Merge pull request #435 from micmac1/17-ast13-AST-2019-003
(for 17.01) asterisk-13.x: fix AST-2019-003
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commit
039faa9ef9
2 changed files with 40 additions and 1 deletions
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@ -10,7 +10,7 @@ include $(TOPDIR)/rules.mk
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PKG_NAME:=asterisk13
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PKG_NAME:=asterisk13
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PKG_VERSION:=13.19.2
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PKG_VERSION:=13.19.2
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PKG_RELEASE:=3
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PKG_RELEASE:=4
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PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
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PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
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PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases/
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PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases/
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39
net/asterisk-13.x/patches/080-AST-2019-003-13.diff
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39
net/asterisk-13.x/patches/080-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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