Merge "channels/chan_sip.c: use binding IP address for outgoing TCP SIP connections"
[asterisk/asterisk.git] / channels / chan_sip.c
index d60927d..e7c15bc 100644 (file)
@@ -13097,7 +13097,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
        /* Opus mandates 2 channels in rtpmap */
        if (ast_format_cmp(format, ast_format_opus) == AST_FORMAT_CMP_EQUAL) {
                ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u/2\r\n", rtp_code, mime, rate);
-       } else if ((35 <= rtp_code) || !(sip_cfg.compactheaders)) {
+       } else if ((AST_RTP_PT_LAST_STATIC < rtp_code) || !(sip_cfg.compactheaders)) {
                ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u\r\n", rtp_code, mime, rate);
        }
 
@@ -28952,7 +28952,7 @@ static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct as
                                        return -1;
                                }
                                if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
-                                       ast_queue_control(p->owner, AST_CONTROL_SRCCHANGE);
+                                       ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
                                }
                        }
                        sched_check_pendings(p);