osmo-bsc: Negotiate rtp ip address/port with BTS

This patch adds the support for the RTP IP-Address/Port assignment.
The post communicated via the assignment request is now transmitted
via RSL/IPACC to the BTS. The Response containing the RX-Port at
the BTS side is communicated back to the MSC.

Since we plan to add a private MGCPGW to each BSC, this has to
be extended. Currently it only creates a direct connection to
the BTS. This will be introduced with a future patch.

5 files changed, 136 insertions, 17 deletions

diff --git a/openbsc/include/openbsc/gsm_data_shared.h b/openbsc/include/openbsc/gsm_data_shared.h
index 0790807b1..537da43de 100644
--- a/openbsc/include/openbsc/gsm_data_shared.h
+++ b/openbsc/include/openbsc/gsm_data_shared.h
@@ -252,6 +252,16 @@ struct gsm_lchan {
 		uint8_t speech_mode;
 #ifdef ROLE_BSC
 		struct rtp_socket *rtp_socket;
+
+		/* info we need to postpone the AoIP
+		 * assignment completed message */
+		struct {
+			uint8_t rr_cause;
+			uint8_t chosen_channel;
+			uint8_t encr_alg_id;
+			uint8_t speech_mode;
+			bool valid;
+		} ass_compl;
 #else
 		struct osmo_rtp_socket *rtp_socket;
 #endif
diff --git a/openbsc/include/openbsc/osmo_bsc.h b/openbsc/include/openbsc/osmo_bsc.h
index 5d3ff6a7f..41a6bb003 100644
--- a/openbsc/include/openbsc/osmo_bsc.h
+++ b/openbsc/include/openbsc/osmo_bsc.h
@@ -26,6 +26,7 @@ struct osmo_bsc_sccp_con {
 
 	/* for audio handling */
 	uint16_t cic;
+	uint32_t rtp_ip;
 	int rtp_port;
 
 	/* for advanced ping/pong */
diff --git a/openbsc/src/osmo-bsc/osmo_bsc_api.c b/openbsc/src/osmo-bsc/osmo_bsc_api.c
index d15ec40cd..f7343f743 100644
--- a/openbsc/src/osmo-bsc/osmo_bsc_api.c
+++ b/openbsc/src/osmo-bsc/osmo_bsc_api.c
@@ -435,11 +435,28 @@ static void bsc_assign_compl(struct gsm_subscriber_connection *conn, uint8_t rr_
 	struct msgb *resp;
 	return_when_not_connected(conn);
 
-	LOGP(DMSC, LOGL_INFO, "Tx MSC ASSIGN COMPL\n");
-
-	resp = gsm0808_create_assignment_completed(rr_cause, chosen_channel,
-						   encr_alg_id, speech_model);
-	queue_msg_or_return(resp);
+	if (is_ipaccess_bts(conn->bts) && conn->sccp_con->rtp_ip) {
+		/* NOTE: In a network that makes use of an IPA base station
+		 * and AoIP, we have to wait until the BTS reports its RTP
+		 * IP/Port combination back to BSC via RSL. Unfortunately, the
+		 * IPA protocol sends its Abis assignment complete message
+		 * before it sends its RTP IP/Port via IPACC. So we will now
+		 * postpone the AoIP assignment completed message until we
+		 * know the RTP IP/Port combination. */
+		LOGP(DMSC, LOGL_INFO, "POSTPONE MSC ASSIGN COMPL\n");
+		conn->lchan->abis_ip.ass_compl.rr_cause = rr_cause;
+		conn->lchan->abis_ip.ass_compl.chosen_channel = chosen_channel;
+		conn->lchan->abis_ip.ass_compl.encr_alg_id = encr_alg_id;
+		conn->lchan->abis_ip.ass_compl.speech_mode = speech_model;
+		conn->lchan->abis_ip.ass_compl.valid = true;
+
+	} else {
+		/* NOTE: Send the A assignment complete message immediately. */
+		LOGP(DMSC, LOGL_INFO, "Tx MSC ASSIGN COMPL\n");
+		resp = gsm0808_create_assignment_completed(rr_cause, chosen_channel,
+							   encr_alg_id, speech_model);
+		queue_msg_or_return(resp);
+	}
 }
 
 static void bsc_assign_fail(struct gsm_subscriber_connection *conn,
diff --git a/openbsc/src/osmo-bsc/osmo_bsc_audio.c b/openbsc/src/osmo-bsc/osmo_bsc_audio.c
index 116020900..79b513aa3 100644
--- a/openbsc/src/osmo-bsc/osmo_bsc_audio.c
+++ b/openbsc/src/osmo-bsc/osmo_bsc_audio.c
@@ -26,15 +26,52 @@
 #include <openbsc/gsm_data.h>
 #include <openbsc/debug.h>
 #include <openbsc/signal.h>
+#include <osmocom/gsm/gsm0808.h>
+#include <openbsc/osmo_bsc_sigtran.h>
 
 #include <arpa/inet.h>
 
+/* Generate and send assignment complete message */
+static int send_aoip_ass_compl(struct gsm_subscriber_connection *conn, struct gsm_lchan *lchan)
+{
+	struct msgb *resp;
+	struct sockaddr_storage rtp_addr;
+	struct sockaddr_in rtp_addr_in;
+
+	OSMO_ASSERT(lchan->abis_ip.ass_compl.valid == true);
+
+	/* Package RTP-Address data */
+	memset(&rtp_addr_in, 0, sizeof(rtp_addr_in));
+	rtp_addr_in.sin_family = AF_INET;
+	rtp_addr_in.sin_port = htons(lchan->abis_ip.bound_port);
+	rtp_addr_in.sin_addr.s_addr = htonl(lchan->abis_ip.bound_ip);
+	memset(&rtp_addr, 0, sizeof(rtp_addr));
+	memcpy(&rtp_addr, &rtp_addr_in, sizeof(rtp_addr_in));
+
+	/* Generate message */
+	resp = gsm0808_create_ass_compl(lchan->abis_ip.ass_compl.rr_cause,
+					lchan->abis_ip.ass_compl.chosen_channel,
+					lchan->abis_ip.ass_compl.encr_alg_id,
+					lchan->abis_ip.ass_compl.speech_mode,
+					&rtp_addr,
+					NULL,
+					NULL);
+
+	if (!resp) {
+		LOGP(DMSC, LOGL_ERROR, "Failed to generate assignment completed message!\n"); \
+		return -EINVAL;
+	}
+
+	return osmo_bsc_sigtran_send(conn->sccp_con, resp);
+}
+
 static int handle_abisip_signal(unsigned int subsys, unsigned int signal,
 				 void *handler_data, void *signal_data)
 {
 	struct gsm_subscriber_connection *con;
 	struct gsm_lchan *lchan = signal_data;
 	int rc;
+	uint32_t rtp_ip;
 
 	if (subsys != SS_ABISIP)
 		return 0;
@@ -49,11 +86,19 @@ static int handle_abisip_signal(unsigned int subsys, unsigned int signal,
 		 * TODO: handle handover here... then the audio should go to
 		 * the old mgcp port..
 		 */
+
 		/* we can ask it to connect now */
 		LOGP(DMSC, LOGL_DEBUG, "Connecting BTS to port: %d conn: %d\n",
 		     con->sccp_con->rtp_port, lchan->abis_ip.conn_id);
 
-		rc = rsl_ipacc_mdcx(lchan, ntohl(INADDR_ANY),
+		/* If AoIP is in use, the rtp_ip, which has been communicated
+		 * via the A interface as connect_ip */
+		if(con->sccp_con->rtp_ip)
+			rtp_ip = con->sccp_con->rtp_ip;
+		else
+			rtp_ip = ntohl(INADDR_ANY);
+
+		rc = rsl_ipacc_mdcx(lchan, rtp_ip,
 				    con->sccp_con->rtp_port,
 				    lchan->abis_ip.rtp_payload2);
 		if (rc < 0) {
@@ -61,6 +106,18 @@ static int handle_abisip_signal(unsigned int subsys, unsigned int signal,
 			return rc;
 		}
 		break;
+
+	case S_ABISIP_MDCX_ACK:
+		if (is_ipaccess_bts(con->bts) && con->sccp_con->rtp_ip) {
+			/* NOTE: This is only relevant on AoIP networks with
+			 * IPA based base stations. See also osmo_bsc_api.c,
+			 * function bsc_assign_compl() */
+			LOGP(DMSC, LOGL_INFO, "Tx MSC ASSIGN COMPL (POSTPONED)\n");
+			if (send_aoip_ass_compl(con, lchan) != 0)
+				return -EINVAL;
+		}
+		break;
+	break;
 	}
 
 	return 0;
diff --git a/openbsc/src/osmo-bsc/osmo_bsc_bssap.c b/openbsc/src/osmo-bsc/osmo_bsc_bssap.c
index 57a62a5eb..e705e708d 100644
--- a/openbsc/src/osmo-bsc/osmo_bsc_bssap.c
+++ b/openbsc/src/osmo-bsc/osmo_bsc_bssap.c
@@ -29,7 +29,11 @@
 
 #include <osmocom/gsm/protocol/gsm_08_08.h>
 #include <osmocom/gsm/gsm0808.h>
+#include <osmocom/gsm/gsm0808_utils.h>
 #include <openbsc/osmo_bsc_sigtran.h>
+#include <osmocom/core/byteswap.h>
+
+#define IP_V4_ADDR_LEN 4
 
 /*
  * helpers for the assignment command
@@ -305,10 +309,14 @@ static int bssmap_handle_assignm_req(struct osmo_bsc_sccp_con *conn,
 	struct bsc_msc_data *msc;
 	struct tlv_parsed tp;
 	uint8_t *data;
-	uint8_t timeslot;
-	uint8_t multiplex;
+	uint8_t timeslot = 0;
+	uint8_t multiplex = 0;
 	enum gsm48_chan_mode chan_mode = GSM48_CMODE_SIGN;
 	int i, supported, port, full_rate = -1;
+	bool aoip = false;
+	struct sockaddr_storage rtp_addr;
+	struct sockaddr_in *rtp_addr_in;
+	int rc;
 
 	if (!conn->conn) {
 		LOGP(DMSC, LOGL_ERROR, "No lchan/msc_data in cipher mode command.\n");
@@ -322,15 +330,25 @@ static int bssmap_handle_assignm_req(struct osmo_bsc_sccp_con *conn,
 		goto reject;
 	}
 
-	if (!TLVP_PRESENT(&tp, GSM0808_IE_CIRCUIT_IDENTITY_CODE)) {
-		LOGP(DMSC, LOGL_ERROR, "Identity code missing. Audio routing will not work.\n");
+	/* Detect if a CIC code is present, if so, we use the classic ip.access
+	 * method to calculate the RTP port */
+	if (TLVP_PRESENT(&tp, GSM0808_IE_CIRCUIT_IDENTITY_CODE)) {
+		conn->cic = osmo_load16be(TLVP_VAL(&tp, GSM0808_IE_CIRCUIT_IDENTITY_CODE));
+		timeslot = conn->cic & 0x1f;
+		multiplex = (conn->cic & ~0x1f) >> 5;
+	} else if(TLVP_PRESENT(&tp, GSM0808_IE_AOIP_TRASP_ADDR)) {
+		/* Decode AoIP transport address element */
+		rc = gsm0808_dec_aoip_trasp_addr(&rtp_addr, TLVP_VAL(&tp, GSM0808_IE_AOIP_TRASP_ADDR), TLVP_LEN(&tp, GSM0808_IE_AOIP_TRASP_ADDR));
+		if (rc < 0) {
+			LOGP(DMSC, LOGL_ERROR, "Unable to decode aoip transport address.\n");
+			goto reject;
+		}
+		aoip = true;
+	} else {
+		LOGP(DMSC, LOGL_ERROR, "transport address missing. Audio routing will not work.\n");
 		goto reject;
 	}
 
-	conn->cic = osmo_load16be(TLVP_VAL(&tp, GSM0808_IE_CIRCUIT_IDENTITY_CODE));
-	timeslot = conn->cic & 0x1f;
-	multiplex = (conn->cic & ~0x1f) >> 5;
-
 	/*
 	 * Currently we only support a limited subset of all
 	 * possible channel types. The limitation ends by not using
@@ -383,9 +401,25 @@ static int bssmap_handle_assignm_req(struct osmo_bsc_sccp_con *conn,
 		goto reject;
 	}
 
-	/* map it to a MGCP Endpoint and a RTP port */
-	port = mgcp_timeslot_to_endpoint(multiplex, timeslot);
-	conn->rtp_port = rtp_calculate_port(port, msc->rtp_base);
+	if (aoip == false) {
+		/* map it to a MGCP Endpoint and a RTP port */
+		port = mgcp_timeslot_to_endpoint(multiplex, timeslot);
+		conn->rtp_port = rtp_calculate_port(port, msc->rtp_base);
+		conn->rtp_ip = 0;
+	} else {
+		/* use address / port supplied with the AoIP
+		 * transport address element */
+		if(rtp_addr.ss_family == AF_INET)
+		{
+			rtp_addr_in = (struct sockaddr_in *)&rtp_addr;
+			conn->rtp_port = osmo_ntohs(rtp_addr_in->sin_port);
+			memcpy(&conn->rtp_ip, &rtp_addr_in->sin_addr.s_addr, IP_V4_ADDR_LEN);
+			conn->rtp_ip = osmo_ntohl(conn->rtp_ip);
+		} else {
+			LOGP(DMSC, LOGL_ERROR, "Unsopported addressing scheme. (supports only IPV4)\n");
+			goto reject;
+		}
+	}
 
 	return gsm0808_assign_req(conn->conn, chan_mode, full_rate);