pjproject: sync patches with asterisk 20.5.2

[feed/telephony.git] / libs / pjproject / patches / 0004-config_site.patch

diff --git a/libs/pjproject/patches/0004-config_site.patch b/libs/pjproject/patches/0004-config_site.patch
index 7e1c897062be56c9031c223a435d4acda68cfdd4..d9d67aecfeb873af4a4b1d9aa4fde6c160bf68d9 100644 (file)
--- a/libs/pjproject/patches/0004-config_site.patch
+++ b/libs/pjproject/patches/0004-config_site.patch
@@ -1,6 +1,6 @@
 --- /dev/null
 +++ b/pjlib/include/pj/config_site.h
-@@ -0,0 +1,79 @@
+@@ -0,0 +1,91 @@
 +/*
 + * Asterisk config_site.h
 + */
@@ -58,19 +58,19 @@
 +  Enabling it will result in SEGFAULTS when URIs containing escape sequences are encountered.
 +*/
 +#undef PJSIP_UNESCAPE_IN_PLACE
-+#define PJSIP_MAX_PKT_LEN                     32000
++#define PJSIP_MAX_PKT_LEN                     65535
 +
 +#undef PJ_TODO
 +#define PJ_TODO(x)
 +
 +/* Defaults too low for WebRTC */
-+#define PJ_ICE_MAX_CAND 32
++#define PJ_ICE_MAX_CAND 64
 +#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_ATTR   (PJMEDIA_MAX_SDP_FMT*3 + 4)
 +#define       PJMEDIA_MAX_SDP_MEDIA   16
 +
 +/*
@@ -80,3 +80,15 @@
 + */
 +#define PJSIP_TCP_KEEP_ALIVE_INTERVAL 0
 +#define PJSIP_TLS_KEEP_ALIVE_INTERVAL 0
++
++#define PJSIP_TSX_UAS_CONTINUE_ON_TP_ERROR 0
++#define PJ_SSL_SOCK_OSSL_USE_THREAD_CB 0
++#define PJSIP_AUTH_ALLOW_MULTIPLE_AUTH_HEADER 1
++
++/*
++ * The default is 32 with 8 being used by pjproject itself.
++ * Since this value is used in invites, dialogs, transports
++ * and subscriptions as well as the global pjproject endpoint,
++ * we don't want to increase it too much.
++ */
++#define PJSIP_MAX_MODULE 38