Menu
▾
▴
[Astlinux-commits] SF.net SVN: astlinux:[7816] branches/1.0/package
|
From: <abe...@us...> - 2016-09-03 01:08:10
|
Revision: 7816
http://sourceforge.net/p/astlinux/code/7816
Author: abelbeck
Date: 2016-09-03 01:08:07 +0000 (Sat, 03 Sep 2016)
Log Message:
-----------
asterisk, version bump to 13.11.0 with pjsip version bump to 2.5.5
Modified Paths:
--------------
branches/1.0/package/asterisk/asterisk-13-configure-menuselect-cross-fix.patch
branches/1.0/package/asterisk/asterisk-13-extension-changed-verbosity-chan_sip.patch
branches/1.0/package/asterisk/asterisk.mk
branches/1.0/package/pjsip/asterisk-config_site.h
branches/1.0/package/pjsip/pjsip.mk
Added Paths:
-----------
branches/1.0/package/pjsip/pjsip-0002-Fix-1946-Avoid-deinitialization-of-uninitialized-cli.patch
Removed Paths:
-------------
branches/1.0/package/asterisk/asterisk-13-SILK-CODEC-support.patch
branches/1.0/package/pjsip/pjsip-0001-evsub-Add-APIs-to-add-decrement-an-event-subscriptio.patch
branches/1.0/package/pjsip/pjsip-0002-sip_transport_tcp-tls-Set-factory-on-transports-crea.patch
Deleted: branches/1.0/package/asterisk/asterisk-13-SILK-CODEC-support.patch
===================================================================
--- branches/1.0/package/asterisk/asterisk-13-SILK-CODEC-support.patch 2016-09-02 16:50:57 UTC (rev 7815)
+++ branches/1.0/package/asterisk/asterisk-13-SILK-CODEC-support.patch 2016-09-03 01:08:07 UTC (rev 7816)
@@ -1,346 +0,0 @@
-From 28501051b47e6bb8968bb016abf0b3493c05fa21 Mon Sep 17 00:00:00 2001
-From: Mark Michelson <mmi...@di...>
-Date: Thu, 30 Jun 2016 15:58:53 -0500
-Subject: [PATCH] Update support for SILK format.
-
-This commit adds scaffolding in order to support the SILK audio format
-on calls. Roughly, this is what is added:
-
-* Cached silk formats. One for each possible sample rate.
-* ast_codec structures for each possible sample rate.
-* RTP payload mappings for "SILK".
-
-In addition, this change overhauls the res_format_attr_silk file in the
-following ways:
-
-* The "samplerate" attribute is scrapped. That's native to the format.
-* There are far more checks to ensure that attributes have been
- allocated before attempting to reference them.
-* We do not SDP fmtp lines for attributes set to 0.
-
-These changes make way to be able to install a codec_silk module and
-have it actually work. It also should allow for passthrough silk calls
-in Asterisk.
-
-Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e
----
- include/asterisk/format_cache.h | 8 +++++
- main/codec_builtin.c | 63 ++++++++++++++++++++++++++++++++++++++
- main/format_cache.c | 20 ++++++++++++
- main/rtp_engine.c | 10 ++++++
- res/res_format_attr_silk.c | 64 ++++++++++++++++++++-------------------
- 5 files changed, 134 insertions(+), 31 deletions(-)
-
-diff --git a/include/asterisk/format_cache.h b/include/asterisk/format_cache.h
-index 9f4e06a..ff03bb4 100644
---- a/include/asterisk/format_cache.h
-+++ b/include/asterisk/format_cache.h
-@@ -224,6 +224,14 @@ extern struct ast_format *ast_format_t140_red;
- extern struct ast_format *ast_format_none;
-
- /*!
-+ * \brief Built-in SILK format.
-+ */
-+extern struct ast_format *ast_format_silk8;
-+extern struct ast_format *ast_format_silk12;
-+extern struct ast_format *ast_format_silk16;
-+extern struct ast_format *ast_format_silk24;
-+
-+/*!
- * \brief Initialize format cache support within the core.
- *
- * \retval 0 success
-diff --git a/main/codec_builtin.c b/main/codec_builtin.c
-index d3f6517..1d329bc 100644
---- a/main/codec_builtin.c
-+++ b/main/codec_builtin.c
-@@ -772,6 +772,65 @@ static struct ast_codec t140 = {
- .type = AST_MEDIA_TYPE_TEXT,
- };
-
-+static int silk_samples(struct ast_frame *frame)
-+{
-+ /* XXX This is likely not at all what's intended from this callback. However,
-+ * since SILK is variable bit rate, I have no idea how to take a frame of data
-+ * and determine the number of samples present. Instead, we base this on the
-+ * sample rate of the codec and the expected number of samples to receive in 20ms.
-+ * In testing, this has worked just fine.
-+ */
-+ return ast_format_get_sample_rate(frame->subclass.format) / 50;
-+}
-+
-+static struct ast_codec silk8 = {
-+ .name = "silk",
-+ .description = "SILK Codec (8 KHz)",
-+ .type = AST_MEDIA_TYPE_AUDIO,
-+ .sample_rate = 8000,
-+ .minimum_ms = 20,
-+ .maximum_ms = 100,
-+ .default_ms = 20,
-+ .minimum_bytes = 160,
-+ .samples_count = silk_samples
-+};
-+
-+static struct ast_codec silk12 = {
-+ .name = "silk",
-+ .description = "SILK Codec (12 KHz)",
-+ .type = AST_MEDIA_TYPE_AUDIO,
-+ .sample_rate = 12000,
-+ .minimum_ms = 20,
-+ .maximum_ms = 100,
-+ .default_ms = 20,
-+ .minimum_bytes = 240,
-+ .samples_count = silk_samples
-+};
-+
-+static struct ast_codec silk16 = {
-+ .name = "silk",
-+ .description = "SILK Codec (16 KHz)",
-+ .type = AST_MEDIA_TYPE_AUDIO,
-+ .sample_rate = 16000,
-+ .minimum_ms = 20,
-+ .maximum_ms = 100,
-+ .default_ms = 20,
-+ .minimum_bytes = 320,
-+ .samples_count = silk_samples
-+};
-+
-+static struct ast_codec silk24 = {
-+ .name = "silk",
-+ .description = "SILK Codec (24 KHz)",
-+ .type = AST_MEDIA_TYPE_AUDIO,
-+ .sample_rate = 24000,
-+ .minimum_ms = 20,
-+ .maximum_ms = 100,
-+ .default_ms = 20,
-+ .minimum_bytes = 480,
-+ .samples_count = silk_samples
-+};
-+
- #define CODEC_REGISTER_AND_CACHE(codec) \
- ({ \
- int __res_ ## __LINE__ = 0; \
-@@ -843,6 +902,10 @@ int ast_codec_builtin_init(void)
- res |= CODEC_REGISTER_AND_CACHE(t140red);
- res |= CODEC_REGISTER_AND_CACHE(t140);
- res |= CODEC_REGISTER_AND_CACHE(none);
-+ res |= CODEC_REGISTER_AND_CACHE_NAMED("silk8", silk8);
-+ res |= CODEC_REGISTER_AND_CACHE_NAMED("silk12", silk12);
-+ res |= CODEC_REGISTER_AND_CACHE_NAMED("silk16", silk16);
-+ res |= CODEC_REGISTER_AND_CACHE_NAMED("silk24", silk24);
-
- return res;
- }
-diff --git a/main/format_cache.c b/main/format_cache.c
-index 6638a78..74ebfe8 100644
---- a/main/format_cache.c
-+++ b/main/format_cache.c
-@@ -232,6 +232,14 @@ struct ast_format *ast_format_t140_red;
- */
- struct ast_format *ast_format_none;
-
-+/*!
-+ * \brief Built-in "silk" format
-+ */
-+struct ast_format *ast_format_silk8;
-+struct ast_format *ast_format_silk12;
-+struct ast_format *ast_format_silk16;
-+struct ast_format *ast_format_silk24;
-+
- /*! \brief Number of buckets to use for the media format cache (should be prime for performance reasons) */
- #define CACHE_BUCKETS 53
-
-@@ -331,6 +339,10 @@ static void format_cache_shutdown(void)
- ao2_replace(ast_format_t140_red, NULL);
- ao2_replace(ast_format_t140, NULL);
- ao2_replace(ast_format_none, NULL);
-+ ao2_replace(ast_format_silk8, NULL);
-+ ao2_replace(ast_format_silk12, NULL);
-+ ao2_replace(ast_format_silk16, NULL);
-+ ao2_replace(ast_format_silk24, NULL);
- }
-
- int ast_format_cache_init(void)
-@@ -426,6 +438,14 @@ static void set_cached_format(const char *name, struct ast_format *format)
- ao2_replace(ast_format_t140, format);
- } else if (!strcmp(name, "none")) {
- ao2_replace(ast_format_none, format);
-+ } else if (!strcmp(name, "silk8")) {
-+ ao2_replace(ast_format_silk8, format);
-+ } else if (!strcmp(name, "silk12")) {
-+ ao2_replace(ast_format_silk12, format);
-+ } else if (!strcmp(name, "silk16")) {
-+ ao2_replace(ast_format_silk16, format);
-+ } else if (!strcmp(name, "silk24")) {
-+ ao2_replace(ast_format_silk24, format);
- }
- }
-
-diff --git a/main/rtp_engine.c b/main/rtp_engine.c
-index 462d4c5..8d46bfd 100644
---- a/main/rtp_engine.c
-+++ b/main/rtp_engine.c
-@@ -2198,6 +2198,11 @@ int ast_rtp_engine_init(void)
- /* Opus and VP8 */
- set_next_mime_type(ast_format_opus, 0, "audio", "opus", 48000);
- set_next_mime_type(ast_format_vp8, 0, "video", "VP8", 90000);
-+ /* DA SILK */
-+ set_next_mime_type(ast_format_silk8, 0, "audio", "silk", 8000);
-+ set_next_mime_type(ast_format_silk12, 0, "audio", "silk", 12000);
-+ set_next_mime_type(ast_format_silk16, 0, "audio", "silk", 16000);
-+ set_next_mime_type(ast_format_silk24, 0, "audio", "silk", 24000);
-
- /* Define the static rtp payload mappings */
- add_static_payload(0, ast_format_ulaw, 0);
-@@ -2243,6 +2248,11 @@ int ast_rtp_engine_init(void)
- add_static_payload(100, ast_format_vp8, 0);
- add_static_payload(107, ast_format_opus, 0);
-
-+ add_static_payload(108, ast_format_silk8, 0);
-+ add_static_payload(109, ast_format_silk12, 0);
-+ add_static_payload(113, ast_format_silk16, 0);
-+ add_static_payload(114, ast_format_silk24, 0);
-+
- return 0;
- }
-
-diff --git a/res/res_format_attr_silk.c b/res/res_format_attr_silk.c
-index dcbbe4c..d52ec74 100644
---- a/res/res_format_attr_silk.c
-+++ b/res/res_format_attr_silk.c
-@@ -40,7 +40,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
- * \note The only attribute that affects compatibility here is the sample rate.
- */
- struct silk_attr {
-- unsigned int samplerate;
- unsigned int maxbitrate;
- unsigned int dtx;
- unsigned int fec;
-@@ -54,10 +53,15 @@ static void silk_destroy(struct ast_format *format)
- ast_free(attr);
- }
-
-+static void attr_init(struct silk_attr *attr)
-+{
-+ memset(attr, 0, sizeof(*attr));
-+}
-+
- static int silk_clone(const struct ast_format *src, struct ast_format *dst)
- {
- struct silk_attr *original = ast_format_get_attribute_data(src);
-- struct silk_attr *attr = ast_calloc(1, sizeof(*attr));
-+ struct silk_attr *attr = ast_malloc(sizeof(*attr));
-
- if (!attr) {
- return -1;
-@@ -65,6 +69,8 @@ static int silk_clone(const struct ast_format *src, struct ast_format *dst)
-
- if (original) {
- *attr = *original;
-+ } else {
-+ attr_init(attr);
- }
-
- ast_format_set_attribute_data(dst, attr);
-@@ -109,17 +115,17 @@ static void silk_generate_sdp_fmtp(const struct ast_format *format, unsigned int
- ast_str_append(str, 0, "a=fmtp:%u maxaveragebitrate=%u\r\n", payload, attr->maxbitrate);
- }
-
-- ast_str_append(str, 0, "a=fmtp:%u usedtx=%u\r\n", payload, attr->dtx);
-- ast_str_append(str, 0, "a=fmtp:%u useinbandfec=%u\r\n", payload, attr->fec);
-+ if (attr->dtx) {
-+ ast_str_append(str, 0, "a=fmtp:%u usedtx=%u\r\n", payload, attr->dtx);
-+ }
-+ if (attr->fec) {
-+ ast_str_append(str, 0, "a=fmtp:%u useinbandfec=%u\r\n", payload, attr->fec);
-+ }
- }
-
- static enum ast_format_cmp_res silk_cmp(const struct ast_format *format1, const struct ast_format *format2)
- {
-- struct silk_attr *attr1 = ast_format_get_attribute_data(format1);
-- struct silk_attr *attr2 = ast_format_get_attribute_data(format2);
--
-- if (((!attr1 || !attr1->samplerate) && (!attr2 || !attr2->samplerate)) ||
-- (attr1->samplerate == attr2->samplerate)) {
-+ if (ast_format_get_sample_rate(format1) == ast_format_get_sample_rate(format2)) {
- return AST_FORMAT_CMP_EQUAL;
- }
-
-@@ -130,13 +136,10 @@ static struct ast_format *silk_getjoint(const struct ast_format *format1, const
- {
- struct silk_attr *attr1 = ast_format_get_attribute_data(format1);
- struct silk_attr *attr2 = ast_format_get_attribute_data(format2);
-- unsigned int samplerate;
- struct ast_format *jointformat;
- struct silk_attr *attr_res;
-
-- samplerate = attr1->samplerate & attr2->samplerate;
-- /* sample rate is the only attribute that has any bearing on if joint capabilities exist or not */
-- if (samplerate) {
-+ if (ast_format_get_sample_rate(format1) != ast_format_get_sample_rate(format2)) {
- return NULL;
- }
-
-@@ -145,22 +148,25 @@ static struct ast_format *silk_getjoint(const struct ast_format *format1, const
- return NULL;
- }
- attr_res = ast_format_get_attribute_data(jointformat);
-- attr_res->samplerate = samplerate;
-
-- /* Take the lowest max bitrate */
-- attr_res->maxbitrate = MIN(attr1->maxbitrate, attr2->maxbitrate);
-+ if (!attr1 || !attr2) {
-+ attr_init(attr_res);
-+ } else {
-+ /* Take the lowest max bitrate */
-+ attr_res->maxbitrate = MIN(attr1->maxbitrate, attr2->maxbitrate);
-
-- /* Only do dtx if both sides want it. DTX is a trade off between
-- * computational complexity and bandwidth. */
-- attr_res->dtx = attr1->dtx && attr2->dtx ? 1 : 0;
-+ /* Only do dtx if both sides want it. DTX is a trade off between
-+ * computational complexity and bandwidth. */
-+ attr_res->dtx = attr1->dtx && attr2->dtx ? 1 : 0;
-
-- /* Only do FEC if both sides want it. If a peer specifically requests not
-- * to receive with FEC, it may be a waste of bandwidth. */
-- attr_res->fec = attr1->fec && attr2->fec ? 1 : 0;
-+ /* Only do FEC if both sides want it. If a peer specifically requests not
-+ * to receive with FEC, it may be a waste of bandwidth. */
-+ attr_res->fec = attr1->fec && attr2->fec ? 1 : 0;
-
-- /* Use the maximum packetloss percentage between the two attributes. This affects how
-- * much redundancy is used in the FEC. */
-- attr_res->packetloss_percentage = MAX(attr1->packetloss_percentage, attr2->packetloss_percentage);
-+ /* Use the maximum packetloss percentage between the two attributes. This affects how
-+ * much redundancy is used in the FEC. */
-+ attr_res->packetloss_percentage = MAX(attr1->packetloss_percentage, attr2->packetloss_percentage);
-+ }
-
- return jointformat;
- }
-@@ -183,9 +189,7 @@ static struct ast_format *silk_set(const struct ast_format *format, const char *
- }
- attr = ast_format_get_attribute_data(cloned);
-
-- if (!strcasecmp(name, "sample_rate")) {
-- attr->samplerate = val;
-- } else if (!strcasecmp(name, "max_bitrate")) {
-+ if (!strcasecmp(name, "max_bitrate")) {
- attr->maxbitrate = val;
- } else if (!strcasecmp(name, "dtx")) {
- attr->dtx = val;
-@@ -205,9 +209,7 @@ static const void *silk_get(const struct ast_format *format, const char *name)
- struct silk_attr *attr = ast_format_get_attribute_data(format);
- unsigned int *val;
-
-- if (!strcasecmp(name, "sample_rate")) {
-- val = &attr->samplerate;
-- } else if (!strcasecmp(name, "max_bitrate")) {
-+ if (!strcasecmp(name, "max_bitrate")) {
- val = &attr->maxbitrate;
- } else if (!strcasecmp(name, "dtx")) {
- val = &attr->dtx;
---
-1.7.9.5
Modified: branches/1.0/package/asterisk/asterisk-13-configure-menuselect-cross-fix.patch
===================================================================
--- branches/1.0/package/asterisk/asterisk-13-configure-menuselect-cross-fix.patch 2016-09-02 16:50:57 UTC (rev 7815)
+++ branches/1.0/package/asterisk/asterisk-13-configure-menuselect-cross-fix.patch 2016-09-03 01:08:07 UTC (rev 7816)
@@ -1,6 +1,6 @@
--- asterisk-13.1.0/configure.ac.orig 2014-01-24 16:52:23.000000000 -0600
+++ asterisk-13.1.0/configure.ac 2014-01-24 16:55:14.000000000 -0600
-@@ -2636,11 +2636,13 @@
+@@ -2625,11 +2625,13 @@
AC_OUTPUT
${ac_cv_path_EGREP} 'CURSES|GTK2|OSARCH|NEWT' makeopts > makeopts.acbak2
Modified: branches/1.0/package/asterisk/asterisk-13-extension-changed-verbosity-chan_sip.patch
===================================================================
--- branches/1.0/package/asterisk/asterisk-13-extension-changed-verbosity-chan_sip.patch 2016-09-02 16:50:57 UTC (rev 7815)
+++ branches/1.0/package/asterisk/asterisk-13-extension-changed-verbosity-chan_sip.patch 2016-09-03 01:08:07 UTC (rev 7816)
@@ -1,6 +1,6 @@
--- asterisk-11.17.1/channels/chan_sip.c.orig 2015-04-23 10:22:04.000000000 -0500
+++ asterisk-11.17.1/channels/chan_sip.c 2015-04-23 10:22:40.000000000 -0500
-@@ -17351,7 +17351,7 @@
+@@ -17319,7 +17319,7 @@
}
if (!force) {
Modified: branches/1.0/package/asterisk/asterisk.mk
===================================================================
--- branches/1.0/package/asterisk/asterisk.mk 2016-09-02 16:50:57 UTC (rev 7815)
+++ branches/1.0/package/asterisk/asterisk.mk 2016-09-03 01:08:07 UTC (rev 7816)
@@ -9,7 +9,7 @@
ifeq ($(BR2_PACKAGE_ASTERISK_v11),y)
ASTERISK_VERSION := 11.23.0
else
-ASTERISK_VERSION := 13.10.0
+ASTERISK_VERSION := 13.11.0
endif
endif
ASTERISK_SOURCE := asterisk-$(ASTERISK_VERSION).tar.gz
@@ -271,6 +271,7 @@
(cd $(ASTERISK_DIR); \
menuselect/menuselect --enable res_pktccops menuselect.makeopts; \
menuselect/menuselect --disable CORE-SOUNDS-EN-GSM --disable MOH-OPSOUND-WAV menuselect.makeopts; \
+ menuselect/menuselect --disable BUILD_NATIVE menuselect.makeopts; \
)
endif
touch $@
Modified: branches/1.0/package/pjsip/asterisk-config_site.h
===================================================================
--- branches/1.0/package/pjsip/asterisk-config_site.h 2016-09-02 16:50:57 UTC (rev 7815)
+++ branches/1.0/package/pjsip/asterisk-config_site.h 2016-09-03 01:08:07 UTC (rev 7816)
@@ -19,7 +19,7 @@
#define PJ_SCANNER_USE_BITWISE 0
#define PJ_OS_HAS_CHECK_STACK 0
#define PJ_LOG_MAX_LEVEL 3
-#define PJ_ENABLE_EXTRA_CHECK 0
+#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)
@@ -37,3 +37,7 @@
#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 * 2)
Deleted: branches/1.0/package/pjsip/pjsip-0001-evsub-Add-APIs-to-add-decrement-an-event-subscriptio.patch
===================================================================
--- branches/1.0/package/pjsip/pjsip-0001-evsub-Add-APIs-to-add-decrement-an-event-subscriptio.patch 2016-09-02 16:50:57 UTC (rev 7815)
+++ branches/1.0/package/pjsip/pjsip-0001-evsub-Add-APIs-to-add-decrement-an-event-subscriptio.patch 2016-09-03 01:08:07 UTC (rev 7816)
@@ -1,72 +0,0 @@
-From a5030c9b33b2c936879fbacb1d2ea5edc2979181 Mon Sep 17 00:00:00 2001
-From: George Joseph <gj...@di...>
-Date: Sat, 18 Jun 2016 10:14:34 -0600
-Subject: [PATCH] evsub: Add APIs to add/decrement an event subscription's
- group lock
-
-These APIs can be used to ensure that the evsub isn't destroyed before
-an application is finished using it.
----
- pjsip/include/pjsip-simple/evsub.h | 20 ++++++++++++++++++++
- pjsip/src/pjsip-simple/evsub.c | 14 ++++++++++++++
- 2 files changed, 34 insertions(+)
-
-diff --git a/pjsip/include/pjsip-simple/evsub.h b/pjsip/include/pjsip-simple/evsub.h
-index 2dc4d69..31f85f8 100644
---- a/pjsip/include/pjsip-simple/evsub.h
-+++ b/pjsip/include/pjsip-simple/evsub.h
-@@ -490,6 +490,26 @@ PJ_DECL(void) pjsip_evsub_set_mod_data( pjsip_evsub *sub, unsigned mod_id,
- PJ_DECL(void*) pjsip_evsub_get_mod_data( pjsip_evsub *sub, unsigned mod_id );
-
-
-+/**
-+ * Increment the event subscription's group lock.
-+ *
-+ * @param sub The server subscription instance.
-+ *
-+ * @return PJ_SUCCESS on success.
-+ */
-+PJ_DEF(pj_status_t) pjsip_evsub_add_ref(pjsip_evsub *sub);
-+
-+
-+/**
-+ * Decrement the event subscription's group lock.
-+ *
-+ * @param sub The server subscription instance.
-+ *
-+ * @return PJ_SUCCESS on success.
-+ */
-+PJ_DEF(pj_status_t) pjsip_evsub_dec_ref(pjsip_evsub *sub);
-+
-+
-
- PJ_END_DECL
-
-diff --git a/pjsip/src/pjsip-simple/evsub.c b/pjsip/src/pjsip-simple/evsub.c
-index 7cd8859..68a9564 100644
---- a/pjsip/src/pjsip-simple/evsub.c
-+++ b/pjsip/src/pjsip-simple/evsub.c
-@@ -831,7 +831,21 @@ static pj_status_t evsub_create( pjsip_dialog *dlg,
- return PJ_SUCCESS;
- }
-
-+/*
-+ * Increment the event subscription's group lock.
-+ */
-+PJ_DEF(pj_status_t) pjsip_evsub_add_ref(pjsip_evsub *sub)
-+{
-+ return pj_grp_lock_add_ref(sub->grp_lock);
-+}
-
-+/*
-+ * Decrement the event subscription's group lock.
-+ */
-+PJ_DEF(pj_status_t) pjsip_evsub_dec_ref(pjsip_evsub *sub)
-+{
-+ return pj_grp_lock_dec_ref(sub->grp_lock);
-+}
-
- /*
- * Create client subscription session.
---
-2.5.5
Added: branches/1.0/package/pjsip/pjsip-0002-Fix-1946-Avoid-deinitialization-of-uninitialized-cli.patch
===================================================================
--- branches/1.0/package/pjsip/pjsip-0002-Fix-1946-Avoid-deinitialization-of-uninitialized-cli.patch (rev 0)
+++ branches/1.0/package/pjsip/pjsip-0002-Fix-1946-Avoid-deinitialization-of-uninitialized-cli.patch 2016-09-03 01:08:07 UTC (rev 7816)
@@ -0,0 +1,55 @@
+From 33fd755e819dc85a96718abc0ae26a9b46f14800 Mon Sep 17 00:00:00 2001
+From: nanang <nanang@localhost>
+Date: Thu, 28 Jul 2016 08:21:45 +0000
+Subject: [PATCH 2/3] Fix #1946: Avoid deinitialization of uninitialized client
+ auth session.
+
+---
+ pjsip/src/pjsip/sip_dialog.c | 18 ++++++------------
+ 1 file changed, 6 insertions(+), 12 deletions(-)
+
+diff --git a/pjsip/src/pjsip/sip_dialog.c b/pjsip/src/pjsip/sip_dialog.c
+index f03885d..421ddc4 100644
+--- a/pjsip/src/pjsip/sip_dialog.c
++++ b/pjsip/src/pjsip/sip_dialog.c
+@@ -92,6 +92,12 @@ static pj_status_t create_dialog( pjsip_user_agent *ua,
+ pj_list_init(&dlg->inv_hdr);
+ pj_list_init(&dlg->rem_cap_hdr);
+
++ /* Init client authentication session. */
++ status = pjsip_auth_clt_init(&dlg->auth_sess, dlg->endpt,
++ dlg->pool, 0);
++ if (status != PJ_SUCCESS)
++ goto on_error;
++
+ status = pj_mutex_create_recursive(pool, dlg->obj_name, &dlg->mutex_);
+ if (status != PJ_SUCCESS)
+ goto on_error;
+@@ -283,12 +289,6 @@ PJ_DEF(pj_status_t) pjsip_dlg_create_uac( pjsip_user_agent *ua,
+ /* Initial route set is empty. */
+ pj_list_init(&dlg->route_set);
+
+- /* Init client authentication session. */
+- status = pjsip_auth_clt_init(&dlg->auth_sess, dlg->endpt,
+- dlg->pool, 0);
+- if (status != PJ_SUCCESS)
+- goto on_error;
+-
+ /* Register this dialog to user agent. */
+ status = pjsip_ua_register_dlg( ua, dlg );
+ if (status != PJ_SUCCESS)
+@@ -506,12 +506,6 @@ pj_status_t create_uas_dialog( pjsip_user_agent *ua,
+ }
+ dlg->route_set_frozen = PJ_TRUE;
+
+- /* Init client authentication session. */
+- status = pjsip_auth_clt_init(&dlg->auth_sess, dlg->endpt,
+- dlg->pool, 0);
+- if (status != PJ_SUCCESS)
+- goto on_error;
+-
+ /* Increment the dialog's lock since tsx may cause the dialog to be
+ * destroyed prematurely (such as in case of transport error).
+ */
+--
+2.7.4
Deleted: branches/1.0/package/pjsip/pjsip-0002-sip_transport_tcp-tls-Set-factory-on-transports-crea.patch
===================================================================
--- branches/1.0/package/pjsip/pjsip-0002-sip_transport_tcp-tls-Set-factory-on-transports-crea.patch 2016-09-02 16:50:57 UTC (rev 7815)
+++ branches/1.0/package/pjsip/pjsip-0002-sip_transport_tcp-tls-Set-factory-on-transports-crea.patch 2016-09-03 01:08:07 UTC (rev 7816)
@@ -1,47 +0,0 @@
-From b7cb93b0e1729589a71e8b30d9a9893f0918e2a2 Mon Sep 17 00:00:00 2001
-From: George Joseph <geo...@fa...>
-Date: Mon, 30 May 2016 11:58:22 -0600
-Subject: [PATCH] sip_transport_tcp/tls: Set factory on transports created
- from accept
-
-The ability to re-use tcp and tls transports when a factory is
-specified now depends on transport->factory being set which is a new field
-in 2.5. This was being set only on new outgoing sockets not on
-incoming sockets. The result was that a client REGISTER created a new
-socket but without the factory set, the next outgoing request to the
-client, OPTIONS, INVITE, etc, would attempt to create another socket
-which the client would refuse.
-
-This patch sets the factory on transports created as a result of an
-accept.
----
- pjsip/src/pjsip/sip_transport_tcp.c | 1 +
- pjsip/src/pjsip/sip_transport_tls.c | 1 +
- 2 files changed, 2 insertions(+)
-
-diff --git a/pjsip/src/pjsip/sip_transport_tcp.c b/pjsip/src/pjsip/sip_transport_tcp.c
-index 1bbb324..00eb8fc 100644
---- a/pjsip/src/pjsip/sip_transport_tcp.c
-+++ b/pjsip/src/pjsip/sip_transport_tcp.c
-@@ -713,6 +713,7 @@ static pj_status_t tcp_create( struct tcp_listener *listener,
- tcp->base.send_msg = &tcp_send_msg;
- tcp->base.do_shutdown = &tcp_shutdown;
- tcp->base.destroy = &tcp_destroy_transport;
-+ tcp->base.factory = &listener->factory;
-
- /* Create group lock */
- status = pj_grp_lock_create(pool, NULL, &tcp->grp_lock);
-diff --git a/pjsip/src/pjsip/sip_transport_tls.c b/pjsip/src/pjsip/sip_transport_tls.c
-index a83ac32..36ee70d 100644
---- a/pjsip/src/pjsip/sip_transport_tls.c
-+++ b/pjsip/src/pjsip/sip_transport_tls.c
-@@ -742,6 +742,7 @@ static pj_status_t tls_create( struct tls_listener *listener,
- tls->base.send_msg = &tls_send_msg;
- tls->base.do_shutdown = &tls_shutdown;
- tls->base.destroy = &tls_destroy_transport;
-+ tls->base.factory = &listener->factory;
-
- tls->ssock = ssock;
-
---
-2.5.5
Modified: branches/1.0/package/pjsip/pjsip.mk
===================================================================
--- branches/1.0/package/pjsip/pjsip.mk 2016-09-02 16:50:57 UTC (rev 7815)
+++ branches/1.0/package/pjsip/pjsip.mk 2016-09-03 01:08:07 UTC (rev 7816)
@@ -4,7 +4,7 @@
#
################################################################################
-PJSIP_VERSION = 2.5
+PJSIP_VERSION = 2.5.5
PJSIP_SOURCE = pjproject-$(PJSIP_VERSION).tar.bz2
PJSIP_SITE = http://www.pjsip.org/release/$(PJSIP_VERSION)
PJSIP_INSTALL_STAGING = YES
@@ -17,23 +17,36 @@
PJSIP_POST_PATCH_HOOKS += PJSIP_CUSTOM_CONFIG
PJSIP_CONF_ENV = \
- CFLAGS="$(TARGET_CFLAGS) -O2" \
+ CFLAGS="$(TARGET_CFLAGS) -O2 -Wno-unused-but-set-variable -Wno-unused-variable -Wno-unused-label -Wno-unused-function -Wno-strict-aliasing" \
LDFLAGS="-L$(STAGING_DIR)/usr/lib" \
LD="$(TARGET_CC)"
PJSIP_CONF_OPT = \
- --with-external-srtp \
--disable-speex-codec \
--disable-speex-aec \
--disable-gsm-codec \
+ --disable-ilbc-codec \
+ --disable-l16-codec \
+ --disable-g711-codec \
+ --disable-g722-codec \
+ --disable-g7221-codec \
+ --disable-opencore-amr \
+ --disable-webrtc \
+ --disable-silk \
+ --disable-opus \
--disable-video \
--disable-v4l2 \
--disable-sound \
- --disable-opencore-amr \
- --disable-ilbc-codec \
- --disable-g7221-codec \
+ --disable-ext-sound \
+ --disable-oss \
+ --disable-sdl \
+ --disable-libyuv \
--disable-resample \
- --without-libyuv
+ --disable-ffmpeg \
+ --disable-openh264 \
+ --disable-ipp \
+ --without-external-pa \
+ --with-external-srtp
ifeq ($(BR2_PACKAGE_OPENSSL),y)
PJSIP_CONF_OPT += --with-ssl=$(STAGING_DIR)/usr
This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.
|
