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pjproject: update config_site.h

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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>
This commit is contained in:
Andre Heider 2018-09-29 09:23:28 +02:00
parent cef38a1293
commit ef17798ecf
2 changed files with 12 additions and 25 deletions
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2
libs/pjproject/Makefile
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@ -11,7 +11,7 @@ include $(TOPDIR)/rules.mk
PKG_NAME:=pjproject PKG_NAME:=pjproject
PKG_VERSION:=2.7.2 PKG_VERSION:=2.7.2
PKG_RELEASE:=2 PKG_RELEASE:=3
PKG_SOURCE:=pjproject-$(PKG_VERSION).tar.bz2 PKG_SOURCE:=pjproject-$(PKG_VERSION).tar.bz2
PKG_SOURCE_URL:=http://www.pjsip.org/release/$(PKG_VERSION) PKG_SOURCE_URL:=http://www.pjsip.org/release/$(PKG_VERSION)

35
libs/pjproject/patches/150-config_site.patch
View file

@ -1,6 +1,6 @@
--- /dev/null --- /dev/null
+++ b/pjlib/include/pj/config_site.h +++ b/pjlib/include/pj/config_site.h
@@ -0,0 +1,92 @@ @@ -0,0 +1,79 @@
+/* +/*
+ * Asterisk config_site.h + * Asterisk config_site.h
+ */ + */
@ -8,37 +8,16 @@
+#include <sys/select.h> +#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. + * 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. + * 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 + * The reason for the disable is that while Asterisk works fine with older libsrtp
+ * versions, newer versions of pjproject won't compile with them. + * 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 +#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 PJ_HAS_IPV6 1
+#define NDEBUG 1 +#define NDEBUG 1
+
+#define PJ_MAX_HOSTNAME (256) +#define PJ_MAX_HOSTNAME (256)
+#define PJSIP_MAX_URL_SIZE (512) +#define PJSIP_MAX_URL_SIZE (512)
+#ifdef PJ_HAS_LINUX_EPOLL +#ifdef PJ_HAS_LINUX_EPOLL
@ -79,7 +58,7 @@
+ Enabling it will result in SEGFAULTS when URIs containing escape sequences are encountered. + Enabling it will result in SEGFAULTS when URIs containing escape sequences are encountered.
+*/ +*/
+#undef PJSIP_UNESCAPE_IN_PLACE +#undef PJSIP_UNESCAPE_IN_PLACE
+#define PJSIP_MAX_PKT_LEN 6000 +#define PJSIP_MAX_PKT_LEN 32000
+ +
+#undef PJ_TODO +#undef PJ_TODO
+#define PJ_TODO(x) +#define PJ_TODO(x)
@ -93,3 +72,11 @@
+#define PJMEDIA_MAX_SDP_BANDW 4 +#define PJMEDIA_MAX_SDP_BANDW 4
+#define PJMEDIA_MAX_SDP_ATTR (PJMEDIA_MAX_SDP_FMT*2 + 4) +#define PJMEDIA_MAX_SDP_ATTR (PJMEDIA_MAX_SDP_FMT*2 + 4)
+#define PJMEDIA_MAX_SDP_MEDIA 16 +#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