[asterisk-commits] mmichelson: branch 11 r400908 - in /branches/11: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Mon Oct 14 16:42:30 2013
New Revision: 400908

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=400908
Log:
Do not increment the SDP version between 183 and 200 responses.

Bumping the SDP version number can cause interoperability problems
since receivers of the responses will expect that a 200 SDP will
be identical to a previous 183 SDP.

(closes issue ASTERISK-21204)
reported by NITESH BANSAL

Patches:
	dont-increment-session-version-in-2xx-after-183.patch uploaded by NITESH BANSAL (License #6418)
........

Merged revisions 400906 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Modified:
    branches/11/   (props changed)
    branches/11/channels/chan_sip.c

Propchange: branches/11/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Modified: branches/11/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/11/channels/chan_sip.c?view=diff&rev=400908&r1=400907&r2=400908
==============================================================================
--- branches/11/channels/chan_sip.c (original)
+++ branches/11/channels/chan_sip.c Mon Oct 14 16:42:30 2013
@@ -7258,6 +7258,7 @@
 {
 	int res = 0;
 	struct sip_pvt *p = ast_channel_tech_pvt(ast);
+	int oldsdp = FALSE;
 
 	if (!p) {
 		ast_debug(1, "Asked to answer channel %s without tech pvt; ignoring\n",
@@ -7268,10 +7269,14 @@
 	if (ast_channel_state(ast) != AST_STATE_UP) {
 		try_suggested_sip_codec(p);
 
+		if (ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT)) {
+			oldsdp = TRUE;
+		}
+
 		ast_setstate(ast, AST_STATE_UP);
 		ast_debug(1, "SIP answering channel: %s\n", ast_channel_name(ast));
 		ast_rtp_instance_update_source(p->rtp);
-		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE, TRUE);
+		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, oldsdp, TRUE);
 		ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 	}
 	sip_pvt_unlock(p);