--- channels/chan_sip.c.orig	2011-10-17 19:46:26.000000000 +0200
+++ channels/chan_sip.c	2011-12-09 11:48:43.876869404 +0100
@@ -709,6 +709,7 @@
 
 static int global_relaxdtmf;        /*!< Relax DTMF */
 static int global_prematuremediafilter;   /*!< Enable/disable premature frames in a call (causing 183 early media) */
+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 */
@@ -1321,6 +1322,8 @@
 static struct sip_auth *find_realm_authentication(struct sip_auth_container *credentials, const char *realm);
 
 /*--- Misc functions */
+static void _sip_rtp_prod(const char *function, struct sip_pvt *p);
+#define sip_rtp_prod(p) _sip_rtp_prod(__PRETTY_FUNCTION__, p)
 static void check_rtp_timeout(struct sip_pvt *dialog, time_t t);
 static int reload_config(enum channelreloadreason reason);
 static int expire_register(const void *data);
@@ -6325,6 +6328,7 @@
 		ast_rtp_instance_update_source(p->rtp);
 		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE, TRUE);
 		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;
@@ -6694,6 +6698,7 @@
 			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;
@@ -7292,6 +7297,19 @@
 	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_instance_sendcng(dialog->rtp, 0);
+	}
+}
 
 /*! \brief Generate 32 byte random string for callid's etc */
 static char *generate_random_string(char *buf, size_t size)
@@ -19520,6 +19538,7 @@
 				ast_queue_control(p->owner, AST_CONTROL_RINGING);
 			}
 		}
+		sip_rtp_prod(p); /* sip_pvt_locked in handle_request_do */
 		check_pendings(p);
 		break;
 
@@ -19642,6 +19661,7 @@
 		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;
 
@@ -25037,8 +25057,8 @@
 	/* 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 || !ast_sockaddr_isnull(&dialog->redirip))
 		return;
 
@@ -27371,6 +27391,7 @@
 	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;
@@ -27561,6 +27582,8 @@
 			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);