kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-10-12 15:45:18 +00:00
Code Issues Packages Projects Releases Wiki Activity

chan_sip: Do not increment the SDP version between 183 and 200 responses.

Browse Source 
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
This commit is contained in:
Mark Michelson
2013-10-14 22:03:22 +00:00
parent 61b60fab2d
commit 47e910bfe6
1 changed files with 6 additions and 1 deletions
Download Patch File Download Diff File Expand all files Collapse all files

7
channels/chan_sip.c
View File

@@ -7417,6 +7417,7 @@ static int sip_answer(struct ast_channel *ast)
{ {
int res = 0; int res = 0;
struct sip_pvt *p = ast_channel_tech_pvt(ast); struct sip_pvt *p = ast_channel_tech_pvt(ast);
int oldsdp = FALSE;
if (!p) { if (!p) {
ast_debug(1, "Asked to answer channel %s without tech pvt; ignoring\n", 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) { if (ast_channel_state(ast) != AST_STATE_UP) {
try_suggested_sip_codec(p); try_suggested_sip_codec(p);
if (ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT)) {
oldsdp = TRUE;
}
ast_setstate(ast, AST_STATE_UP); ast_setstate(ast, AST_STATE_UP);
ast_debug(1, "SIP answering channel: %s\n", ast_channel_name(ast)); ast_debug(1, "SIP answering channel: %s\n", ast_channel_name(ast));
ast_rtp_instance_update_source(p->rtp); 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); ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
} }
sip_pvt_unlock(p); sip_pvt_unlock(p);