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

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@41768 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index aa6e69a896795ba0e91ca8cd36d281e6c73129fb..7c7552f638508246d8a8500fe8e06015c08014fb 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -12937,14 +12937,18 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
        if (!p) 
                return -1;
        ast_mutex_lock(&p->lock);
-       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;