Merged revisions 41768 via svnmerge from
authorJoshua Colp <jcolp@digium.com>
Fri, 1 Sep 2006 22:51:46 +0000 (22:51 +0000)
committerJoshua Colp <jcolp@digium.com>
Fri, 1 Sep 2006 22:51:46 +0000 (22:51 +0000)
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r41768 | file | 2006-09-01 18:49:07 -0400 (Fri, 01 Sep 2006) | 2 lines

Only wipe the redirected audio & video IP/port if it's specified, and trigger a reinvite.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41769 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 072b41f..643cec3 100644 (file)
@@ -16169,14 +16169,18 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
                return 0;
        }
 
-       if (rtp) 
+       if (rtp) {
                changed |= ast_rtp_get_peer(rtp, &p->redirip);
-       else
+       } else if (p->redirip.sin_addr.s_addr || ntohs(p->redirip.sin_port) != 0) {
                memset(&p->redirip, 0, sizeof(p->redirip));
-       if (vrtp)
+               changed = 1;
+       }
+       if (vrtp) {
                changed |= ast_rtp_get_peer(vrtp, &p->vredirip);
-       else
+       } else if (p->vredirip.sin_addr.s_addr || ntohs(p->vredirip.sin_port) != 0) {
                memset(&p->vredirip, 0, sizeof(p->vredirip));
+               changed = 1;
+       }
        if (codecs && (p->redircodecs != codecs)) {
                p->redircodecs = codecs;
                changed = 1;