diff -urN asterisk-1.6.1.11/channels/chan_sip.c asterisk-1.6.1.11-constantssrc/channels/chan_sip.c
--- asterisk-1.6.1.11/channels/chan_sip.c	2009-11-30 15:51:28.000000000 -0500
+++ asterisk-1.6.1.11-constantssrc/channels/chan_sip.c	2009-12-03 21:20:39.000000000 -0500
@@ -6766,6 +6766,10 @@
 		ast_rtp_set_rtptimeout(p->rtp, global_rtptimeout);
 		ast_rtp_set_rtpholdtimeout(p->rtp, global_rtpholdtimeout);
 		ast_rtp_set_rtpkeepalive(p->rtp, global_rtpkeepalive);
+		if (ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
+                        ast_rtp_set_constantssrc(p->rtp);
+                }
+
 		if (p->vrtp) {
 			ast_rtp_setqos(p->vrtp, global_tos_video, global_cos_video, "SIP VRTP");
 			ast_rtp_setdtmf(p->vrtp, 0);
@@ -6773,6 +6777,9 @@
 			ast_rtp_set_rtptimeout(p->vrtp, global_rtptimeout);
 			ast_rtp_set_rtpholdtimeout(p->vrtp, global_rtpholdtimeout);
 			ast_rtp_set_rtpkeepalive(p->vrtp, global_rtpkeepalive);
+			if (ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
+                        	ast_rtp_set_constantssrc(p->vrtp);
+	                }
 		}
 		if (p->trtp) {
 			ast_rtp_setqos(p->trtp, global_tos_text, global_cos_text, "SIP TRTP");
@@ -18987,14 +18994,6 @@
 				ast_debug(1, "No compatible codecs for this SIP call.\n");
 				return -1;
 			}
-			if (ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
-				if (p->rtp) {
-					ast_rtp_set_constantssrc(p->rtp);
-				}
-				if (p->vrtp) {
-					ast_rtp_set_constantssrc(p->vrtp);
-				}
-			}
 		} else {	/* No SDP in invite, call control session */
 			p->jointcapability = p->capability;
 			ast_debug(2, "No SDP in Invite, third party call control\n");