[Pkg-voip-commits] [asterisk] 01/02: Fix one-way audio with chan_sip when transcoding

[Bernhard Schmidt]

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berni pushed a commit to branch stretch
in repository asterisk.

commit bf82da252013b0621c7526a26b5faba892ff2faa
Author: Bernhard Schmidt <Bernhard.Schmidt at lrz.de>
Date:   Tue Sep 12 13:08:04 2017 +0200

    Fix one-way audio with chan_sip when transcoding
    
    Import upstream commit to properly select codecs to transcode
    
    Closes: #875450
---
 debian/patches/875450-chan_sip-oneway-audio.patch | 33 +++++++++++++++++++++++
 debian/patches/series                             |  3 +++
 2 files changed, 36 insertions(+)

diff --git a/debian/patches/875450-chan_sip-oneway-audio.patch b/debian/patches/875450-chan_sip-oneway-audio.patch
new file mode 100644
index 0000000..a5bfe39
--- /dev/null
+++ b/debian/patches/875450-chan_sip-oneway-audio.patch
@@ -0,0 +1,33 @@
+From: Vitezslav Novy <a1 at vnovy.net>
+Date: Mon, 8 May 2017 20:40:47 +0200
+Subject: [PATCH] chan_sip: Change sip_get_codec() to return correct codec list
+Origin: upstream, https://github.com/asterisk/asterisk/commit/93b7f84c1ac61208607ec6f7360b594dee921a1b
+Bug: https://issues.asterisk.org/jira/browse/ASTERISK-26143
+Bug-Debian: https://bugs.debian.org/875450
+
+Return cahnnel nativeformats to fix bridge technology selection process.
+Same approach as in pjsip module.
+
+ASTERISK-26143
+Reported-by: Henning Holtschneider
+
+Change-Id: I64e863753954d6ad67a9e722df2ebc328705ad48
+---
+ channels/chan_sip.c | 4 +---
+ 1 file changed, 1 insertion(+), 3 deletions(-)
+
+diff --git a/channels/chan_sip.c b/channels/chan_sip.c
+index e7c15bcbd0c..930dc0f6756 100644
+--- a/channels/chan_sip.c
++++ b/channels/chan_sip.c
+@@ -33588,9 +33588,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
+ 
+ static void sip_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
+ {
+-	struct sip_pvt *p = ast_channel_tech_pvt(chan);
+-
+-	ast_format_cap_append_from_cap(result, !ast_format_cap_count(p->peercaps) ? p->caps : p->peercaps, AST_MEDIA_TYPE_UNKNOWN);
++	ast_format_cap_append_from_cap(result, ast_channel_nativeformats(chan), AST_MEDIA_TYPE_UNKNOWN);
+ }
+ 
+ static struct ast_rtp_glue sip_rtp_glue = {
diff --git a/debian/patches/series b/debian/patches/series
index ea6b3bc..a9c0e84 100644
--- a/debian/patches/series
+++ b/debian/patches/series
@@ -43,6 +43,9 @@ pjsip_unresolved_symbol.patch
 859911-pjsip-set-rtp-source-address.patch
 859911-pjsip-set-rtp-source-address-part2.patch
 
+# Bug#875450 - chan_sip: One way audio when transcoding
+875450-chan_sip-oneway-audio.patch
+
 AST-2017-004.patch
 AST-2017-005-13.13.diff
 AST-2017-006-13.diff

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