[asterisk-commits] igorg: trunk r377594 - in /trunk: ./ channels/chan_unistim.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Dec 10 00:56:07 CST 2012


Author: igorg
Date: Mon Dec 10 00:56:04 2012
New Revision: 377594

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=377594
Log:
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
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Merged revisions 377593 from http://svn.asterisk.org/svn/asterisk/branches/11

Modified:
    trunk/   (props changed)
    trunk/channels/chan_unistim.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.

Modified: trunk/channels/chan_unistim.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_unistim.c?view=diff&rev=377594&r1=377593&r2=377594
==============================================================================
--- trunk/channels/chan_unistim.c (original)
+++ trunk/channels/chan_unistim.c Mon Dec 10 00:56:04 2012
@@ -2666,9 +2666,9 @@
 			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 @@
 			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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