Fix a bug where the codecs of the called party leg were not properly sent back to the caller call leg when reinvited.

(closes issue #13569)
Reported by: bkw918


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@195095 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -18559,11 +18559,8 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
 		changed = 1;
 	}
 	if (codecs) {
-		if ((p->redircodecs != codecs)) {
+		if (p->redircodecs != codecs && (p->jointcapability & codecs) != p->jointcapability) {
 			p->redircodecs = codecs;
-			changed = 1;
-		}
-		if ((p->capability & codecs) != p->capability) {
 			p->jointcapability &= codecs;
 			p->capability &= codecs;
 			changed = 1;

main/rtp.c

@@ -3019,7 +3019,8 @@ static enum ast_bridge_result bridge_native_loop(struct ast_channel *c0, struct
 			oldcodec1 = codec1;
 		}
 		if ((inaddrcmp(&t0, &ac0)) ||
-		    (vp0 && inaddrcmp(&vt0, &vac0))) {
+		    (vp0 && inaddrcmp(&vt0, &vac0)) ||
+		    (codec0 != oldcodec0)) {
 			if (option_debug > 1) {
 				ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
 					c0->name, ast_inet_ntoa(t0.sin_addr), ntohs(t0.sin_port), codec0);