Merge pull request #438 from micmac1/ast13-AST-2019-002-and-003
asterisk-1{3,6}.x: fix AST-2019-002 & AST-2019-003
This commit is contained in:
Jiri Slachta
GitHub
commit
f900204c89
6 changed files with 155 additions and 2 deletions
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@ -11,7 +11,7 @@ include $(TOPDIR)/rules.mk
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PKG_NAME:=asterisk13
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PKG_VERSION:=13.24.0
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PKG_RELEASE:=2
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PKG_RELEASE:=3
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PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
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PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases
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35
net/asterisk-13.x/patches/110-AST-2019-002-13.diff
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35
net/asterisk-13.x/patches/110-AST-2019-002-13.diff
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@ -0,0 +1,35 @@
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From a9d8b56831146166abc7fb8abe8ae8aaff295358 Mon Sep 17 00:00:00 2001
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From: George Joseph <gjoseph@digium.com>
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Date: Wed, 12 Jun 2019 12:03:04 -0600
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Subject: [PATCH] res_pjsip_messaging: Check for body in in-dialog message
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We now check that a body exists and it has a length > 0 before
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attempting to process it.
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ASTERISK-28447
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Reported-by: Gil Richard
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Change-Id: Ic469544b22ab848734636588d4c93426cc6f4b1f
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---
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diff --git a/res/res_pjsip_messaging.c b/res/res_pjsip_messaging.c
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index 10c5f29..76d37f2 100644
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--- a/res/res_pjsip_messaging.c
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+++ b/res/res_pjsip_messaging.c
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@@ -91,10 +91,13 @@
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static const pj_str_t text = { "text", 4};
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static const pj_str_t application = { "application", 11};
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+ if (!(rdata->msg_info.msg->body && rdata->msg_info.msg->body->len > 0)) {
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+ return res;
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+ }
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+
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/* We'll accept any text/ or application/ content type */
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- if (rdata->msg_info.msg->body && rdata->msg_info.msg->body->len
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- && (pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &text) == 0
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- || pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &application) == 0)) {
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+ if (pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &text) == 0
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+ || pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &application) == 0) {
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res = PJSIP_SC_OK;
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} else if (rdata->msg_info.ctype
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&& (pj_stricmp(&rdata->msg_info.ctype->media.type, &text) == 0
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39
net/asterisk-13.x/patches/120-AST-2019-003-13.diff
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39
net/asterisk-13.x/patches/120-AST-2019-003-13.diff
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@ -0,0 +1,39 @@
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From 3ab9291a563656dfebcb7de67c86351541f3de1c Mon Sep 17 00:00:00 2001
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From: Francesco Castellano <francesco.castellano@messagenet.it>
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Date: Fri, 28 Jun 2019 18:15:31 +0200
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Subject: [PATCH] chan_sip: Handle invalid SDP answer to T.38 re-invite
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The chan_sip module performs a T.38 re-invite using a single media
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stream of udptl, and expects the SDP answer to be the same.
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If an SDP answer is received instead that contains an additional
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media stream with no joint codec a crash will occur as the code
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assumes that at least one joint codec will exist in this
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scenario.
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This change removes this assumption.
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ASTERISK-28465
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Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87
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---
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diff --git a/channels/chan_sip.c b/channels/chan_sip.c
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index 7c8928d..223ff3c 100644
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--- a/channels/chan_sip.c
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+++ b/channels/chan_sip.c
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@@ -10911,7 +10911,13 @@
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ast_rtp_lookup_mime_multiple2(s3, NULL, newnoncodeccapability, 0, 0));
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}
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- if (portno != -1 || vportno != -1 || tportno != -1) {
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+ /* When UDPTL is negotiated it is expected that there are no compatible codecs as audio or
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+ * video is not being transported, thus we continue in this function further up if that is
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+ * the case. If we receive an SDP answer containing both a UDPTL stream and another media
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+ * stream however we need to check again to ensure that there is at least one joint codec
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+ * instead of assuming there is one.
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+ */
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+ if ((portno != -1 || vportno != -1 || tportno != -1) && ast_format_cap_count(newjointcapability)) {
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/* We are now ready to change the sip session and RTP structures with the offered codecs, since
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they are acceptable */
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unsigned int framing;
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@ -10,7 +10,7 @@ include $(TOPDIR)/rules.mk
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AST_MAJOR_VERSION:=16
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PKG_NAME:=asterisk$(AST_MAJOR_VERSION)
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PKG_VERSION:=$(AST_MAJOR_VERSION).3.0
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PKG_RELEASE:=3
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PKG_RELEASE:=4
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PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
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PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases
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40
net/asterisk-16.x/patches/140-AST-2019-002-16.diff
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40
net/asterisk-16.x/patches/140-AST-2019-002-16.diff
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@ -0,0 +1,40 @@
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From 785bf3a755e47d92caef110e6040295764d08127 Mon Sep 17 00:00:00 2001
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From: George Joseph <gjoseph@digium.com>
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Date: Wed, 12 Jun 2019 12:03:04 -0600
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Subject: [PATCH] res_pjsip_messaging: Check for body in in-dialog message
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We now check that a body exists and it has a length > 0 before
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attempting to process it.
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ASTERISK-28447
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Reported-by: Gil Richard
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Change-Id: Ic469544b22ab848734636588d4c93426cc6f4b1f
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---
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res/res_pjsip_messaging.c | 9 ++++++---
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1 file changed, 6 insertions(+), 3 deletions(-)
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diff --git a/res/res_pjsip_messaging.c b/res/res_pjsip_messaging.c
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index 0e10a8f047..930cf84a53 100644
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--- a/res/res_pjsip_messaging.c
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+++ b/res/res_pjsip_messaging.c
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@@ -90,10 +90,13 @@ static enum pjsip_status_code check_content_type_in_dialog(const pjsip_rx_data *
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static const pj_str_t text = { "text", 4};
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static const pj_str_t application = { "application", 11};
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+ if (!(rdata->msg_info.msg->body && rdata->msg_info.msg->body->len > 0)) {
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+ return res;
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+ }
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+
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/* We'll accept any text/ or application/ content type */
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- if (rdata->msg_info.msg->body && rdata->msg_info.msg->body->len
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- && (pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &text) == 0
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- || pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &application) == 0)) {
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+ if (pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &text) == 0
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+ || pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &application) == 0) {
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res = PJSIP_SC_OK;
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} else if (rdata->msg_info.ctype
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&& (pj_stricmp(&rdata->msg_info.ctype->media.type, &text) == 0
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--
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2.21.0
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39
net/asterisk-16.x/patches/150-AST-2019-003-16.diff
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39
net/asterisk-16.x/patches/150-AST-2019-003-16.diff
Normal file
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@ -0,0 +1,39 @@
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From 1e4df0215af4f192ed06a7fc7589c799f1ec6091 Mon Sep 17 00:00:00 2001
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From: Francesco Castellano <francesco.castellano@messagenet.it>
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Date: Fri, 28 Jun 2019 18:15:31 +0200
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Subject: [PATCH] chan_sip: Handle invalid SDP answer to T.38 re-invite
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The chan_sip module performs a T.38 re-invite using a single media
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stream of udptl, and expects the SDP answer to be the same.
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If an SDP answer is received instead that contains an additional
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media stream with no joint codec a crash will occur as the code
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assumes that at least one joint codec will exist in this
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scenario.
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This change removes this assumption.
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ASTERISK-28465
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Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87
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---
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diff --git a/channels/chan_sip.c b/channels/chan_sip.c
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index 898b646..a609ff8 100644
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--- a/channels/chan_sip.c
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+++ b/channels/chan_sip.c
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@@ -10965,7 +10965,13 @@
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ast_rtp_lookup_mime_multiple2(s3, NULL, newnoncodeccapability, 0, 0));
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}
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- if (portno != -1 || vportno != -1 || tportno != -1) {
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+ /* When UDPTL is negotiated it is expected that there are no compatible codecs as audio or
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+ * video is not being transported, thus we continue in this function further up if that is
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+ * the case. If we receive an SDP answer containing both a UDPTL stream and another media
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+ * stream however we need to check again to ensure that there is at least one joint codec
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+ * instead of assuming there is one.
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+ */
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+ if ((portno != -1 || vportno != -1 || tportno != -1) && ast_format_cap_count(newjointcapability)) {
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/* We are now ready to change the sip session and RTP structures with the offered codecs, since
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they are acceptable */
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unsigned int framing;
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