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authorHolger Hans Peter Freyther <holger@moiji-mobile.com>2015-08-04 13:32:09 +0200
committerHolger Hans Peter Freyther <holger@moiji-mobile.com>2015-08-04 14:57:49 +0200
commit05a09059145ff4bfb1a09ec50e22fb75f2f4ca1e (patch)
treea87bf66d99d01d453b15407a3254bc2abae247be
parent38b66fe3785595dabb59e9eec30d3809d64d54d6 (diff)
mncc: Implement the direct RTP mode for ip based systems
For the LCR rtp-bridge audio should directly flow to the remote system. In contrast to the original patch audio will now flow directly from the BTS to the remote system. This assumes that BTS and the remote system are in the same network segment and can directly communicate. There are various limitations in the first iteration of the implementation: We could (and in the future) should delay the assignment but currently we are forced to pick the channel and move it to the audio state. In case we are located on a SDCCH we always need to change but if we are on a TCH we could send the ipa.CRCX and change the audio state a lot later. The net effect is that the audio codec selection needs to be done in the NITB code and not in the system connected to it. This only works with ip based systems. For E1 systems one could still use the RTP socket or even try to move this out of the process. There is no code for handover handling and it relies on the remote system dealing with the SSRC change of the system.
-rw-r--r--openbsc/include/openbsc/gsm_data.h4
-rw-r--r--openbsc/src/libmsc/gsm_04_08.c213
2 files changed, 217 insertions, 0 deletions
diff --git a/openbsc/include/openbsc/gsm_data.h b/openbsc/include/openbsc/gsm_data.h
index 90f3c80..bed04e3 100644
--- a/openbsc/include/openbsc/gsm_data.h
+++ b/openbsc/include/openbsc/gsm_data.h
@@ -120,6 +120,10 @@ struct gsm_subscriber_connection {
/* Are we part of a special "silent" call */
int silent_call;
+ /* MNCC rtp bridge markers */
+ int mncc_rtp_bridge;
+ int mncc_rtp_create_pending;
+
/* bsc structures */
struct osmo_bsc_sccp_con *sccp_con;
diff --git a/openbsc/src/libmsc/gsm_04_08.c b/openbsc/src/libmsc/gsm_04_08.c
index 02ffe58..f20eb08 100644
--- a/openbsc/src/libmsc/gsm_04_08.c
+++ b/openbsc/src/libmsc/gsm_04_08.c
@@ -67,6 +67,8 @@
void *tall_locop_ctx;
void *tall_authciphop_ctx;
+static int tch_rtp_signal(struct gsm_lchan *lchan, int signal);
+
static int gsm0408_loc_upd_acc(struct gsm_subscriber_connection *conn);
static int gsm48_tx_simple(struct gsm_subscriber_connection *conn,
uint8_t pdisc, uint8_t msg_type);
@@ -1519,6 +1521,10 @@ static int handle_abisip_signal(unsigned int subsys, unsigned int signal,
if (subsys != SS_ABISIP)
return 0;
+ /* RTP bridge handling */
+ if (lchan->conn && lchan->conn->mncc_rtp_bridge)
+ return tch_rtp_signal(lchan, signal);
+
/* in case we use direct BTS-to-BTS RTP */
if (ipacc_rtp_direct)
return 0;
@@ -2905,11 +2911,211 @@ static int gsm48_cc_rx_userinfo(struct gsm_trans *trans, struct msgb *msg)
static int _gsm48_lchan_modify(struct gsm_trans *trans, void *arg)
{
struct gsm_mncc *mode = arg;
+ struct gsm_lchan *lchan = trans->conn->lchan;
+
+ /*
+ * We were forced to make an assignment a lot earlier and
+ * we should avoid sending another assignment that might
+ * even lead to a different kind of lchan (TCH/F vs. TCH/H).
+ * In case of rtp-bridge it is too late to change things
+ * here.
+ */
+ if (trans->conn->mncc_rtp_bridge && lchan->tch_mode != GSM48_CMODE_SIGN)
+ return 0;
return gsm0808_assign_req(trans->conn, mode->lchan_mode,
trans->conn->lchan->type != GSM_LCHAN_TCH_H);
}
+static void mncc_recv_rtp(struct gsm_network *net, uint32_t callref,
+ int cmd, uint32_t addr, uint16_t port, uint32_t payload_type,
+ uint32_t payload_msg_type)
+{
+ uint8_t data[sizeof(struct gsm_mncc)];
+ struct gsm_mncc_rtp *rtp;
+
+ memset(&data, 0, sizeof(data));
+ rtp = (struct gsm_mncc_rtp *) &data[0];
+
+ rtp->callref = callref;
+ rtp->msg_type = cmd;
+ rtp->ip = addr;
+ rtp->port = port;
+ rtp->payload_type = payload_type;
+ rtp->payload_msg_type = payload_msg_type;
+ mncc_recvmsg(net, NULL, cmd, (struct gsm_mncc *)data);
+}
+
+static void mncc_recv_rtp_sock(struct gsm_network *net, struct gsm_trans *trans, int cmd)
+{
+ struct gsm_lchan *lchan;
+ int msg_type;
+
+ lchan = trans->conn->lchan;
+ switch (lchan->abis_ip.rtp_payload) {
+ case RTP_PT_GSM_FULL:
+ msg_type = GSM_TCHF_FRAME;
+ break;
+ case RTP_PT_GSM_EFR:
+ msg_type = GSM_TCHF_FRAME_EFR;
+ break;
+ case RTP_PT_GSM_HALF:
+ msg_type = GSM_TCHH_FRAME;
+ break;
+ case RTP_PT_AMR:
+ msg_type = GSM_TCH_FRAME_AMR;
+ break;
+ default:
+ LOGP(DMNCC, LOGL_ERROR, "%s unknown payload type %d\n",
+ gsm_lchan_name(lchan), lchan->abis_ip.rtp_payload);
+ msg_type = 0;
+ break;
+ }
+
+ return mncc_recv_rtp(net, trans->callref, cmd,
+ lchan->abis_ip.bound_ip,
+ lchan->abis_ip.bound_port,
+ lchan->abis_ip.rtp_payload,
+ msg_type);
+}
+
+static void mncc_recv_rtp_err(struct gsm_network *net, uint32_t callref, int cmd)
+{
+ return mncc_recv_rtp(net, callref, cmd, 0, 0, 0, 0);
+}
+
+static int tch_rtp_create(struct gsm_network *net, uint32_t callref)
+{
+ struct gsm_bts *bts;
+ struct gsm_lchan *lchan;
+ struct gsm_trans *trans;
+
+ /* Find callref */
+ trans = trans_find_by_callref(net, callref);
+ if (!trans) {
+ LOGP(DMNCC, LOGL_ERROR, "RTP create for non-existing trans\n");
+ mncc_recv_rtp_err(net, callref, MNCC_RTP_CREATE);
+ return -EIO;
+ }
+ log_set_context(BSC_CTX_SUBSCR, trans->subscr);
+ if (!trans->conn) {
+ LOGP(DMNCC, LOGL_NOTICE, "RTP create for trans without conn\n");
+ mncc_recv_rtp_err(net, callref, MNCC_RTP_CREATE);
+ return 0;
+ }
+
+ lchan = trans->conn->lchan;
+ bts = lchan->ts->trx->bts;
+ if (!is_ipaccess_bts(bts)) {
+ /*
+ * I want this to be straight forward and have no audio flow
+ * through the nitb/osmo-mss system. This currently means that
+ * this will not work with BS11/Nokia type BTS. We would need
+ * to have a trau<->rtp bridge for these but still preferable
+ * in another process.
+ */
+ LOGP(DMNCC, LOGL_ERROR, "RTP create only works with IP systems\n");
+ mncc_recv_rtp_err(net, callref, MNCC_RTP_CREATE);
+ return -EINVAL;
+ }
+
+ trans->conn->mncc_rtp_bridge = 1;
+ /*
+ * *sigh* we need to pick a codec now. Pick the most generic one
+ * right now and hope we could fix that later on. This is very
+ * similiar to the above routine.
+ * TODO: Use the default codec version...
+ */
+ if (lchan->tch_mode == GSM48_CMODE_SIGN) {
+ trans->conn->mncc_rtp_create_pending = 1;
+ /* TODO... transport or fix the default type... */
+ return gsm0808_assign_req(trans->conn, GSM48_CMODE_SPEECH_V1,
+ lchan->type != GSM_LCHAN_TCH_H);
+ }
+
+ mncc_recv_rtp_sock(trans->net, trans, MNCC_RTP_CREATE);
+ return 0;
+}
+
+static int tch_rtp_connect(struct gsm_network *net, void *arg)
+{
+ struct gsm_lchan *lchan;
+ struct gsm_trans *trans;
+ struct gsm_mncc_rtp *rtp = arg;
+
+ /* Find callref */
+ trans = trans_find_by_callref(net, rtp->callref);
+ if (!trans) {
+ LOGP(DMNCC, LOGL_ERROR, "RTP connect for non-existing trans\n");
+ mncc_recv_rtp_err(net, rtp->callref, MNCC_RTP_CONNECT);
+ return -EIO;
+ }
+ log_set_context(BSC_CTX_SUBSCR, trans->subscr);
+ if (!trans->conn) {
+ LOGP(DMNCC, LOGL_ERROR, "RTP connect for trans without conn\n");
+ mncc_recv_rtp_err(net, rtp->callref, MNCC_RTP_CONNECT);
+ return 0;
+ }
+
+ lchan = trans->conn->lchan;
+
+ /* TODO: Check if payload_msg_type is compatible with what we have */
+ if (rtp->payload_type != lchan->abis_ip.rtp_payload) {
+ LOGP(DMNCC, LOGL_ERROR, "RTP connect with different RTP payload\n");
+ mncc_recv_rtp_err(net, rtp->callref, MNCC_RTP_CONNECT);
+ }
+
+ /*
+ * FIXME: payload2 can't be sent with MDCX as the osmo-bts code
+ * complains about both rtp and rtp payload2 being present in the
+ * same package!
+ */
+ return rsl_ipacc_mdcx(lchan, rtp->ip, rtp->port, 0);
+}
+
+static int tch_rtp_signal(struct gsm_lchan *lchan, int signal)
+{
+ struct gsm_network *net;
+ struct gsm_trans *tmp, *trans = NULL;
+
+ net = lchan->ts->trx->bts->network;
+ llist_for_each_entry(tmp, &net->trans_list, entry) {
+ if (!tmp->conn)
+ continue;
+ if (tmp->conn->lchan != lchan)
+ continue;
+ trans = tmp;
+ break;
+ }
+
+ if (!trans) {
+ LOGP(DMNCC, LOGL_ERROR, "%s IPA abis signal but no transaction.\n",
+ gsm_lchan_name(lchan));
+ return 0;
+ }
+
+ switch (signal) {
+ case S_ABISIP_CRCX_ACK:
+ if (lchan->conn->mncc_rtp_create_pending) {
+ lchan->conn->mncc_rtp_create_pending = 0;
+ LOGP(DMNCC, LOGL_NOTICE, "%s sending pending RTP create ind.\n",
+ gsm_lchan_name(lchan));
+ mncc_recv_rtp_sock(net, trans, MNCC_RTP_CREATE);
+ }
+ break;
+ case S_ABISIP_MDCX_ACK:
+ if (lchan->conn->mncc_rtp_bridge) {
+ LOGP(DMNCC, LOGL_NOTICE, "%s sending pending RTP connect ind.\n",
+ gsm_lchan_name(lchan));
+ mncc_recv_rtp_sock(net, trans, MNCC_RTP_CONNECT);
+ }
+ break;
+ }
+
+ return 0;
+}
+
+
static struct downstate {
uint32_t states;
int type;
@@ -2990,6 +3196,13 @@ int mncc_tx_to_cc(struct gsm_network *net, int msg_type, void *arg)
return tch_recv_mncc(net, data->callref, 0);
case MNCC_FRAME_RECV:
return tch_recv_mncc(net, data->callref, 1);
+ case MNCC_RTP_CREATE:
+ return tch_rtp_create(net, data->callref);
+ case MNCC_RTP_CONNECT:
+ return tch_rtp_connect(net, arg);
+ case MNCC_RTP_FREE:
+ /* unused right now */
+ return -EIO;
case GSM_TCHF_FRAME:
case GSM_TCHF_FRAME_EFR:
case GSM_TCHH_FRAME: