From 98539ffb32a87c15e03aeacd46f33780cb0aa071 Mon Sep 17 00:00:00 2001
From: Igor Goncharovskiy <igor.goncharovsky@gmail.com>
Date: Mon, 10 Dec 2012 06:56:04 +0000
Subject: [PATCH] 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)
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Merged revisions 377591 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 377592 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 377593 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_unistim.c | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/channels/chan_unistim.c b/channels/chan_unistim.c
index 2179cb65ee..ac4daa68da 100644
--- a/channels/chan_unistim.c
+++ b/channels/chan_unistim.c
@@ -2666,9 +2666,9 @@ static void send_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, pte);
if (unistimdebug) {
@@ -2697,9 +2697,9 @@ static void send_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, pte);
} else {
uint16_t rtcpsin_port = htons(us.sin_port) + 1; /* RTCP port is RTP + 1 */
--
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