From: Igor Goncharovskiy Date: Mon, 10 Dec 2012 06:49:45 +0000 (+0000) Subject: Fix codec mismatch X-Git-Tag: 11.3.0-rc1~3^2~94 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=eb76c8cd0bbb4c02fc9e8ded0c58ee4fcf96f1d2;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 ........ Merged revisions 377592 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377593 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- 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 */