142 lines
4.2 KiB
C
142 lines
4.2 KiB
C
/*
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* ipaccess audio handling
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*
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* (C) 2009-2010 by Holger Hans Peter Freyther <zecke@selfish.org>
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* (C) 2009-2010 by On-Waves
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* All Rights Reserved
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU Affero General Public License as published by
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* the Free Software Foundation; either version 3 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU Affero General Public License for more details.
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*
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* You should have received a copy of the GNU Affero General Public License
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*
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*/
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#include <openbsc/bsc_msc_data.h>
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#include <openbsc/osmo_bsc.h>
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#include <openbsc/abis_rsl.h>
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#include <openbsc/gsm_data.h>
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#include <openbsc/debug.h>
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#include <openbsc/signal.h>
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#include <osmocom/gsm/gsm0808.h>
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#include <osmocom/gsm/gsm0808_utils.h>
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#include <openbsc/osmo_bsc_sigtran.h>
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#include <arpa/inet.h>
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/* Generate and send assignment complete message */
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static int send_aoip_ass_compl(struct gsm_subscriber_connection *conn, struct gsm_lchan *lchan)
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{
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struct msgb *resp;
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struct sockaddr_storage rtp_addr;
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struct sockaddr_in rtp_addr_in;
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struct gsm0808_speech_codec sc;
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OSMO_ASSERT(lchan->abis_ip.ass_compl.valid == true);
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/* Package RTP-Address data */
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memset(&rtp_addr_in, 0, sizeof(rtp_addr_in));
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rtp_addr_in.sin_family = AF_INET;
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rtp_addr_in.sin_port = htons(lchan->abis_ip.bound_port);
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rtp_addr_in.sin_addr.s_addr = htonl(lchan->abis_ip.bound_ip);
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memset(&rtp_addr, 0, sizeof(rtp_addr));
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memcpy(&rtp_addr, &rtp_addr_in, sizeof(rtp_addr_in));
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/* Extrapolate speech codec from speech mode */
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gsm0808_speech_codec_from_chan_type(&sc, lchan->abis_ip.ass_compl.speech_mode);
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/* Generate message */
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resp = gsm0808_create_ass_compl(lchan->abis_ip.ass_compl.rr_cause,
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lchan->abis_ip.ass_compl.chosen_channel,
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lchan->abis_ip.ass_compl.encr_alg_id,
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lchan->abis_ip.ass_compl.speech_mode,
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&rtp_addr,
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&sc,
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NULL);
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if (!resp) {
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LOGP(DMSC, LOGL_ERROR, "Failed to generate assignment completed message!\n"); \
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return -EINVAL;
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}
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return osmo_bsc_sigtran_send(conn->sccp_con, resp);
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}
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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_subscriber_connection *con;
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struct gsm_lchan *lchan = signal_data;
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int rc;
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uint32_t rtp_ip;
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if (subsys != SS_ABISIP)
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return 0;
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con = lchan->conn;
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if (!con || !con->sccp_con)
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return 0;
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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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/* If AoIP is in use, the rtp_ip, which has been communicated
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* via the A interface as connect_ip */
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if(con->sccp_con->rtp_ip)
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rtp_ip = con->sccp_con->rtp_ip;
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else
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rtp_ip = ntohl(INADDR_ANY);
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rc = rsl_ipacc_mdcx(lchan, rtp_ip,
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con->sccp_con->rtp_port,
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lchan->abis_ip.rtp_payload2);
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if (rc < 0) {
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LOGP(DMSC, LOGL_ERROR, "Failed to send MDCX: %d\n", rc);
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return rc;
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}
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break;
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case S_ABISIP_MDCX_ACK:
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if (con->ho_lchan) {
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/* NOTE: When an ho_lchan exists, the MDCX is part of an
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* handover operation (intra-bsc). This means we will not
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* inform the MSC about the event, which means that no
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* assignment complete message is transmitted */
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LOGP(DMSC, LOGL_INFO," RTP connection handover complete\n");
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} else if (is_ipaccess_bts(con->bts) && con->sccp_con->rtp_ip) {
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/* NOTE: This is only relevant on AoIP networks with
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* IPA based base stations. See also osmo_bsc_api.c,
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* function bsc_assign_compl() */
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LOGP(DMSC, LOGL_INFO, "Tx MSC ASSIGN COMPL (POSTPONED)\n");
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if (send_aoip_ass_compl(con, lchan) != 0)
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return -EINVAL;
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}
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break;
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break;
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}
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return 0;
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}
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int osmo_bsc_audio_init(struct gsm_network *net)
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{
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osmo_signal_register_handler(SS_ABISIP, handle_abisip_signal, net);
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return 0;
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}
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