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				synced 2025-11-03 21:43:32 +00:00 
			
		
		
		
	handover: Fix the handover signalling for IP based BTSes
This was reported by Kevin when he was testing handover. The problem is the order of the signal handlers for S_ABISIP_CRCX_ACK. Right now the handover signal handler is called before the one inside the libmsc gsm_04_08.c. This means S_HANDOVER_ACK is signalled _before_ there is a rtp socket created for the channel. The result is that the MDCX will never be sent and the called will not be properly switched _after_ the handover detection. I do not want to play with the order of signal handlers, remove the CRCX ack handling from the handover_logic.c and force the NITB (and later the BSC) to check if the lchan is involved with a handover and do the switching in there. This means right now we do what two signal handlers did in one. Reproduced and tested with the FakeBTS Handover test. Log message: <0004> abis_rsl.c:1954 (bts=1,trx=0,ts=3,ss=0) IPAC_CRCX_ACK ... <000c> gsm_04_08.c:1400 no RTP socket for new_lchan <001a> rtp_proxy.c:533 rtp_socket_create(): success <001a> rtp_proxy.c:615 rtp_socket_bind(rs=0x48703c8, IP=0.0.0.0): ...
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		@@ -11,4 +11,7 @@ int bsc_handover_start(struct gsm_lchan *old_lchan, struct gsm_bts *bts);
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/* clear any operation for this connection */
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void bsc_clear_handover(struct gsm_subscriber_connection *conn, int free_lchan);
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/* Return the old lchan or NULL. This is meant for audio handling */
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struct gsm_lchan *bsc_handover_pending(struct gsm_lchan *new_lchan);
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#endif /* _HANDOVER_H */
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@@ -226,16 +226,6 @@ struct msc_signal_data {
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	struct osmo_msc_data *data;
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};
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/* handover */
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enum signal_ho {
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	S_HANDOVER_ACK,
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};
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struct ho_signal_data {
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	struct gsm_lchan *old_lchan;
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	struct gsm_lchan *new_lchan;
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};
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/* SS_CCCH signals */
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enum signal_ccch {
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	S_CCCH_PAGING_LOAD,
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@@ -329,23 +329,6 @@ static int ho_rsl_detect(struct gsm_lchan *new_lchan)
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	return 0;
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}
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static int ho_ipac_crcx_ack(struct gsm_lchan *new_lchan)
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{
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	struct bsc_handover *ho;
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	struct ho_signal_data sig_ho;
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	ho = bsc_ho_by_new_lchan(new_lchan);
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	if (!ho) {
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		/* it is perfectly normal, we have CRCX even in non-HO cases */
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		return 0;
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	}
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	sig_ho.old_lchan = ho->old_lchan;
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	sig_ho.new_lchan = new_lchan;
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	osmo_signal_dispatch(SS_HO, S_HANDOVER_ACK, &sig_ho);
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	return 0;
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}
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static int ho_logic_sig_cb(unsigned int subsys, unsigned int signal,
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			   void *handler_data, void *signal_data)
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{
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@@ -369,14 +352,6 @@ static int ho_logic_sig_cb(unsigned int subsys, unsigned int signal,
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			return ho_gsm48_ho_fail(lchan);
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		}
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		break;
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	case SS_ABISIP:
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		lchan = signal_data;
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		switch (signal) {
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		case S_ABISIP_CRCX_ACK:
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			return ho_ipac_crcx_ack(lchan);
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			break;
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		}
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		break;
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	default:
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		break;
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	}
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@@ -384,8 +359,16 @@ static int ho_logic_sig_cb(unsigned int subsys, unsigned int signal,
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	return 0;
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}
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struct gsm_lchan *bsc_handover_pending(struct gsm_lchan *new_lchan)
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{
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	struct bsc_handover *ho;
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	ho = bsc_ho_by_new_lchan(new_lchan);
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	if (!ho)
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		return NULL;
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	return ho->old_lchan;
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}
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static __attribute__((constructor)) void on_dso_load_ho_logic(void)
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{
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	osmo_signal_register_handler(SS_LCHAN, ho_logic_sig_cb, NULL);
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	osmo_signal_register_handler(SS_ABISIP, ho_logic_sig_cb, NULL);
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}
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@@ -2,7 +2,7 @@
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 * 3GPP TS 04.08 version 7.21.0 Release 1998 / ETSI TS 100 940 V7.21.0 */
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/* (C) 2008-2009 by Harald Welte <laforge@gnumonks.org>
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 * (C) 2008-2010 by Holger Hans Peter Freyther <zecke@selfish.org>
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 * (C) 2008-2012 by Holger Hans Peter Freyther <zecke@selfish.org>
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 *
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 * All Rights Reserved
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 *
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@@ -51,6 +51,7 @@
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#include <openbsc/silent_call.h>
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#include <openbsc/bsc_api.h>
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#include <openbsc/osmo_msc.h>
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#include <openbsc/handover.h>
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#include <osmocom/abis/e1_input.h>
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#include <osmocom/core/bitvec.h>
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@@ -1378,14 +1379,10 @@ static int setup_trig_pag_evt(unsigned int hooknum, unsigned int event,
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static int tch_recv_mncc(struct gsm_network *net, uint32_t callref, int enable);
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/* handle audio path for handover */
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static int handle_ho_signal(unsigned int subsys, unsigned int signal,
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			    void *handler_data, void *signal_data)
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static int switch_for_handover(struct gsm_lchan *old_lchan,
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			struct gsm_lchan *new_lchan)
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{
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	struct rtp_socket *old_rs, *new_rs, *other_rs;
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	struct ho_signal_data *sig = signal_data;
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	if (subsys != SS_HO || signal != S_HANDOVER_ACK)
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		return 0;
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	if (ipacc_rtp_direct) {
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		LOGP(DHO, LOGL_ERROR, "unable to handover in direct RTP mode\n");
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@@ -1393,15 +1390,15 @@ static int handle_ho_signal(unsigned int subsys, unsigned int signal,
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	}
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	/* RTP Proxy mode */
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	new_rs = sig->new_lchan->abis_ip.rtp_socket;
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	old_rs = sig->old_lchan->abis_ip.rtp_socket;
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	new_rs = new_lchan->abis_ip.rtp_socket;
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	old_rs = old_lchan->abis_ip.rtp_socket;
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	if (!new_rs) {
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		LOGP(DHO, LOGL_ERROR, "no RTP socket for new_lchan\n");
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		return -EIO;
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	}
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	rsl_ipacc_mdcx_to_rtpsock(sig->new_lchan);
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	rsl_ipacc_mdcx_to_rtpsock(new_lchan);
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	if (!old_rs) {
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		LOGP(DHO, LOGL_ERROR, "no RTP socket for old_lchan\n");
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@@ -1413,7 +1410,7 @@ static int handle_ho_signal(unsigned int subsys, unsigned int signal,
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	new_rs->tx_action = old_rs->tx_action;
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	new_rs->transmit = old_rs->transmit;
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	switch (sig->old_lchan->abis_ip.rtp_socket->rx_action) {
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	switch (old_lchan->abis_ip.rtp_socket->rx_action) {
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	case RTP_PROXY:
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		other_rs = old_rs->proxy.other_sock;
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		rtp_socket_proxy(new_rs, other_rs);
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@@ -1435,7 +1432,7 @@ static int handle_ho_signal(unsigned int subsys, unsigned int signal,
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static int handle_abisip_signal(unsigned int subsys, unsigned int signal,
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				 void *handler_data, void *signal_data)
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{
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	struct gsm_lchan *lchan = signal_data;
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	struct gsm_lchan *lchan = signal_data, *old_lchan;
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	int rc;
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	struct gsm_network *net;
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	struct gsm_trans *trans;
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@@ -1476,6 +1473,17 @@ static int handle_abisip_signal(unsigned int subsys, unsigned int signal,
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				tch_recv_mncc(net, trans->callref, 1);
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			}
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		}
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		/*
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		 * TODO: this appears to be too early? Why not until after
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		 * the handover detect or the handover complete?
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		 *
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		 * Do we have a handover pending for this new lchan? In that
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		 * case re-route the audio from the old channel to the new one.
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		 */
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		old_lchan = bsc_handover_pending(lchan);
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		if (old_lchan)
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			switch_for_handover(old_lchan, lchan);
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		break;
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	case S_ABISIP_DLCX_IND:
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		/* the BTS tells us a RTP stream has been disconnected */
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@@ -3253,6 +3261,5 @@ int gsm0408_dispatch(struct gsm_subscriber_connection *conn, struct msgb *msg)
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 */
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static __attribute__((constructor)) void on_dso_load_0408(void)
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{
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	osmo_signal_register_handler(SS_HO, handle_ho_signal, NULL);
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	osmo_signal_register_handler(SS_ABISIP, handle_abisip_signal, NULL);
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}
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@@ -45,6 +45,10 @@ static int handle_abisip_signal(unsigned int subsys, unsigned int signal,
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	switch (signal) {
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	case S_ABISIP_CRCX_ACK:
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		/*
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		 * TODO: handle handover here... then the audio should go to
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		 * the old mgcp port..
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		 */
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		/* we can ask it to connect now */
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		LOGP(DMSC, LOGL_DEBUG, "Connecting BTS to port: %d conn: %d\n",
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		     con->sccp_con->rtp_port, lchan->abis_ip.conn_id);
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