Direct Media calls within private network sometimes get one way audio
[asterisk/asterisk.git] / channels / chan_sip.c
index 2cbef52..cea01d7 100644 (file)
@@ -23075,7 +23075,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
                } else if (!reinvite) {
                        struct ast_sockaddr remote_address = {{0,}};
 
-                       ast_rtp_instance_get_remote_address(p->rtp, &remote_address);
+                       ast_rtp_instance_get_requested_target_address(p->rtp, &remote_address);
                        if (ast_sockaddr_isnull(&remote_address) || (!ast_strlen_zero(p->theirprovtag) && strcmp(p->theirtag, p->theirprovtag))) {
                                ast_log(LOG_WARNING, "Received response: \"200 OK\" from '%s' without SDP\n", p->relatedpeer->name);
                                ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
@@ -32325,7 +32325,7 @@ static int sip_allow_anyrtp_remote(struct ast_channel *chan1, struct ast_rtp_ins
        if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
                struct ast_sockaddr us = { { 0, }, }, them = { { 0, }, };
 
-               ast_rtp_instance_get_remote_address(instance, &them);
+               ast_rtp_instance_get_requested_target_address(instance, &them);
                ast_rtp_instance_get_local_address(instance, &us);
 
                if (ast_apply_acl(acl, &them, "SIP Direct Media ACL: ") == AST_SENSE_DENY) {