chan_sip: Do not increment the SDP version between 183 and 200 responses.
authorMark Michelson <mmichelson@digium.com>
Mon, 14 Oct 2013 22:03:22 +0000 (22:03 +0000)
committerMark Michelson <mmichelson@digium.com>
Mon, 14 Oct 2013 22:03:22 +0000 (22:03 +0000)
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
........

Merged revisions 400908 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 400910 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400912 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index f94f457..b1caa1f 100644 (file)
@@ -7417,6 +7417,7 @@ 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",
@@ -7427,10 +7428,14 @@ 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);