From: Mark Michelson <mmichelson@digium.com>
Date: Mon, 14 Oct 2013 21:42:30 +0000 (+0000)
Subject: chan_sip: Do not increment the SDP version between 183 and 200 responses.
X-Git-Tag: 11.7.0-rc1~33
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=23b6a34d452d1154521f06a6e11a813cfcb3169c;p=thirdparty%2Fasterisk.git

chan_sip: 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400908 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index c68e90a4b1..0580ebb9f1 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -7258,6 +7258,7 @@ static int sip_answer(struct ast_channel *ast)
 {
 	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 @@ static int sip_answer(struct ast_channel *ast)
 	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);