--- channels/chan_sip.c.orig	2011-12-08 14:04:26.811531771 +0100
+++ channels/chan_sip.c	2011-12-08 16:51:58.145534831 +0100
@@ -1081,6 +1081,7 @@ static int global_match_auth_username;
 static int global_relaxdtmf;		/*!< Relax DTMF */
 static int global_prematuremediafilter;	/*!< Enable/disable premature frames in a call (causing 183 early media) */
 static int global_relaxdtmf;			/*!< Relax DTMF */
+static int global_rtpprodhack;		/*!< Prod an RTP stream immediately to get things going */
 static int global_rtptimeout;		/*!< Time out call if no RTP */
 static int global_rtpholdtimeout;	/*!< Time out call if no RTP during hold */
 static int global_rtpkeepalive;		/*!< Send RTP keepalives */
@@ -2643,6 +2644,9 @@ static enum ast_rtp_get_result sip_get_v
 static enum ast_rtp_get_result sip_get_trtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
 static int sip_get_codec(struct ast_channel *chan);
 static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect);
+static void _sip_rtp_prod(const char *function, struct sip_pvt *p);
+#define sip_rtp_prod(p) _sip_rtp_prod(__PRETTY_FUNCTION__, p)
+
 
 /*------ T38 Support --------- */
 static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
@@ -6492,6 +6496,7 @@ static int sip_answer(struct ast_channel
 		ast_rtp_new_source(p->rtp);
 		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
 		ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
+		sip_rtp_prod(p); /* explicitly sip_pvt_lock'ed */
 	}
 	sip_pvt_unlock(p);
 	return res;
@@ -6867,6 +6872,7 @@ static int sip_indicate(struct ast_chann
 			p->invitestate = INV_EARLY_MEDIA;
 			transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
 			ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+			sip_rtp_prod(p); /* explicitly sip_pvt_lock'ed */
 			break;
 		}
 		res = -1;
@@ -7424,6 +7430,19 @@ static struct ast_frame *sip_read(struct
 	return fr;
 }
 
+/*! \brief Poke media session. This is sometimes required to break through a
+ *  stand off where both parties are waiting for the other one to send.
+ *  This is especially an issue when both:
+ *  (a) we're behind NAT and our peer is not, and
+ *  (b) we're an asterisk and we like to wait for incoming RTP before replying
+ */
+static void _sip_rtp_prod(char const *function, struct sip_pvt *dialog)
+{
+	if (global_rtpprodhack) {
+		ast_verb(3, "Prodding channel '%s' RTP (triggered by %s)\n", dialog->owner->name, function);
+		ast_rtp_sendcng(dialog->rtp, 0);
+	}
+}
 
 /*! \brief Generate 32 byte random string for callid's etc */
 static char *generate_random_string(char *buf, size_t size)
@@ -18379,6 +18398,7 @@ static void handle_response_invite(struc
 				ast_queue_control(p->owner, AST_CONTROL_RINGING);
 			}
 		}
+		sip_rtp_prod(p); /* sip_pvt_locked in handle_request_do */
 		check_pendings(p);
 		break;
 
@@ -18467,6 +18487,7 @@ static void handle_response_invite(struc
 		p->invitestate = INV_TERMINATED;
 		ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 		xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
+		sip_rtp_prod(p); /* sip_pvt_locked in handle_request_do */
 		check_pendings(p);
 		break;
 
@@ -23179,8 +23200,8 @@ static void check_rtp_timeout(struct sip
 	/* If we have no RTP or no active owner, no need to check timers */
 	if (!dialog->rtp || !dialog->owner)
 		return;
-	/* If the call is not in UP state or redirected outside Asterisk, no need to check timers */
 
+	/* If the call is not in UP state or redirected outside Asterisk, no need to check timers */
 	if (dialog->owner->_state != AST_STATE_UP || dialog->redirip.sin_addr.s_addr)
 		return;
 
@@ -25345,6 +25366,7 @@ static int reload_config(enum channelrel
 	sip_cfg.allowguest = DEFAULT_ALLOWGUEST;
 	global_callcounter = DEFAULT_CALLCOUNTER;
 	global_match_auth_username = FALSE;		/*!< Match auth username if available instead of From: Default off. */
+	global_rtpprodhack = 0;
 	global_rtptimeout = 0;
 	global_rtpholdtimeout = 0;
 	global_rtpkeepalive = DEFAULT_RTPKEEPALIVE;
@@ -25533,6 +25555,8 @@ static int reload_config(enum channelrel
 			global_relaxdtmf = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "vmexten")) {
 			ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten));
+		} else if (!strcasecmp(v->name, "rtpprodhack")) {
+			global_rtpprodhack = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "rtptimeout")) {
 			if ((sscanf(v->value, "%30d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) {
 				ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);