that only pjproject source files include asterisk_malloc_debug.h.
-+ */
-+
-+/* #if defined(MALLOC_DEBUG) && !defined(_ASTERISK_ASTMM_H)
-+ * #include "asterisk_malloc_debug.h"
-+ * #endif
-+ */
-+
-+/*
 + * Defining PJMEDIA_HAS_SRTP to 0 does NOT disable Asterisk's ability to use srtp.
 + * It only disables the pjmedia srtp transport which Asterisk doesn't use.
 + * The reason for the disable is that while Asterisk works fine with older libsrtp
 + * versions, newer versions of pjproject won't compile with them.
 + */
-+
-+/*
-+ * This doesn't disable SRTP completely, so we have to keep using the external
-+ * libsrtp, otherwise pjsip would just build the internal one.
-+ */
-+
 +#define PJMEDIA_HAS_SRTP 0
 +
-+/*
-+ * Defining PJMEDIA_HAS_WEBRTC_AEC to 0 does NOT disable Asterisk's ability to use
-+ * webrtc.  It only disables the pjmedia webrtc transport which Asterisk doesn't use.
-+ */
-+#define PJMEDIA_HAS_WEBRTC_AEC 0
-+
 +#define PJ_HAS_IPV6 1
 +#define NDEBUG 1
++
 +#define PJ_MAX_HOSTNAME (256)
 +#define PJSIP_MAX_URL_SIZE (512)
 +#ifdef PJ_HAS_LINUX_EPOLL
@@ -79,7 +58,7 @@
 +  Enabling it will result in SEGFAULTS when URIs containing escape sequences are encountered.
 +*/
 +#undef PJSIP_UNESCAPE_IN_PLACE
-+#define PJSIP_MAX_PKT_LEN                     6000
++#define PJSIP_MAX_PKT_LEN                     32000
 +
 +#undef PJ_TODO
 +#define PJ_TODO(x)
@@ -93,3 +72,11 @@
 +#define       PJMEDIA_MAX_SDP_BANDW   4
 +#define       PJMEDIA_MAX_SDP_ATTR   (PJMEDIA_MAX_SDP_FMT*2 + 4)
 +#define       PJMEDIA_MAX_SDP_MEDIA   16
++
++/*
++ * Turn off the periodic sending of CRLNCRLN.  Default is on (90 seconds),
++ * which conflicts with the global section's keep_alive_interval option in
++ * pjsip.conf.
++ */
++#define PJSIP_TCP_KEEP_ALIVE_INTERVAL 0
++#define PJSIP_TLS_KEEP_ALIVE_INTERVAL 0