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Merge pull request #436 from micmac1/18-ast13-AST-2019-003

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(for 18.06) asterisk-1{3,5}.x: fix AST-2019-003
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Jiri Slachta 2019-07-12 22:14:26 +02:00 committed by GitHub
commit 67181ed5da
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4 changed files with 80 additions and 2 deletions
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2
net/asterisk-13.x/Makefile
View file

@ -11,7 +11,7 @@ include $(TOPDIR)/rules.mk
PKG_NAME:=asterisk13
PKG_VERSION:=13.20.0
PKG_RELEASE:=3
PKG_RELEASE:=4
PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases

39
net/asterisk-13.x/patches/130-AST-2019-003-13.diff Normal file
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@ -0,0 +1,39 @@
From 3ab9291a563656dfebcb7de67c86351541f3de1c Mon Sep 17 00:00:00 2001
From: Francesco Castellano <francesco.castellano@messagenet.it>
Date: Fri, 28 Jun 2019 18:15:31 +0200
Subject: [PATCH] chan_sip: Handle invalid SDP answer to T.38 re-invite
The chan_sip module performs a T.38 re-invite using a single media
stream of udptl, and expects the SDP answer to be the same.
If an SDP answer is received instead that contains an additional
media stream with no joint codec a crash will occur as the code
assumes that at least one joint codec will exist in this
scenario.
This change removes this assumption.
ASTERISK-28465
Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87
---
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 7c8928d..223ff3c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10911,7 +10911,13 @@
ast_rtp_lookup_mime_multiple2(s3, NULL, newnoncodeccapability, 0, 0));
}
- if (portno != -1 || vportno != -1 || tportno != -1) {
+ /* When UDPTL is negotiated it is expected that there are no compatible codecs as audio or
+ * video is not being transported, thus we continue in this function further up if that is
+ * the case. If we receive an SDP answer containing both a UDPTL stream and another media
+ * stream however we need to check again to ensure that there is at least one joint codec
+ * instead of assuming there is one.
+ */
+ if ((portno != -1 || vportno != -1 || tportno != -1) && ast_format_cap_count(newjointcapability)) {
/* We are now ready to change the sip session and RTP structures with the offered codecs, since
they are acceptable */
unsigned int framing;

2
net/asterisk-15.x/Makefile
View file

@ -9,7 +9,7 @@ include $(TOPDIR)/rules.mk
PKG_NAME:=asterisk15
PKG_VERSION:=15.3.0
PKG_RELEASE:=5
PKG_RELEASE:=6
PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases

39
net/asterisk-15.x/patches/160-AST-2019-003-15.diff Normal file
View file

@ -0,0 +1,39 @@
From a8cc63a8b2b973d6d34251d74b8d4576d6796dce Mon Sep 17 00:00:00 2001
From: Francesco Castellano <francesco.castellano@messagenet.it>
Date: Fri, 28 Jun 2019 18:15:31 +0200
Subject: [PATCH] chan_sip: Handle invalid SDP answer to T.38 re-invite
The chan_sip module performs a T.38 re-invite using a single media
stream of udptl, and expects the SDP answer to be the same.
If an SDP answer is received instead that contains an additional
media stream with no joint codec a crash will occur as the code
assumes that at least one joint codec will exist in this
scenario.
This change removes this assumption.
ASTERISK-28465
Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87
---
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index fe2ae1e..6251878 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10921,7 +10921,13 @@
ast_rtp_lookup_mime_multiple2(s3, NULL, newnoncodeccapability, 0, 0));
}
- if (portno != -1 || vportno != -1 || tportno != -1) {
+ /* When UDPTL is negotiated it is expected that there are no compatible codecs as audio or
+ * video is not being transported, thus we continue in this function further up if that is
+ * the case. If we receive an SDP answer containing both a UDPTL stream and another media
+ * stream however we need to check again to ensure that there is at least one joint codec
+ * instead of assuming there is one.
+ */
+ if ((portno != -1 || vportno != -1 || tportno != -1) && ast_format_cap_count(newjointcapability)) {
/* We are now ready to change the sip session and RTP structures with the offered codecs, since
they are acceptable */
unsigned int framing;