Merged revisions 318337 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r318337 | twilson | 2011-05-09 15:23:15 -0500 (Mon, 09 May 2011) | 18 lines
  
  Merged revisions 318331 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
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    r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) | 12 lines
    
    Don't offer video to directmedia callee unless caller offered it as well
    
    Make sure that when directmedia is enabled, that video is not offered to the
    callee even if it supports it. p->vrtp will not exist since the caller didn't
    offer video.
    
    (closes issue #19195)
    Reported by: one47
    Patches: 
          sip_cant_add_video_rtp uploaded by one47 (license 23)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Terry Wilson 2011-05-10 00:22:02 +00:00
parent 0886204011
commit 07b3742ad2

View file

@ -11069,11 +11069,15 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
doing_directmedia = (!ast_sockaddr_isnull(&p->redirip) && !(ast_format_cap_is_empty(p->redircaps))) ? TRUE : FALSE;
/* Check if we need video in this call */
if ((ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_VIDEO)) && !p->novideo) {
if (p->vrtp) {
ast_format_cap_joint_copy(p->jointcaps, p->redircaps, tmpcap);
if (doing_directmedia && !ast_format_cap_has_type(tmpcap, AST_FORMAT_TYPE_VIDEO)) {
ast_debug(2, "This call needs video offers, but caller probably did not offer it!\n");
} else if (p->vrtp) {
needvideo = TRUE;
ast_debug(2, "This call needs video offers!\n");
} else
} else {
ast_debug(2, "This call needs video offers, but there's no video support enabled!\n");
}
}
/* Check if we need text in this call */
if ((ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_TEXT)) && !p->notext) {