From: Igor Goncharovskiy Date: Mon, 10 Dec 2012 06:40:18 +0000 (+0000) Subject: Fix codec mismatch X-Git-Tag: 1.8.21.0-rc1~69 X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=32d4c02de2d77947ea95e1ecae12c83a2107d11a;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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377591 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- diff --git a/channels/chan_unistim.c b/channels/chan_unistim.c index fcbe8a124a..f8ed7d6701 100644 --- a/channels/chan_unistim.c +++ b/channels/chan_unistim.c @@ -2151,9 +2151,9 @@ 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); @@ -2181,9 +2181,9 @@ 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 {