[asterisk-commits] kmoore: branch 12 r415342 - /branches/12/res/res_pjsip_sdp_rtp.c

[SVN commits to the Asterisk project]

Author: kmoore
Date: Fri Jun  6 14:08:42 2014
New Revision: 415342

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=415342
Log:
PJSIP: Remove premature write of raw formats

Currently, there are situations that can occur when using chan_pjsip
and certain dialplan applications (notably ChanSpy()) that can cause
the channel to get no audio with scrolling warnings about format
mismatches. This is caused by a failure to update translation paths on
a mid-call native format update since the raw formats have already
been updated by res_pjsip_sdp_rtp.c in set_caps(). Removing the
premature raw format updates allows the translation paths to be setup
correctly and the raw read and write formats with them.

AFS-63 #close

Modified:
    branches/12/res/res_pjsip_sdp_rtp.c

Modified: branches/12/res/res_pjsip_sdp_rtp.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/res/res_pjsip_sdp_rtp.c?view=diff&rev=415342&r1=415341&r2=415342
==============================================================================
--- branches/12/res/res_pjsip_sdp_rtp.c (original)
+++ branches/12/res/res_pjsip_sdp_rtp.c Fri Jun  6 14:08:42 2014
@@ -251,8 +251,6 @@
 
 		/* Apply the new formats to the channel, potentially changing read/write formats while doing so */
 		ast_format_cap_copy(ast_channel_nativeformats(session->channel), caps);
-		ast_format_copy(ast_channel_rawwriteformat(session->channel), &fmt);
-		ast_format_copy(ast_channel_rawreadformat(session->channel), &fmt);
 		ast_set_read_format(session->channel, ast_channel_readformat(session->channel));
 		ast_set_write_format(session->channel, ast_channel_writeformat(session->channel));
 	}