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Merged revisions 336307 via svnmerge from

Browse Source 
https://origsvn.digium.com/svn/asterisk/branches/10

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  r336307 | jrose | 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) | 20 lines
  
  Merged revisions 336294 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines
    
    Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
    
    In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would
    break when starting a call with directmedia. This patch queues a new type of control frame
    so that our RTP bridge loop can properly detect when these situations occur and check to see
    if peers need to be updated in order to send their media to the proper location.
    
    (Closes issue ASTERISK-18340)
    Reported by: Thomas Arimont
    (Closes issue ASTERISK-17725)
    Reported by: kwk
    Tested by: twilson, jrose
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Jonathan Rose
2011-09-16 21:20:02 +00:00
parent 9112b5c75d
commit beae2df26e
4 changed files with 16 additions and 6 deletions
Download Patch File Download Diff File Expand all files Collapse all files

5
channels/chan_sip.c
View File

@@ -20141,7 +20141,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
"Channel: %s\r\nChanneltype: %s\r\nUniqueid: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
p->owner->name, "SIP", p->owner->uniqueid, p->callid, p->fullcontact, p->peername);
} else { /* RE-invite */
ast_queue_frame(p->owner, &ast_null_frame);
ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
}
} else {
/* It's possible we're getting an 200 OK after we've tried to disconnect
@@ -23024,6 +23024,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
} else {
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL)), p->session_modify == TRUE ? FALSE : TRUE, FALSE);
ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
}
}
@@ -29286,7 +29287,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
memset(&p->tredirip, 0, sizeof(p->tredirip));
changed = 1;
}
if (cap && !(ast_format_cap_is_empty(cap) && !(ast_format_cap_identical(p->redircaps, cap)))) {
if (cap && !ast_format_cap_is_empty(cap) && !ast_format_cap_identical(p->redircaps, cap)) {
ast_format_cap_copy(p->redircaps, cap);
changed = 1;
}