add codec_filter.h,c

Related: SYS#5066
Change-Id: I4d90f7ca62f2307a7b93dd164aeecbf4bd98ff0a

4 files changed, 277 insertions, 0 deletions

diff --git a/include/osmocom/msc/Makefile.am b/include/osmocom/msc/Makefile.am
index e1cf2cd68..ef6d6ddbb 100644
--- a/include/osmocom/msc/Makefile.am
+++ b/include/osmocom/msc/Makefile.am
@@ -1,6 +1,7 @@
 noinst_HEADERS = \
 	call_leg.h \
 	cell_id_list.h \
+	codec_filter.h \
 	codec_sdp_cc_t9n.h \
 	db.h \
 	debug.h \
diff --git a/include/osmocom/msc/codec_filter.h b/include/osmocom/msc/codec_filter.h
new file mode 100644
index 000000000..16315c791
--- /dev/null
+++ b/include/osmocom/msc/codec_filter.h
@@ -0,0 +1,66 @@
+/* Filter/overlay codec selections for a voice call, across MS, RAN and CN limitations */
+/*
+ * (C) 2019-2022 by sysmocom - s.m.f.c. GmbH <info@sysmocom.de>
+ * All Rights Reserved
+ *
+ * Author: Neels Hofmeyr
+ *
+ * SPDX-License-Identifier: GPL-2.0+
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ */
+#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;
+
+/* Combine various codec selections to obtain a resulting set of codecs allowed by all of them.
+ * Members reflect the different entities/stages that select codecs in a voice call.
+ * Call codec_filter_run() and obtain the resulting set of codecs in codec_filter.result. */
+struct codec_filter {
+	/* 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 bss;
+
+	/* 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.
+	 * The RTP address:port in result.rtp is not modified by codec_filter_run() -- set it once. */
+	struct sdp_msg result;
+};
+
+void codec_filter_init(struct codec_filter *codec_filter);
+void codec_filter_set_ran(struct codec_filter *codec_filter, enum osmo_rat_type ran_type);
+void codec_filter_set_ms_from_bc(struct codec_filter *codec_filter, const struct gsm_mncc_bearer_cap *ms_bearer_cap);
+void codec_filter_set_bss(struct codec_filter *codec_filter,
+			  const struct gsm0808_speech_codec_list *codec_list_bss_supported);
+int codec_filter_set_remote(struct codec_filter *codec_filter, const char *remote_sdp);
+void codec_filter_set_local_rtp(struct codec_filter *codec_filter, const struct osmo_sockaddr_str *rtp);
+int codec_filter_run(struct codec_filter *codec_filter);
+
+int codec_filter_to_str_buf(char *buf, size_t buflen, const struct codec_filter *codec_filter);
+char *codec_filter_to_str_c(void *ctx, const struct codec_filter *codec_filter);
+const char *codec_filter_to_str(const struct codec_filter *codec_filter);
diff --git a/src/libmsc/Makefile.am b/src/libmsc/Makefile.am
index b843a65d8..cd922c390 100644
--- a/src/libmsc/Makefile.am
+++ b/src/libmsc/Makefile.am
@@ -27,6 +27,7 @@ noinst_LIBRARIES = \
 libmsc_a_SOURCES = \
 	call_leg.c \
 	cell_id_list.c \
+	codec_filter.c \
 	codec_sdp_cc_t9n.c \
 	sccp_ran.c \
 	msc_vty.c \
diff --git a/src/libmsc/codec_filter.c b/src/libmsc/codec_filter.c
new file mode 100644
index 000000000..199ef7111
--- /dev/null
+++ b/src/libmsc/codec_filter.c
@@ -0,0 +1,209 @@
+/* Filter/overlay codec selections for a voice call, across MS, RAN and CN limitations */
+/*
+ * (C) 2019-2022 by sysmocom - s.m.f.c. GmbH <info@sysmocom.de>
+ * All Rights Reserved
+ *
+ * Author: Neels Hofmeyr
+ *
+ * SPDX-License-Identifier: GPL-2.0+
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ */
+
+#include <osmocom/gsm/protocol/gsm_08_08.h>
+
+#include <osmocom/msc/codec_filter.h>
+#include <osmocom/msc/codec_sdp_cc_t9n.h>
+#include <osmocom/msc/debug.h>
+
+/* Add all known payload types encountered in GSM networks */
+static void sdp_add_all_mobile_codecs(struct sdp_audio_codecs *ac)
+{
+	/* In order of preference. TODO: make configurable */
+	static const enum gsm48_bcap_speech_ver mobile_codecs[] = {
+		GSM48_BCAP_SV_AMR_F	/*!< 4   GSM FR V3 (FR AMR) */,
+		GSM48_BCAP_SV_AMR_H	/*!< 5   GSM HR V3 (HR_AMR) */,
+		GSM48_BCAP_SV_EFR	/*!< 2   GSM FR V2 (GSM EFR) */,
+		GSM48_BCAP_SV_FR	/*!< 0   GSM FR V1 (GSM FR) */,
+		GSM48_BCAP_SV_HR	/*!< 1   GSM HR V1 (GSM HR) */,
+	};
+	int i;
+	for (i = 0; i < ARRAY_SIZE(mobile_codecs); i++)
+		sdp_audio_codecs_add_speech_ver(ac, mobile_codecs[i]);
+}
+
+/* Add all known AMR payload types encountered in UTRAN networks */
+static void sdp_add_all_utran_codecs(struct sdp_audio_codecs *ac)
+{
+	/* In order of preference. TODO: make configurable */
+	static const enum gsm48_bcap_speech_ver utran_codecs[] = {
+		GSM48_BCAP_SV_AMR_F	/*!< 4   GSM FR V3 (FR AMR) */,
+		GSM48_BCAP_SV_AMR_H	/*!< 5   GSM HR V3 (HR_AMR) */,
+		GSM48_BCAP_SV_AMR_OH	/*!< 11  GSM HR V6 (OHR AMR) */,
+		GSM48_BCAP_SV_AMR_FW	/*!< 8   GSM FR V5 (FR AMR-WB) */,
+		GSM48_BCAP_SV_AMR_OFW	/*!< 6   GSM FR V4 (OFR AMR-WB) */,
+		GSM48_BCAP_SV_AMR_OHW	/*!< 7   GSM HR V4 (OHR AMR-WB) */,
+	};
+	int i;
+	for (i = 0; i < ARRAY_SIZE(utran_codecs); i++)
+		sdp_audio_codecs_add_speech_ver(ac, utran_codecs[i]);
+}
+
+void codec_filter_init(struct codec_filter *codec_filter)
+{
+	*codec_filter = (struct codec_filter){};
+}
+
+void codec_filter_set_ran(struct codec_filter *codec_filter, enum osmo_rat_type ran_type)
+{
+	codec_filter->ran = (struct sdp_audio_codecs){};
+
+	switch (ran_type) {
+	default:
+	case OSMO_RAT_GERAN_A:
+		sdp_add_all_mobile_codecs(&codec_filter->ran);
+		break;
+
+	case OSMO_RAT_UTRAN_IU:
+		sdp_add_all_utran_codecs(&codec_filter->ran);
+		break;
+	}
+}
+
+void codec_filter_set_ms_from_bc(struct codec_filter *codec_filter, const struct gsm_mncc_bearer_cap *ms_bearer_cap)
+{
+	codec_filter->ms = (struct sdp_audio_codecs){0};
+	if (ms_bearer_cap)
+		sdp_audio_codecs_from_bearer_cap(&codec_filter->ms, ms_bearer_cap);
+}
+
+void codec_filter_set_bss(struct codec_filter *codec_filter,
+			  const struct gsm0808_speech_codec_list *codec_list_bss_supported)
+{
+	codec_filter->bss = (struct sdp_audio_codecs){};
+	if (codec_list_bss_supported)
+		sdp_audio_codecs_from_speech_codec_list(&codec_filter->bss, codec_list_bss_supported);
+}
+
+int codec_filter_set_remote(struct codec_filter *codec_filter, const char *remote_sdp)
+{
+	return sdp_msg_from_sdp_str(&codec_filter->remote, remote_sdp);
+}
+
+void codec_filter_set_local_rtp(struct codec_filter *codec_filter, const struct osmo_sockaddr_str *rtp)
+{
+	if (!rtp)
+		codec_filter->result.rtp = (struct osmo_sockaddr_str){0};
+	else
+		codec_filter->result.rtp = *rtp;
+}
+
+/* Render intersections of all known audio codec constraints to reach a resulting choice of favorite audio codec, plus
+ * possible set of alternative audio codecs, in codec_filter->result. (The result.rtp address remains unchanged.) */
+int codec_filter_run(struct codec_filter *codec_filter)
+{
+	struct sdp_audio_codecs *r = &codec_filter->result.audio_codecs;
+	struct sdp_audio_codec *a = &codec_filter->assignment;
+	*r = codec_filter->ran;
+	if (codec_filter->ms.count)
+		sdp_audio_codecs_intersection(r, &codec_filter->ms, false);
+	if (codec_filter->bss.count)
+		sdp_audio_codecs_intersection(r, &codec_filter->bss, false);
+	if (codec_filter->remote.audio_codecs.count)
+		sdp_audio_codecs_intersection(r, &codec_filter->remote.audio_codecs, true);
+
+#if ALLOW_REASSIGNMENT
+	/* If osmo-msc were able to trigger a re-assignment after the remote side has picked a codec mismatching the
+	 * initial Assignment, then this code here would make sense: keep the other codecs as available to choose from,
+	 * but put the currently assigned codec in the first position. */
+	if (sdp_audio_codec_is_set(a)) {
+		/* Assignment has completed, the chosen codec should be the first of the resulting SDP.
+		 * Make sure this is actually listed in the result SDP and move to first place. */
+		struct sdp_audio_codec *select = sdp_audio_codec_by_descr(r, a);
+
+		if (!select) {
+			/* Not present. Add. */
+			if (sdp_audio_codec_by_payload_type(r, a->payload_type, false)) {
+				/* Oh crunch, that payload type number is already in use.
+				 * Find an unused one. */
+				for (a->payload_type = 96; a->payload_type <= 127; a->payload_type++) {
+					if (!sdp_audio_codec_by_payload_type(r, a->payload_type, false))
+						break;
+				}
+
+				if (a->payload_type > 127)
+					return -ENOSPC;
+			}
+			select = sdp_audio_codec_add_copy(r, a);
+		}
+
+		sdp_audio_codecs_select(r, select);
+	}
+#else
+	/* Currently, osmo-msc does not trigger re-assignment if the remote side has picked a codec that the local side
+	 * would also support, but the local side has already assigned a mismatching codec before. Mismatching codecs
+	 * means call failure. So, currently, if locally, Assignment has already happened, it makes sense to send only
+	 * the assigned codec as available choice to the remote side. */
+	if (sdp_audio_codec_is_set(a)) {
+		/* Assignment has completed, the chosen codec should be the the only possible one. */
+		struct sdp_audio_codecs assigned_codec = {};
+		sdp_audio_codecs_add_copy(&assigned_codec, a);
+		sdp_audio_codecs_intersection(r, &assigned_codec, false);
+	}
+#endif
+	return 0;
+}
+
+int codec_filter_to_str_buf(char *buf, size_t buflen, const struct codec_filter *codec_filter)
+{
+	struct osmo_strbuf sb = { .buf = buf, .len = buflen };
+	OSMO_STRBUF_PRINTF(sb, "RAN={");
+	OSMO_STRBUF_APPEND(sb, sdp_audio_codecs_to_str_buf, &codec_filter->ran);
+	OSMO_STRBUF_PRINTF(sb, "}");
+
+	if (codec_filter->bss.count) {
+		OSMO_STRBUF_PRINTF(sb, " bss={");
+		OSMO_STRBUF_APPEND(sb, sdp_audio_codecs_to_str_buf, &codec_filter->bss);
+		OSMO_STRBUF_PRINTF(sb, "}");
+	}
+
+	if (codec_filter->ms.count) {
+		OSMO_STRBUF_PRINTF(sb, " MS={");
+		OSMO_STRBUF_APPEND(sb, sdp_audio_codecs_to_str_buf, &codec_filter->ms);
+		OSMO_STRBUF_PRINTF(sb, "}");
+	}
+
+	if (codec_filter->remote.audio_codecs.count
+	    || osmo_sockaddr_str_is_nonzero(&codec_filter->remote.rtp)) {
+		OSMO_STRBUF_PRINTF(sb, " remote=");
+		OSMO_STRBUF_APPEND(sb, sdp_msg_to_str_buf, &codec_filter->remote);
+	}
+
+	if (sdp_audio_codec_is_set(&codec_filter->assignment)) {
+		OSMO_STRBUF_PRINTF(sb, " assigned=");
+		OSMO_STRBUF_APPEND(sb, sdp_audio_codec_to_str_buf, &codec_filter->assignment);
+	}
+
+	OSMO_STRBUF_PRINTF(sb, " result=");
+	OSMO_STRBUF_APPEND(sb, sdp_msg_to_str_buf, &codec_filter->result);
+
+	return sb.chars_needed;
+}
+
+char *codec_filter_to_str_c(void *ctx, const struct codec_filter *codec_filter)
+{
+	OSMO_NAME_C_IMPL(ctx, 128, "codec_filter_to_str_c-ERROR", codec_filter_to_str_buf, codec_filter)
+}
+
+const char *codec_filter_to_str(const struct codec_filter *codec_filter)
+{
+	return codec_filter_to_str_c(OTC_SELECT, codec_filter);
+}