From 78f56215c37b59751451705155b2deefe521f23c Mon Sep 17 00:00:00 2001 From: Igor Goncharovskiy Date: Mon, 10 Dec 2012 06:41:47 +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) ........ 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 --- 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 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 { -- 2.47.2