neels has submitted this change. (
https://gerrit.osmocom.org/c/osmo-msc/+/30116 )
(
4 is the latest approved patch-set.
No files were changed between the latest approved patch-set and the submitted
one.
)Change subject: [codecs filter] add codec_filter.h,c
......................................................................
[codecs filter] add codec_filter.h,c
Add the infrastructure to store and filter all codec limitiations from
the different stages: MS, BSS, CN and remote call leg. Upcoming patches
will properly collect these and find an optimal codec.
No functional change, yet.
Related: SYS#5066
Change-Id: I4d90f7ca62f2307a7b93dd164aeecbf4bd98ff0a
---
M include/osmocom/msc/Makefile.am
A include/osmocom/msc/codec_filter.h
M src/libmsc/Makefile.am
A src/libmsc/codec_filter.c
4 files changed, 299 insertions(+), 0 deletions(-)
Approvals:
laforge: Looks good to me, approved
pespin: Looks good to me, but someone else must approve
Jenkins Builder: Verified
diff --git a/include/osmocom/msc/Makefile.am b/include/osmocom/msc/Makefile.am
index 80f8972..34b4902 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_mapping.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 0000000..16315c7
--- /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 <[email protected]>
+ * 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 c7fe4bb..bde70c9 100644
--- a/src/libmsc/Makefile.am
+++ b/src/libmsc/Makefile.am
@@ -27,6 +27,7 @@
libmsc_a_SOURCES = \
call_leg.c \
cell_id_list.c \
+ codec_filter.c \
codec_mapping.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 0000000..c77aeb9
--- /dev/null
+++ b/src/libmsc/codec_filter.c
@@ -0,0 +1,215 @@
+/* Filter/overlay codec selections for a voice call, across MS, RAN and CN
limitations */
+/*
+ * (C) 2019-2023 by sysmocom - s.m.f.c. GmbH <[email protected]>
+ * All Rights Reserved
+ *
+ * Author: Neels Hofmeyr
+ *
+ * SPDX-License-Identifier: AGPL-3.0+
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Affero General Public License as published by
+ * the Free Software Foundation; either version 3 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 Affero General Public License for more details.
+ *
+ * You should have received a copy of the GNU Affero General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <osmocom/gsm/protocol/gsm_08_08.h>
+
+#include <osmocom/msc/codec_filter.h>
+#include <osmocom/msc/codec_mapping.h>
+#include <osmocom/msc/debug.h>
+
+/* Add all known payload types encountered in GSM networks */
+static void sdp_add_all_geran_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_geran_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 0
+ /* Future: 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. So
far we only offer the single assigned
+ * codec, because we have no way to deal with the remote side picking a
different codec.
+ * Another approach would be to postpone assignment until we know the
codecs from the remote side. */
+ 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_codecs_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_codecs_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 is different
+ * from the already assigned codec.
+ * So, if locally, Assignment has already chosen a codec, this is the
single definitive result to be used
+ * towards the CN. */
+ if (sdp_audio_codec_is_set(a)) {
+ /* Assignment has completed, the chosen codec should be the the
only possible one. */
+ *r = (struct sdp_audio_codecs){};
+ sdp_audio_codecs_add_copy(r, a);
+ }
+#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_APPEND(sb, sdp_msg_to_str_buf, &codec_filter->result);
+ OSMO_STRBUF_PRINTF(sb, " (from:");
+
+ 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);
+ }
+
+ 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 (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->bss.count) {
+ OSMO_STRBUF_PRINTF(sb, " bss={");
+ OSMO_STRBUF_APPEND(sb, sdp_audio_codecs_to_str_buf,
&codec_filter->bss);
+ OSMO_STRBUF_PRINTF(sb, "}");
+ }
+
+ OSMO_STRBUF_PRINTF(sb, " RAN={");
+ OSMO_STRBUF_APPEND(sb, sdp_audio_codecs_to_str_buf, &codec_filter->ran);
+ OSMO_STRBUF_PRINTF(sb, "}");
+
+ OSMO_STRBUF_PRINTF(sb, ")");
+
+ 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);
+}
--
To view, visit https://gerrit.osmocom.org/c/osmo-msc/+/30116
To unsubscribe, or for help writing mail filters, visit
https://gerrit.osmocom.org/settings
Gerrit-Project: osmo-msc
Gerrit-Branch: master
Gerrit-Change-Id: I4d90f7ca62f2307a7b93dd164aeecbf4bd98ff0a
Gerrit-Change-Number: 30116
Gerrit-PatchSet: 8
Gerrit-Owner: neels <[email protected]>
Gerrit-Reviewer: Jenkins Builder
Gerrit-Reviewer: laforge <[email protected]>
Gerrit-Reviewer: neels <[email protected]>
Gerrit-Reviewer: pespin <[email protected]>
Gerrit-MessageType: merged
