add full SDP codec information to the MNCC socket

This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.

To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.

The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.

Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.

Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.

Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)

Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f

10 files changed, 135 insertions, 8 deletions

diff --git a/include/osmocom/msc/Makefile.am b/include/osmocom/msc/Makefile.am
index faf1048dc..572430309 100644
--- a/include/osmocom/msc/Makefile.am
+++ b/include/osmocom/msc/Makefile.am
@@ -1,6 +1,8 @@
 noinst_HEADERS = \
 	call_leg.h \
+	cc_sdp.h \
 	cell_id_list.h \
+	codec_sdp_cc_t9n.h \
 	db.h \
 	debug.h \
 	e_link.h \
diff --git a/include/osmocom/msc/call_leg.h b/include/osmocom/msc/call_leg.h
index a225b6683..c7d3b9739 100644
--- a/include/osmocom/msc/call_leg.h
+++ b/include/osmocom/msc/call_leg.h
@@ -12,6 +12,7 @@ struct gsm_network;
 struct gsm_trans;
 struct rtp_stream;
 enum rtp_direction;
+struct sdp_audio_codecs;
 
 extern struct osmo_tdef g_mgw_tdefs[];
 
@@ -74,7 +75,8 @@ int call_leg_local_bridge(struct call_leg *cl1, uint32_t call_id1, struct gsm_tr
 int call_leg_ensure_rtp_alloc(struct call_leg *cl, enum rtp_direction dir, uint32_t call_id,
 			      struct gsm_trans *for_trans);
 int call_leg_ensure_ci(struct call_leg *cl, enum rtp_direction dir, uint32_t call_id, struct gsm_trans *for_trans,
-		       const enum mgcp_codecs *codec_if_known, const struct osmo_sockaddr_str *remote_addr_if_known);
+		       const struct sdp_audio_codecs *codecs_if_known,
+		       const struct osmo_sockaddr_str *remote_addr_if_known);
 struct osmo_sockaddr_str *call_leg_local_ip(struct call_leg *cl, enum rtp_direction dir);
 
 void call_leg_rtp_stream_gone(struct call_leg *cl, struct rtp_stream *rtps);
diff --git a/include/osmocom/msc/cc_sdp.h b/include/osmocom/msc/cc_sdp.h
new file mode 100644
index 000000000..549a888d3
--- /dev/null
+++ b/include/osmocom/msc/cc_sdp.h
@@ -0,0 +1,42 @@
+#pragma once
+
+#include <osmocom/gsm/gsm_utils.h>
+#include <osmocom/gsm/mncc.h>
+#include <osmocom/mgcp_client/mgcp_client.h>
+
+#include <osmocom/msc/sdp_msg.h>
+
+struct gsm0808_speech_codec_list;
+
+struct cc_sdp {
+	/* The fixed set of codecs available on the RAN type, per definition. */
+	struct sdp_audio_codecs ran;
+	/* The codecs advertised by the MS Bearer Capabilities */
+	struct sdp_audio_codecs ms;
+	/* If known, the set of codecs the current RAN cell allows / has available.
+	 * This may not be available if the BSC does not issue this information early enough.
+	 * Should be ignored if empty. */
+	struct sdp_audio_codecs cell;
+
+	/* SDP as last received from the remote call leg. */
+	struct sdp_msg remote;
+
+	/* After a channel was assigned, this reflects the chosen codec. */
+	struct sdp_audio_codec assignment;
+
+	/* Resulting choice of supported codecs, usually the intersection of the above,
+	 * and the local RTP address to be sent to the remote call leg. */
+	struct sdp_msg result;
+};
+
+void cc_sdp_init(struct cc_sdp *cc_sdp,
+		 enum osmo_rat_type ran_type,
+		 const struct gsm_mncc_bearer_cap *ms_bearer_cap,
+		 const struct gsm0808_speech_codec_list *codec_list_bss_supported);
+void cc_sdp_set_cell(struct cc_sdp *cc_sdp,
+		     const struct gsm0808_speech_codec_list *codec_list_bss_supported);
+int cc_sdp_filter(struct cc_sdp *cc_sdp);
+
+int cc_sdp_name_buf(char *buf, size_t buflen, const struct cc_sdp *cc_sdp);
+char *cc_sdp_name_c(void *ctx, const struct cc_sdp *cc_sdp);
+const char *cc_sdp_name(const struct cc_sdp *cc_sdp);
diff --git a/include/osmocom/msc/codec_sdp_cc_t9n.h b/include/osmocom/msc/codec_sdp_cc_t9n.h
new file mode 100644
index 000000000..b082f159e
--- /dev/null
+++ b/include/osmocom/msc/codec_sdp_cc_t9n.h
@@ -0,0 +1,58 @@
+/* Routines for translation ("t9n") between SDP codec names and CC/BSSMAP codec constants */
+#pragma once
+
+#include <osmocom/gsm/protocol/gsm_04_08.h>
+#include <osmocom/gsm/protocol/gsm_08_08.h>
+#include <osmocom/mgcp_client/mgcp_client.h>
+#include <osmocom/msc/sdp_msg.h>
+#include <osmocom/gsm/mncc.h>
+
+#define NO_MGCP_CODEC 0xffffffff
+
+extern const struct gsm_mncc_bearer_cap bearer_cap_empty;
+
+enum codec_frhr {
+	CODEC_FRHR_NONE = 0,
+	CODEC_FRHR_FR,
+	CODEC_FRHR_HR,
+};
+
+struct codec_mapping {
+	struct sdp_audio_codec sdp;
+	enum mgcp_codecs mgcp;
+	unsigned int speech_ver_count;
+	enum gsm48_bcap_speech_ver speech_ver[8];
+	uint32_t mncc_payload_msg_type;
+	/* gsm0808_speech_codec_type corresponds to gsm0808_speech_codec[_list]->type */
+	bool has_gsm0808_speech_codec_type;
+	enum gsm0808_speech_codec_type gsm0808_speech_codec_type;
+	enum gsm0808_permitted_speech perm_speech;
+	enum codec_frhr frhr;
+};
+
+extern const struct codec_mapping codec_map[];
+#define foreach_codec_mapping(CODEC_MAPPING) \
+	for ((CODEC_MAPPING) = codec_map; (CODEC_MAPPING) < codec_map + ARRAY_SIZE(codec_map); (CODEC_MAPPING)++)
+
+const struct codec_mapping *codec_mapping_by_speech_ver(enum gsm48_bcap_speech_ver speech_ver);
+const struct codec_mapping *codec_mapping_by_gsm0808_speech_codec_type(enum gsm0808_speech_codec_type sct,
+								       uint16_t cfg);
+const struct codec_mapping *codec_mapping_by_perm_speech(enum gsm0808_permitted_speech perm_speech);
+const struct codec_mapping *codec_mapping_by_subtype_name(const char *subtype_name);
+const struct codec_mapping *codec_mapping_by_mgcp_codec(enum mgcp_codecs mgcp);
+
+int bearer_cap_add_speech_ver(struct gsm_mncc_bearer_cap *bearer_cap, enum gsm48_bcap_speech_ver speech_ver);
+int sdp_audio_codec_add_to_bearer_cap(struct gsm_mncc_bearer_cap *bearer_cap, const struct sdp_audio_codec *codec);
+int sdp_audio_codecs_to_bearer_cap(struct gsm_mncc_bearer_cap *bearer_cap, const struct sdp_audio_codecs *ac);
+int bearer_cap_set_radio(struct gsm_mncc_bearer_cap *bearer_cap);
+
+struct sdp_audio_codec *sdp_audio_codecs_add_speech_ver(struct sdp_audio_codecs *ac,
+							enum gsm48_bcap_speech_ver speech_ver);
+struct sdp_audio_codec *sdp_audio_codecs_add_mgcp_codec(struct sdp_audio_codecs *ac, enum mgcp_codecs mgcp_codec);
+void sdp_audio_codecs_from_bearer_cap(struct sdp_audio_codecs *ac, const struct gsm_mncc_bearer_cap *bc);
+
+void sdp_audio_codecs_from_speech_codec_list(struct sdp_audio_codecs *ac, const struct gsm0808_speech_codec_list *cl);
+
+int sdp_audio_codecs_to_gsm0808_channel_type(struct gsm0808_channel_type *ct, const struct sdp_audio_codecs *ac);
+
+enum mgcp_codecs sdp_audio_codec_to_mgcp_codec(const struct sdp_audio_codec *codec);
diff --git a/include/osmocom/msc/gsm_04_08.h b/include/osmocom/msc/gsm_04_08.h
index 661da42b3..fc8ebd192 100644
--- a/include/osmocom/msc/gsm_04_08.h
+++ b/include/osmocom/msc/gsm_04_08.h
@@ -45,6 +45,11 @@ int gsm48_send_rr_app_info(struct msc_a *msc_a, uint8_t apdu_id, uint8_t apdu_le
 int gsm48_send_rr_ass_cmd(struct gsm_lchan *dest_lchan, struct gsm_lchan *lchan, uint8_t power_class);
 int gsm48_send_ho_cmd(struct gsm_lchan *old_lchan, struct gsm_lchan *new_lchan, uint8_t power_command, uint8_t ho_ref);
 
+void gsm48_cc_rx_setup_cn_local_rtp_port_known(struct gsm_trans *trans);
+void gsm48_cc_rx_call_conf_cn_local_rtp_port_known(struct gsm_trans *trans);
+int cc_cn_local_rtp_port_known(struct gsm_trans *trans);
+int cc_assignment_done(struct gsm_trans *trans);
+
 int mncc_tx_to_cc(struct gsm_network *net, void *arg);
 
 /* convert a ASCII phone number to call-control BCD */
diff --git a/include/osmocom/msc/msc_a.h b/include/osmocom/msc/msc_a.h
index 18973f9af..b9311cd11 100644
--- a/include/osmocom/msc/msc_a.h
+++ b/include/osmocom/msc/msc_a.h
@@ -121,6 +121,9 @@ struct msc_a {
 	 *                   \-------RTP--> (ISUP) <--RTP--> <--RTP-->
 	 */
 	struct {
+		/* Codec List (BSS Supported) as received during Complete Layer 3 Information */
+		struct gsm0808_speech_codec_list codec_list_bss_supported;
+
 		/* All of the RTP stream handling */
 		struct call_leg *call_leg;
 		struct mncc_call *mncc_forwarding_to_remote_ran;
@@ -203,6 +206,7 @@ void msc_a_up_ciph_res(struct msc_a *msc_a, bool success, const char *imeisv);
 bool msc_a_is_accepted(const struct msc_a *msc_a);
 bool msc_a_is_establishing_auth_ciph(const struct msc_a *msc_a);
 
+int msc_a_ensure_cn_local_rtp(struct msc_a *msc_a, struct gsm_trans *cc_trans);
 int msc_a_try_call_assignment(struct gsm_trans *cc_trans);
 
 const char *msc_a_cm_service_type_to_use(enum osmo_cm_service_type cm_service_type);
diff --git a/include/osmocom/msc/msc_ho.h b/include/osmocom/msc/msc_ho.h
index 99956f1e6..a3f60c7f2 100644
--- a/include/osmocom/msc/msc_ho.h
+++ b/include/osmocom/msc/msc_ho.h
@@ -31,7 +31,7 @@
 #include <osmocom/msc/neighbor_ident.h>
 #include <osmocom/msc/ran_msg.h>
 #include <osmocom/msc/mncc_call.h>
-
+#include <osmocom/msc/sdp_msg.h>
 
 struct gsm0808_handover_required;
 
@@ -92,7 +92,7 @@ struct msc_ho_state {
 	struct {
 		/* Saved RTP IP:port and codec in case we need to roll back */
 		struct osmo_sockaddr_str ran_remote_rtp;
-		enum mgcp_codecs codec;
+		struct sdp_audio_codecs codecs;
 	} old_cell;
 };
 
diff --git a/include/osmocom/msc/rtp_stream.h b/include/osmocom/msc/rtp_stream.h
index c53c4f179..c654edd54 100644
--- a/include/osmocom/msc/rtp_stream.h
+++ b/include/osmocom/msc/rtp_stream.h
@@ -5,6 +5,7 @@
 
 #include <osmocom/core/sockaddr_str.h>
 #include <osmocom/mgcp_client/mgcp_client.h>
+#include <osmocom/msc/sdp_msg.h>
 
 struct gsm_trans;
 
@@ -37,9 +38,9 @@ struct rtp_stream {
 	struct osmo_sockaddr_str remote;
 	bool remote_sent_to_mgw;
 
-	bool codec_known;
-	enum mgcp_codecs codec;
-	bool codec_sent_to_mgw;
+	bool codecs_known;
+	struct sdp_audio_codecs codecs;
+	bool codecs_sent_to_mgw;
 
 	struct osmo_mgcpc_ep_ci *ci;
 
@@ -64,8 +65,11 @@ struct rtp_stream *rtp_stream_alloc(struct call_leg *parent_call_leg, enum rtp_d
 int rtp_stream_ensure_ci(struct rtp_stream *rtps, struct osmo_mgcpc_ep *at_endpoint);
 int rtp_stream_do_mdcx(struct rtp_stream *rtps);
 
-void rtp_stream_set_codec(struct rtp_stream *rtps, enum mgcp_codecs codec);
+void rtp_stream_set_codecs(struct rtp_stream *rtps, const struct sdp_audio_codecs *codecs);
+void rtp_stream_set_one_codec(struct rtp_stream *rtps, const struct sdp_audio_codec *codec);
+bool rtp_stream_set_codecs_from_mgcp_codec(struct rtp_stream *rtps, enum mgcp_codecs codec);
 void rtp_stream_set_remote_addr(struct rtp_stream *rtps, const struct osmo_sockaddr_str *r);
+void rtp_stream_set_remote_addr_and_codecs(struct rtp_stream *rtps, const struct sdp_msg *sdp);
 void rtp_stream_set_remote_osmux_cid(struct rtp_stream *rtps, uint8_t osmux_cid);
 int rtp_stream_commit(struct rtp_stream *rtps);
 
diff --git a/include/osmocom/msc/sdp_msg.h b/include/osmocom/msc/sdp_msg.h
index 7ac6036f6..a8ecba573 100644
--- a/include/osmocom/msc/sdp_msg.h
+++ b/include/osmocom/msc/sdp_msg.h
@@ -36,7 +36,10 @@ struct sdp_msg {
 
 const char *sdp_msg_line_end(const char *src);
 
-int sdp_audio_codec_cmp(const struct sdp_audio_codec *a, const struct sdp_audio_codec *b);
+int sdp_audio_codec_cmp(const struct sdp_audio_codec *a, const struct sdp_audio_codec *b,
+			bool cmp_fmtp, bool cmp_payload_type);
+int sdp_audio_codecs_cmp(const struct sdp_audio_codecs *a, const struct sdp_audio_codecs *b,
+			 bool cmp_fmtp, bool cmp_payload_type);
 
 struct sdp_audio_codec *sdp_audio_codec_add(struct sdp_audio_codecs *ac, unsigned int payload_type,
 					    const char *subtype_name, unsigned int rate, const char *fmtp);
diff --git a/include/osmocom/msc/transaction.h b/include/osmocom/msc/transaction.h
index 928b137ae..c091be244 100644
--- a/include/osmocom/msc/transaction.h
+++ b/include/osmocom/msc/transaction.h
@@ -10,6 +10,7 @@
 #include <osmocom/msc/debug.h>
 #include <osmocom/gsm/gsm0411_smc.h>
 #include <osmocom/gsm/gsm0411_smr.h>
+#include <osmocom/msc/cc_sdp.h>
 
 struct vty;
 
@@ -87,6 +88,10 @@ struct gsm_trans {
 	/* bearer capabilities (rate and codec) */
 	struct gsm_mncc_bearer_cap bearer_cap;
 
+	/* if true, TCH_RTP_CREATE is sent after the
+	 * assignment is done */
+	bool tch_rtp_create;
+
 	union {
 		struct {
 
@@ -100,6 +105,8 @@ struct gsm_trans {
 			struct osmo_timer_list timer_guard;
 			struct gsm_mncc msg;	/* stores setup/disconnect/release message */
 			bool mncc_initiated;	/* Whether an MNCC Release is necessary on failure */
+			bool mncc_release_sent; /* Mark when special error handling already did MNCC rel */
+			struct cc_sdp sdp;	/* Track SDP codec info from BSS and remote call leg */
 		} cc;
 		struct {
 			struct gsm411_smc_inst smc_inst;