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| author | Jiri Slachta <jiri@slachta.eu> | 2019-07-12 22:14:51 +0200 |
|---|---|---|
| committer | GitHub <noreply@github.com> | 2019-07-12 22:14:51 +0200 |
| commit | 039faa9ef975fc44dd9ef3c022f7d4e7c7e4222f (patch) | |
| tree | 9a808b26bc91517956af30a09deb4b524cdbdab9 | |
| parent | 31398a3759c99bdb2dca359b78e25c17b6dd434a (diff) | |
| parent | 15e6a1b558aced85f4ffac1ea388a217312085d0 (diff) | |
Merge pull request #435 from micmac1/17-ast13-AST-2019-003
(for 17.01) asterisk-13.x: fix AST-2019-003
| -rw-r--r-- | net/asterisk-13.x/Makefile | 2 | ||||
| -rw-r--r-- | net/asterisk-13.x/patches/080-AST-2019-003-13.diff | 39 |
2 files changed, 40 insertions, 1 deletions
diff --git a/net/asterisk-13.x/Makefile b/net/asterisk-13.x/Makefile index b789fbb..07d42f8 100644 --- a/net/asterisk-13.x/Makefile +++ b/net/asterisk-13.x/Makefile @@ -10,7 +10,7 @@ include $(TOPDIR)/rules.mk PKG_NAME:=asterisk13 PKG_VERSION:=13.19.2 -PKG_RELEASE:=3 +PKG_RELEASE:=4 PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases/ diff --git a/net/asterisk-13.x/patches/080-AST-2019-003-13.diff b/net/asterisk-13.x/patches/080-AST-2019-003-13.diff new file mode 100644 index 0000000..afc8325 --- /dev/null +++ b/net/asterisk-13.x/patches/080-AST-2019-003-13.diff @@ -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; |