From: Igor Goncharovskiy <igor.goncharovsky@gmail.com>
Date: Mon, 10 Dec 2012 06:41:47 +0000 (+0000)
Subject: Fix codec mismatch
X-Git-Tag: 10.12.0-rc2~3^2~23
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=78f56215c37b59751451705155b2deefe521f23c;p=thirdparty%2Fasterisk.git

Fix codec mismatch

Fix code to send in both rx and tx open stream messages correct codecs. Found that on phase 0/1 phones wrong codecs cause to no audio in some situations.

(issue ASTERISK-20183)
........

Merged revisions 377591 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@377592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_unistim.c b/channels/chan_unistim.c
index 177eba5558..d506eed0a4 100644
--- a/channels/chan_unistim.c
+++ b/channels/chan_unistim.c
@@ -2183,9 +2183,9 @@ static void start_rtp(struct unistim_subchannel *sub)
 			buffsend[16] = (htons(sin.sin_port) & 0x00ff);
 			buffsend[20] = (us.sin_port & 0xff00) >> 8;
 			buffsend[19] = (us.sin_port & 0x00ff);
-			buffsend[11] = codec;
 		}
-		buffsend[12] = codec;
+		buffsend[11] = codec; /* rx */
+		buffsend[12] = codec; /* tx */
 		send_client(SIZE_HEADER + sizeof(packet_send_open_audio_stream_tx), buffsend,
 				   sub->parent->parent->session);
 
@@ -2213,9 +2213,9 @@ static void start_rtp(struct unistim_subchannel *sub)
 			buffsend[16] = (htons(sin.sin_port) & 0x00ff);
 			buffsend[20] = (us.sin_port & 0xff00) >> 8;
 			buffsend[19] = (us.sin_port & 0x00ff);
-			buffsend[12] = codec;
 		}
-		buffsend[11] = codec;
+		buffsend[11] = codec; /* rx */
+		buffsend[12] = codec; /* tx */
 		send_client(SIZE_HEADER + sizeof(packet_send_open_audio_stream_rx), buffsend,
 				   sub->parent->parent->session);
 	} else {