diff -urN asterisk-1.6.1.10/channels/chan_sip.c asterisk-1.6.1.10.fixed/channels/chan_sip.c
--- asterisk-1.6.1.10/channels/chan_sip.c	2009-11-13 10:57:13.000000000 -0500
+++ asterisk-1.6.1.10.fixed/channels/chan_sip.c	2009-11-25 10:30:29.000000000 -0500
@@ -4762,9 +4762,6 @@
 		ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout);
 		ast_rtp_set_rtpholdtimeout(dialog->rtp, peer->rtpholdtimeout);
 		ast_rtp_set_rtpkeepalive(dialog->rtp, peer->rtpkeepalive);
-		if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
-			ast_rtp_set_constantssrc(dialog->rtp);
-		}
 		/* Set Frame packetization */
 		ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
 		dialog->autoframing = peer->autoframing;
@@ -4775,9 +4772,6 @@
 		ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout);
 		ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout);
 		ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive);
-		if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
-			ast_rtp_set_constantssrc(dialog->vrtp);
-		}
 	}
 	if (dialog->trtp) { /* Realtime text */
 		ast_rtp_setdtmf(dialog->trtp, 0);
@@ -18978,14 +18972,6 @@
 				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");
diff -urN asterisk-1.6.1.10/include/asterisk/rtp.h asterisk-1.6.1.10.fixed/include/asterisk/rtp.h
--- asterisk-1.6.1.10/include/asterisk/rtp.h	2009-09-30 14:58:49.000000000 -0400
+++ asterisk-1.6.1.10.fixed/include/asterisk/rtp.h	2009-11-25 10:30:45.000000000 -0500
@@ -210,9 +210,6 @@
 
 int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc);
 
-/*! \brief When changing sources, don't generate a new SSRC */
-void ast_rtp_set_constantssrc(struct ast_rtp *rtp);
-
 void ast_rtp_new_source(struct ast_rtp *rtp);
 
 /*! \brief  Setting RTP payload types from lines in a SDP description: */
diff -urN asterisk-1.6.1.10/main/rtp.c asterisk-1.6.1.10.fixed/main/rtp.c
--- asterisk-1.6.1.10/main/rtp.c	2009-10-19 19:56:31.000000000 -0400
+++ asterisk-1.6.1.10.fixed/main/rtp.c	2009-11-25 10:30:08.000000000 -0500
@@ -181,7 +181,6 @@
 	struct sockaddr_in strict_rtp_address;  /*!< Remote address information for strict RTP purposes */
 
 	int set_marker_bit:1;           /*!< Whether to set the marker bit or not */
-	unsigned int constantssrc:1;
 	struct rtp_red *red;
 };
 
@@ -2610,18 +2609,10 @@
 	return ast_netsock_set_qos(rtp->s, type_of_service, class_of_service, desc);
 }
 
-void ast_rtp_set_constantssrc(struct ast_rtp *rtp)
-{
-	rtp->constantssrc = 1;
-}
-
 void ast_rtp_new_source(struct ast_rtp *rtp)
 {
 	if (rtp) {
 		rtp->set_marker_bit = 1;
-		if (!rtp->constantssrc) {
-			rtp->ssrc = ast_random();
-		}
 	}
 }