Change in osmo-msc[master]: [codecs filter] add codec_filter.h,c

neels has uploaded this change for review. ( https://gerrit.osmocom.org/c/osmo-msc/+/30116 ) 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, 279 insertions(+), 0 deletions(-) git pull ssh://gerrit.osmocom.org:29418/osmo-msc refs/changes/16/30116/1 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 &lt;info(a)sysmocom.de&gt; + * 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..e3c49c9 --- /dev/null +++ b/src/libmsc/codec_filter.c @@ -0,0 +1,211 @@ +/* Filter/overlay codec selections for a voice call, across MS, RAN and CN limitations */ +/* + * (C) 2019-2022 by sysmocom - s.m.f.c. GmbH &lt;info(a)sysmocom.de&gt; + * 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_mapping.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_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: 1 Gerrit-Owner: neels &lt;nhofmeyr(a)sysmocom.de&gt; Gerrit-MessageType: newchange