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authorPhilipp Maier <pmaier@sysmocom.de>2017-04-09 12:32:51 +0200
committerNeels Hofmeyr <neels@hofmeyr.de>2017-08-30 14:09:31 +0200
commit39f62bbcbf4309492a6d5bc07213cd74f650d41e (patch)
tree6d56d892a52151c67452c9ac08c898a152181c68 /src/osmo-bsc/osmo_bsc_bssap.c
parent88d8bcb752a611c2c23fb442362665ecd7e31ba0 (diff)
Implement AoIP, port to M3UA SIGTRAN (large addition and refactoring)
This was originally a long series of commits converging to the final result seen in this patch. It does not make much sense to review the smaller steps' trial and error, we need to review this entire change as a whole. Implement AoIP in osmo-msc and osmo-bsc. Change over to the new libosmo-sigtran API with support for proper SCCP/M3UA/SCTP stacking, as mandated by 3GPP specifications for the IuCS and IuPS interfaces. From here on, a separate osmo-stp process is required for SCCP routing between OsmoBSC / OsmoHNBGW <-> OsmoMSC / OsmoSGSN jenkins.sh: build from libosmo-sccp and osmo-iuh master branches now for new M3UA SIGTRAN. Patch-by: pmaier, nhofmeyr, laforge Change-Id: I5ae4e05ee7c57cad341ea5e86af37c1f6b0ffa77
Diffstat (limited to 'src/osmo-bsc/osmo_bsc_bssap.c')
-rw-r--r--src/osmo-bsc/osmo_bsc_bssap.c307
1 files changed, 240 insertions, 67 deletions
diff --git a/src/osmo-bsc/osmo_bsc_bssap.c b/src/osmo-bsc/osmo_bsc_bssap.c
index 100f66441..4353b9ad0 100644
--- a/src/osmo-bsc/osmo_bsc_bssap.c
+++ b/src/osmo-bsc/osmo_bsc_bssap.c
@@ -29,11 +29,21 @@
#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 <openbsc/a_reset.h>
+#include <osmocom/core/byteswap.h>
+
+#define IP_V4_ADDR_LEN 4
/*
* helpers for the assignment command
*/
-enum gsm0808_permitted_speech audio_support_to_gsm88(struct gsm_audio_support *audio)
+
+/* Helper function for match_codec_pref(), looks up a matching permitted speech
+ * value for a given msc audio codec pref */
+enum gsm0808_permitted_speech audio_support_to_gsm88(struct gsm_audio_support
+ *audio)
{
if (audio->hr) {
switch (audio->ver) {
@@ -47,8 +57,9 @@ enum gsm0808_permitted_speech audio_support_to_gsm88(struct gsm_audio_support *a
return GSM0808_PERM_HR3;
break;
default:
- LOGP(DMSC, LOGL_ERROR, "Wrong speech mode: %d\n", audio->ver);
- return GSM0808_PERM_FR1;
+ LOGP(DMSC, LOGL_ERROR, "Wrong speech mode: %d\n",
+ audio->ver);
+ return GSM0808_PERM_FR1;
}
} else {
switch (audio->ver) {
@@ -62,12 +73,15 @@ enum gsm0808_permitted_speech audio_support_to_gsm88(struct gsm_audio_support *a
return GSM0808_PERM_FR3;
break;
default:
- LOGP(DMSC, LOGL_ERROR, "Wrong speech mode: %d\n", audio->ver);
+ LOGP(DMSC, LOGL_ERROR, "Wrong speech mode: %d\n",
+ audio->ver);
return GSM0808_PERM_HR1;
}
}
}
+/* Helper function for match_codec_pref(), looks up a matching chan mode for
+ * a given permitted speech value */
enum gsm48_chan_mode gsm88_to_chan_mode(enum gsm0808_permitted_speech speech)
{
switch (speech) {
@@ -83,16 +97,130 @@ enum gsm48_chan_mode gsm88_to_chan_mode(enum gsm0808_permitted_speech speech)
case GSM0808_PERM_FR3:
return GSM48_CMODE_SPEECH_AMR;
break;
+ default:
+ LOGP(DMSC, LOGL_FATAL,
+ "Unsupported permitted speech selected, assuming AMR as channel mode...\n");
+ return GSM48_CMODE_SPEECH_AMR;
+ }
+}
+
+/* Helper function for match_codec_pref(), tests if a given audio support
+ * matches one of the permitted speech settings of the channel type element.
+ * The matched permitted speech value is then also compared against the
+ * speech codec list. (optional, only relevant for AoIP) */
+static bool test_codec_pref(const struct gsm0808_channel_type *ct,
+ const struct gsm0808_speech_codec_list *scl,
+ uint8_t perm_spch)
+{
+ unsigned int i;
+ bool match = false;
+ struct gsm0808_speech_codec sc;
+ int rc;
+
+ /* Try to finde the given permitted speech value in the
+ * codec list of the channel type element */
+ for (i = 0; i < ct->perm_spch_len; i++) {
+ if (ct->perm_spch[i] == perm_spch) {
+ match = true;
+ break;
+ }
+ }
+
+ /* If we do not have a speech codec list to test against,
+ * we just exit early (will be always the case in non-AoIP networks) */
+ if (!scl)
+ return match;
+
+ /* If we failed to match until here, there is no
+ * point in testing further */
+ if (match == false)
+ return false;
+
+ /* Extrapolate speech codec data */
+ rc = gsm0808_speech_codec_from_chan_type(&sc, perm_spch);
+ if (rc < 0)
+ return false;
+
+ /* Try to find extrapolated speech codec data in
+ * the speech codec list */
+ for (i = 0; i < scl->len; i++) {
+ if (memcmp(&sc, &scl->codec[i], sizeof(sc)) == 0)
+ return true;
}
- LOGP(DMSC, LOGL_FATAL, "Should not be reached.\n");
- return GSM48_CMODE_SPEECH_AMR;
+ return false;
+}
+
+/* Helper function for bssmap_handle_assignm_req(), matches the codec
+ * preferences from the MSC with the codec preferences */
+static int match_codec_pref(int *full_rate, enum gsm48_chan_mode *chan_mode,
+ const struct gsm0808_channel_type *ct,
+ const struct gsm0808_speech_codec_list *scl,
+ const struct bsc_msc_data *msc)
+{
+ unsigned int i;
+ uint8_t perm_spch;
+ bool match = false;
+
+ for (i = 0; i < msc->audio_length; i++) {
+ perm_spch = audio_support_to_gsm88(msc->audio_support[i]);
+ if (test_codec_pref(ct, scl, perm_spch)) {
+ match = true;
+ break;
+ }
+ }
+
+ /* Exit without result, in case no match can be deteched */
+ if (!match) {
+ *full_rate = -1;
+ *chan_mode = GSM48_CMODE_SIGN;
+ return -1;
+ }
+
+ /* Check if the result is a half or full rate codec */
+ if (perm_spch == GSM0808_PERM_HR1 || perm_spch == GSM0808_PERM_HR2
+ || perm_spch == GSM0808_PERM_HR3 || perm_spch == GSM0808_PERM_HR4
+ || perm_spch == GSM0808_PERM_HR6)
+ *full_rate = 0;
+ else
+ *full_rate = 1;
+
+ /* Lookup a channel mode for the selected codec */
+ *chan_mode = gsm88_to_chan_mode(perm_spch);
+
+ return 0;
}
static int bssmap_handle_reset_ack(struct bsc_msc_data *msc,
struct msgb *msg, unsigned int length)
{
- LOGP(DMSC, LOGL_NOTICE, "Reset ACK from MSC\n");
+ LOGP(DMSC, LOGL_NOTICE, "RESET ACK from MSC: %s\n",
+ osmo_sccp_addr_name(osmo_ss7_instance_find(msc->a.cs7_instance),
+ &msc->a.msc_addr));
+
+ /* Inform the FSM that controls the RESET/RESET-ACK procedure
+ * that we have successfully received the reset-ack message */
+ a_reset_ack_confirm(msc->a.reset);
+
+ return 0;
+}
+
+/* Handle MSC sided reset */
+static int bssmap_handle_reset(struct bsc_msc_data *msc,
+ struct msgb *msg, unsigned int length)
+{
+ LOGP(DMSC, LOGL_NOTICE, "RESET from MSC: %s\n",
+ osmo_sccp_addr_name(osmo_ss7_instance_find(msc->a.cs7_instance),
+ &msc->a.msc_addr));
+
+ /* Instruct the bsc to close all open sigtran connections and to
+ * close all active channels on the BTS side as well */
+ osmo_bsc_sigtran_reset(msc);
+
+ /* Inform the MSC that we have received the reset request and
+ * that we acted accordingly */
+ osmo_bsc_sigtran_tx_reset_ack(msc);
+
return 0;
}
@@ -199,7 +327,7 @@ static int bssmap_handle_clear_command(struct osmo_bsc_sccp_con *conn,
return -1;
}
- bsc_queue_for_msc(conn, resp);
+ osmo_bsc_sigtran_send(conn, resp);
return 0;
}
@@ -276,7 +404,7 @@ reject:
return -1;
}
- bsc_queue_for_msc(conn, resp);
+ osmo_bsc_sigtran_send(conn, resp);
return -1;
}
@@ -291,99 +419,141 @@ static int bssmap_handle_assignm_req(struct osmo_bsc_sccp_con *conn,
struct msgb *resp;
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;
+ int port, full_rate = -1;
+ bool aoip = false;
+ struct sockaddr_storage rtp_addr;
+ struct sockaddr_in *rtp_addr_in;
+ struct gsm0808_channel_type ct;
+ struct gsm0808_speech_codec_list scl;
+ struct gsm0808_speech_codec_list *scl_ptr = NULL;
+ int rc;
+ const uint8_t *data;
+ char len;
if (!conn->conn) {
- LOGP(DMSC, LOGL_ERROR, "No lchan/msc_data in cipher mode command.\n");
+ LOGP(DMSC, LOGL_ERROR,
+ "No lchan/msc_data in cipher mode command.\n");
return -1;
}
+ msc = conn->msc;
+
tlv_parse(&tp, gsm0808_att_tlvdef(), msg->l4h + 1, length - 1, 0, 0);
+ /* Check for channel type element, if its missing, immediately reject */
if (!TLVP_PRESENT(&tp, GSM0808_IE_CHANNEL_TYPE)) {
LOGP(DMSC, LOGL_ERROR, "Mandatory channel type not present.\n");
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 */
+ data = TLVP_VAL(&tp, GSM0808_IE_AOIP_TRASP_ADDR);
+ len = TLVP_LEN(&tp, GSM0808_IE_AOIP_TRASP_ADDR);
+ rc = gsm0808_dec_aoip_trasp_addr(&rtp_addr, data, len);
+ 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;
+ /* Decode speech codec list (AoIP) */
+ if (aoip) {
+ /* Check for speech codec list element */
+ if (!TLVP_PRESENT(&tp, GSM0808_IE_SPEECH_CODEC_LIST)) {
+ LOGP(DMSC, LOGL_ERROR,
+ "Mandatory speech codec list not present.\n");
+ goto reject;
+ }
- /*
- * Currently we only support a limited subset of all
- * possible channel types. The limitation ends by not using
- * multi-slot, limiting the channel coding, speech...
- */
- if (TLVP_LEN(&tp, GSM0808_IE_CHANNEL_TYPE) < 3) {
- LOGP(DMSC, LOGL_ERROR, "ChannelType len !=3 not supported: %d\n",
- TLVP_LEN(&tp, GSM0808_IE_CHANNEL_TYPE));
- goto reject;
+ /* Decode Speech Codec list */
+ data = TLVP_VAL(&tp, GSM0808_IE_SPEECH_CODEC_LIST);
+ len = TLVP_LEN(&tp, GSM0808_IE_SPEECH_CODEC_LIST);
+ rc = gsm0808_dec_speech_codec_list(&scl, data, len);
+ if (rc < 0) {
+ LOGP(DMSC, LOGL_ERROR,
+ "Unable to decode speech codec list\n");
+ goto reject;
+ }
+ scl_ptr = &scl;
}
- /*
- * Try to figure out if we support the proposed speech codecs. For
- * now we will always pick the full rate codecs.
- */
-
- data = (uint8_t *) TLVP_VAL(&tp, GSM0808_IE_CHANNEL_TYPE);
- if ((data[0] & 0xf) != 0x1) {
- LOGP(DMSC, LOGL_ERROR, "ChannelType != speech: %d\n", data[0]);
+ /* Decode Channel Type element */
+ data = TLVP_VAL(&tp, GSM0808_IE_CHANNEL_TYPE);
+ len = TLVP_LEN(&tp, GSM0808_IE_CHANNEL_TYPE);
+ rc = gsm0808_dec_channel_type(&ct, data, len);
+ if (rc < 0) {
+ LOGP(DMSC, LOGL_ERROR, "unable to decode channel type.\n");
goto reject;
}
- /*
- * go through the list of preferred codecs of our gsm network
- * and try to find it among the permitted codecs. If we found
- * it we will send chan_mode to the right mode and break the
- * inner loop. The outer loop will exit due chan_mode having
- * the correct value.
- */
- full_rate = 0;
- msc = conn->msc;
- for (supported = 0;
- chan_mode == GSM48_CMODE_SIGN && supported < msc->audio_length;
- ++supported) {
-
- int perm_val = audio_support_to_gsm88(msc->audio_support[supported]);
- for (i = 2; i < TLVP_LEN(&tp, GSM0808_IE_CHANNEL_TYPE); ++i) {
- if ((data[i] & 0x7f) == perm_val) {
- chan_mode = gsm88_to_chan_mode(perm_val);
- full_rate = (data[i] & 0x4) == 0;
- break;
- } else if ((data[i] & 0x80) == 0x00) {
- break;
- }
- }
+ /* Currently we only support a limited subset of all
+ * possible channel types. The limitation ends by not using
+ * multi-slot, limiting the channel coding to speech */
+ if (ct.ch_indctr != GSM0808_CHAN_SPEECH) {
+ LOGP(DMSC, LOGL_ERROR,
+ "Unsupported channel type, currently only speech is supported!\n");
+ goto reject;
}
- if (chan_mode == GSM48_CMODE_SIGN) {
+ /* Match codec information from the assignment command against the
+ * local preferences of the BSC */
+ rc = match_codec_pref(&full_rate, &chan_mode, &ct, scl_ptr, msc);
+ if (rc < 0) {
LOGP(DMSC, LOGL_ERROR, "No supported audio type found.\n");
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);
reject:
- resp = gsm0808_create_assignment_failure(GSM0808_CAUSE_NO_RADIO_RESOURCE_AVAILABLE, NULL);
+ resp =
+ gsm0808_create_assignment_failure
+ (GSM0808_CAUSE_NO_RADIO_RESOURCE_AVAILABLE, NULL);
if (!resp) {
LOGP(DMSC, LOGL_ERROR, "Channel allocation failure.\n");
return -1;
}
- bsc_queue_for_msc(conn, resp);
+ osmo_bsc_sigtran_send(conn, resp);
return -1;
}
@@ -404,6 +574,9 @@ static int bssmap_rcvmsg_udt(struct bsc_msc_data *msc,
case BSS_MAP_MSG_RESET_ACKNOWLEDGE:
ret = bssmap_handle_reset_ack(msc, msg, length);
break;
+ case BSS_MAP_MSG_RESET:
+ ret = bssmap_handle_reset(msc, msg, length);
+ break;
case BSS_MAP_MSG_PAGING:
ret = bssmap_handle_paging(msc, msg, length);
break;
@@ -524,8 +697,8 @@ int bsc_handle_udt(struct bsc_msc_data *msc,
return 0;
}
-int bsc_handle_dt1(struct osmo_bsc_sccp_con *conn,
- struct msgb *msg, unsigned int len)
+int bsc_handle_dt(struct osmo_bsc_sccp_con *conn,
+ struct msgb *msg, unsigned int len)
{
if (len < sizeof(struct bssmap_header)) {
LOGP(DMSC, LOGL_ERROR, "The header is too short.\n");