pjproject: add config_site.h

Copied from Asterisk, sets some sane values. For instance it enables
IPv6 support.

Also it disables DEBUG. With debug enabled 'pjproject enables "assert"
functions which can cause Asterisk to crash unexpectedly' (quote from
Asterisk wiki).

Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>

diff --git a/libs/pjproject/patches/150-config_site.patch b/libs/pjproject/patches/150-config_site.patch
new file mode 100644 (file)
index 0000000..5805137
--- /dev/null
+++ b/libs/pjproject/patches/150-config_site.patch
@@ -0,0 +1,95 @@
+--- /dev/null
++++ b/pjlib/include/pj/config_site.h
+@@ -0,0 +1,92 @@
++/*
++ * Asterisk config_site.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.
++ * 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
++#define PJ_IOQUEUE_MAX_HANDLES        (5000)
++#else
++#define PJ_IOQUEUE_MAX_HANDLES        (FD_SETSIZE)
++#endif
++#define PJ_IOQUEUE_HAS_SAFE_UNREG 1
++#define PJ_IOQUEUE_MAX_EVENTS_IN_SINGLE_POLL (16)
++
++#define PJ_SCANNER_USE_BITWISE        0
++#define PJ_OS_HAS_CHECK_STACK 0
++
++#ifndef PJ_LOG_MAX_LEVEL
++#define PJ_LOG_MAX_LEVEL              6
++#endif
++
++#define PJ_ENABLE_EXTRA_CHECK 1
++#define PJSIP_MAX_TSX_COUNT           ((64*1024)-1)
++#define PJSIP_MAX_DIALOG_COUNT        ((64*1024)-1)
++#define PJSIP_UDP_SO_SNDBUF_SIZE      (512*1024)
++#define PJSIP_UDP_SO_RCVBUF_SIZE      (512*1024)
++#define PJ_DEBUG                      0
++#define PJSIP_SAFE_MODULE             0
++#define PJ_HAS_STRICMP_ALNUM          0
++
++/*
++ * Do not ever enable PJ_HASH_USE_OWN_TOLOWER because the algorithm is
++ * inconsistently used when calculating the hash value and doesn't
++ * convert the same characters as pj_tolower()/tolower().  Thus you
++ * can get different hash values if the string hashed has certain
++ * characters in it.  (ASCII '@', '[', '\\', ']', '^', and '_')
++ */
++#undef PJ_HASH_USE_OWN_TOLOWER
++
++/*
++  It is imperative that PJSIP_UNESCAPE_IN_PLACE remain 0 or undefined.
++  Enabling it will result in SEGFAULTS when URIs containing escape sequences are encountered.
++*/
++#undef PJSIP_UNESCAPE_IN_PLACE
++#define PJSIP_MAX_PKT_LEN                     6000
++
++#undef PJ_TODO
++#define PJ_TODO(x)
++
++/* Defaults too low for WebRTC */
++#define PJ_ICE_MAX_CAND 32
++#define PJ_ICE_MAX_CHECKS (PJ_ICE_MAX_CAND * PJ_ICE_MAX_CAND)
++
++/* Increase limits to allow more formats */
++#define       PJMEDIA_MAX_SDP_FMT   64
++#define       PJMEDIA_MAX_SDP_BANDW   4
++#define       PJMEDIA_MAX_SDP_ATTR   (PJMEDIA_MAX_SDP_FMT*2 + 4)
++#define       PJMEDIA_MAX_SDP_MEDIA   16