diff options
author | Neels Hofmeyr <neels@hofmeyr.de> | 2019-10-21 03:24:11 +0200 |
---|---|---|
committer | Neels Hofmeyr <neels@hofmeyr.de> | 2020-01-06 18:00:40 +0100 |
commit | f31a1ccd9a3eb474936f5b946287581514b29436 (patch) | |
tree | ec59ae18c6e54b3530bf2f56e09d99bc23390b0d /src/libmsc | |
parent | 02dd265d68b771bf315cfe6620c9b2371edea828 (diff) |
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
Diffstat (limited to 'src/libmsc')
-rw-r--r-- | src/libmsc/Makefile.am | 2 | ||||
-rw-r--r-- | src/libmsc/call_leg.c | 17 | ||||
-rw-r--r-- | src/libmsc/cc_sdp.c | 179 | ||||
-rw-r--r-- | src/libmsc/codec_sdp_cc_t9n.c | 424 | ||||
-rw-r--r-- | src/libmsc/gsm_04_08_cc.c | 408 | ||||
-rw-r--r-- | src/libmsc/mncc_call.c | 37 | ||||
-rw-r--r-- | src/libmsc/msc_a.c | 196 | ||||
-rw-r--r-- | src/libmsc/msc_ho.c | 18 | ||||
-rw-r--r-- | src/libmsc/msc_t.c | 7 | ||||
-rw-r--r-- | src/libmsc/rtp_stream.c | 111 | ||||
-rw-r--r-- | src/libmsc/sdp_msg.c | 82 |
11 files changed, 1283 insertions, 198 deletions
diff --git a/src/libmsc/Makefile.am b/src/libmsc/Makefile.am index e6a2dc164..b226ea465 100644 --- a/src/libmsc/Makefile.am +++ b/src/libmsc/Makefile.am @@ -29,7 +29,9 @@ noinst_LIBRARIES = \ libmsc_a_SOURCES = \ call_leg.c \ + cc_sdp.c \ cell_id_list.c \ + codec_sdp_cc_t9n.c \ sccp_ran.c \ msc_vty.c \ db.c \ diff --git a/src/libmsc/call_leg.c b/src/libmsc/call_leg.c index b1d0b1e48..725b8bbdb 100644 --- a/src/libmsc/call_leg.c +++ b/src/libmsc/call_leg.c @@ -316,7 +316,8 @@ struct osmo_sockaddr_str *call_leg_local_ip(struct call_leg *cl, enum rtp_direct * MDCX. */ 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) { if (call_leg_ensure_rtp_alloc(cl, dir, call_id, for_trans)) return -EIO; @@ -325,8 +326,8 @@ int call_leg_ensure_ci(struct call_leg *cl, enum rtp_direction dir, uint32_t cal cl->rtp[dir]->use_osmux = true; cl->rtp[dir]->remote_osmux_cid = -1; /* wildcard */ } - if (codec_if_known) - rtp_stream_set_codec(cl->rtp[dir], *codec_if_known); + if (codecs_if_known) + rtp_stream_set_codecs(cl->rtp[dir], codecs_if_known); if (remote_addr_if_known && osmo_sockaddr_str_is_nonzero(remote_addr_if_known)) rtp_stream_set_remote_addr(cl->rtp[dir], remote_addr_if_known); return rtp_stream_ensure_ci(cl->rtp[dir], cl->mgw_endpoint); @@ -335,22 +336,22 @@ int call_leg_ensure_ci(struct call_leg *cl, enum rtp_direction dir, uint32_t cal int call_leg_local_bridge(struct call_leg *cl1, uint32_t call_id1, struct gsm_trans *trans1, struct call_leg *cl2, uint32_t call_id2, struct gsm_trans *trans2) { - enum mgcp_codecs codec; + struct sdp_audio_codecs *codecs; cl1->local_bridge = cl2; cl2->local_bridge = cl1; /* We may just copy the codec info we have for the RAN side of the first leg to the CN side of both legs. This * also means that if both legs use different codecs the MGW must perform transcoding on the second leg. */ - if (!cl1->rtp[RTP_TO_RAN] || !cl1->rtp[RTP_TO_RAN]->codec_known) { + if (!cl1->rtp[RTP_TO_RAN] || !cl1->rtp[RTP_TO_RAN]->codecs_known) { LOG_CALL_LEG(cl1, LOGL_ERROR, "RAN-side RTP stream codec is not known, not ready for bridging\n"); return -EINVAL; } - codec = cl1->rtp[RTP_TO_RAN]->codec; + codecs = &cl1->rtp[RTP_TO_RAN]->codecs; call_leg_ensure_ci(cl1, RTP_TO_CN, call_id1, trans1, - &codec, &cl2->rtp[RTP_TO_CN]->local); + codecs, &cl2->rtp[RTP_TO_CN]->local); call_leg_ensure_ci(cl2, RTP_TO_CN, call_id2, trans2, - &codec, &cl1->rtp[RTP_TO_CN]->local); + codecs, &cl1->rtp[RTP_TO_CN]->local); return 0; } diff --git a/src/libmsc/cc_sdp.c b/src/libmsc/cc_sdp.c new file mode 100644 index 000000000..eeb9ab640 --- /dev/null +++ b/src/libmsc/cc_sdp.c @@ -0,0 +1,179 @@ +#include <osmocom/gsm/protocol/gsm_08_08.h> + +#include <osmocom/msc/cc_sdp.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]); +} + +static void cc_sdp_set_ran(struct cc_sdp *cc_sdp, enum osmo_rat_type ran_type) +{ + cc_sdp->ran = (struct sdp_audio_codecs){}; + + switch (ran_type) { + default: + case OSMO_RAT_GERAN_A: + sdp_add_all_mobile_codecs(&cc_sdp->ran); + break; + + case OSMO_RAT_UTRAN_IU: + sdp_add_all_utran_codecs(&cc_sdp->ran); + break; + } +} + +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) +{ + *cc_sdp = (struct cc_sdp){}; + cc_sdp_set_ran(cc_sdp, ran_type); + + if (ms_bearer_cap) + sdp_audio_codecs_from_bearer_cap(&cc_sdp->ms, ms_bearer_cap); + + if (codec_list_bss_supported) + cc_sdp_set_cell(cc_sdp, codec_list_bss_supported); +} + +void cc_sdp_set_cell(struct cc_sdp *cc_sdp, + const struct gsm0808_speech_codec_list *codec_list_bss_supported) +{ + cc_sdp->cell = (struct sdp_audio_codecs){}; + if (codec_list_bss_supported) + sdp_audio_codecs_from_speech_codec_list(&cc_sdp->cell, codec_list_bss_supported); +} + +/* 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 cc_sdp->result. (The result.rtp address remains unchanged.) */ +int cc_sdp_filter(struct cc_sdp *cc_sdp) +{ + struct sdp_audio_codecs *r = &cc_sdp->result.audio_codecs; + struct sdp_audio_codec *a = &cc_sdp->assignment; + *r = cc_sdp->ran; + if (cc_sdp->ms.count) + sdp_audio_codecs_intersection(r, &cc_sdp->ms, false); + if (cc_sdp->cell.count) + sdp_audio_codecs_intersection(r, &cc_sdp->cell, false); + if (cc_sdp->remote.audio_codecs.count) + sdp_audio_codecs_intersection(r, &cc_sdp->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 (a->subtype_name[0]) { + /* 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 (a->subtype_name[0]) { + /* Assignment has completed, the chosen codec should be the the only possible one. */ + struct sdp_audio_codecs assigned_codec = {}; + sdp_audio_codec_add_copy(&assigned_codec, a); + sdp_audio_codecs_intersection(r, &assigned_codec, false); + } +#endif + return 0; +} + +int cc_sdp_name_buf(char *buf, size_t buflen, const struct cc_sdp *cc_sdp) +{ + struct osmo_strbuf sb = { .buf = buf, .len = buflen }; + OSMO_STRBUF_PRINTF(sb, "RAN={"); + OSMO_STRBUF_APPEND(sb, sdp_audio_codecs_name_buf, &cc_sdp->ran); + OSMO_STRBUF_PRINTF(sb, "}"); + + if (cc_sdp->cell.count) { + OSMO_STRBUF_PRINTF(sb, " cell={"); + OSMO_STRBUF_APPEND(sb, sdp_audio_codecs_name_buf, &cc_sdp->cell); + OSMO_STRBUF_PRINTF(sb, "}"); + } + + if (cc_sdp->ms.count) { + OSMO_STRBUF_PRINTF(sb, " MS={"); + OSMO_STRBUF_APPEND(sb, sdp_audio_codecs_name_buf, &cc_sdp->ms); + OSMO_STRBUF_PRINTF(sb, "}"); + } + + if (cc_sdp->remote.audio_codecs.count + || osmo_sockaddr_str_is_nonzero(&cc_sdp->remote.rtp)) { + OSMO_STRBUF_PRINTF(sb, " remote="); + OSMO_STRBUF_APPEND(sb, sdp_msg_name_buf, &cc_sdp->remote); + } + + if (cc_sdp->assignment.subtype_name[0]) { + OSMO_STRBUF_PRINTF(sb, " assigned="); + OSMO_STRBUF_APPEND(sb, sdp_audio_codec_name_buf, &cc_sdp->assignment); + } + + OSMO_STRBUF_PRINTF(sb, " result="); + OSMO_STRBUF_APPEND(sb, sdp_msg_name_buf, &cc_sdp->result); + + return sb.chars_needed; +} + +char *cc_sdp_name_c(void *ctx, const struct cc_sdp *cc_sdp) +{ + OSMO_NAME_C_IMPL(ctx, 128, "cc_sdp_name_c-ERROR", cc_sdp_name_buf, cc_sdp) +} + +const char *cc_sdp_name(const struct cc_sdp *cc_sdp) +{ + return cc_sdp_name_c(OTC_SELECT, cc_sdp); +} diff --git a/src/libmsc/codec_sdp_cc_t9n.c b/src/libmsc/codec_sdp_cc_t9n.c new file mode 100644 index 000000000..75b91abfb --- /dev/null +++ b/src/libmsc/codec_sdp_cc_t9n.c @@ -0,0 +1,424 @@ +#include <string.h> + +#include <osmocom/gsm/mncc.h> + +#include <osmocom/msc/sdp_msg.h> +#include <osmocom/msc/codec_sdp_cc_t9n.h> +#include <osmocom/msc/mncc.h> + +const struct codec_mapping codec_map[] = { + /* FIXME: I'm not sure about OFR, OHR -- O means octet-aligned?? */ + { + .sdp = { + .payload_type = 0, + .subtype_name = "PCMU", + .rate = 8000, + }, + .mgcp = CODEC_PCMU_8000_1, + }, + { + .sdp = { + .payload_type = 3, + .subtype_name = "GSM", + .rate = 8000, + }, + .mgcp = CODEC_GSM_8000_1, + .speech_ver_count = 1, + .speech_ver = { GSM48_BCAP_SV_FR }, + .mncc_payload_msg_type = GSM_TCHF_FRAME, + .has_gsm0808_speech_codec_type = true, + .gsm0808_speech_codec_type = GSM0808_SCT_FR1, + .perm_speech = GSM0808_PERM_FR1, + .frhr = CODEC_FRHR_FR, + }, + { + .sdp = { + .payload_type = 8, + .subtype_name = "PCMA", + .rate = 8000, + }, + .mgcp = CODEC_PCMA_8000_1, + }, + { + .sdp = { + .payload_type = 18, + .subtype_name = "G729", + .rate = 8000, + }, + .mgcp = CODEC_G729_8000_1, + }, + { + .sdp = { + .payload_type = 110, + .subtype_name = "GSM-EFR", + .rate = 8000, + }, + .mgcp = CODEC_GSMEFR_8000_1, + .speech_ver_count = 1, + .speech_ver = { GSM48_BCAP_SV_EFR }, + .mncc_payload_msg_type = GSM_TCHF_FRAME_EFR, + .has_gsm0808_speech_codec_type = true, + .gsm0808_speech_codec_type = GSM0808_SCT_FR2, + .perm_speech = GSM0808_PERM_FR2, + .frhr = CODEC_FRHR_FR, + }, + { + .sdp = { + .payload_type = 111, + .subtype_name = "GSM-HR-08", + .rate = 8000, + }, + .mgcp = CODEC_GSMHR_8000_1, + .speech_ver_count = 1, + .speech_ver = { GSM48_BCAP_SV_HR }, + .mncc_payload_msg_type = GSM_TCHH_FRAME, + .has_gsm0808_speech_codec_type = true, + .gsm0808_speech_codec_type = GSM0808_SCT_HR1, + .perm_speech = GSM0808_PERM_HR1, + .frhr = CODEC_FRHR_HR, + }, + { + .sdp = { + .payload_type = 112, + .subtype_name = "AMR", + .rate = 8000, + /* It is important to send this fmtp parameter to a SIP peer in SDP, + * otherwise the voice audio is broken noise. + * However, a SIP peer may offer AMR without this parameter set in its SDP, so fmtp must be + * ignored during codec matching: otherwise an incoming AMR codec without this parameter fails + * to match this entry, and it ends in an aborted call due to no codec match. + * If the peer offers plain "AMR/8000" and we reply with "AMR/8000 fmtp:octet-align=1", + * then everything works out happily, */ + .fmtp = "octet-align=1", + }, + .mgcp = CODEC_AMR_8000_1, + .speech_ver_count = 1, + .speech_ver = { GSM48_BCAP_SV_AMR_F }, + .mncc_payload_msg_type = GSM_TCH_FRAME_AMR, + .has_gsm0808_speech_codec_type = true, + .gsm0808_speech_codec_type = GSM0808_SCT_FR3, + .perm_speech = GSM0808_PERM_FR3, + .frhr = CODEC_FRHR_FR, + }, + { + .sdp = { + .payload_type = 112, + .subtype_name = "AMR", + .rate = 8000, + .fmtp = "octet-align=1;mode-set=0,1,2,3", + }, + .mgcp = CODEC_AMR_8000_1, + .speech_ver_count = 2, + .speech_ver = { GSM48_BCAP_SV_AMR_H, GSM48_BCAP_SV_AMR_OH }, + .mncc_payload_msg_type = GSM_TCH_FRAME_AMR, + .has_gsm0808_speech_codec_type = true, + .gsm0808_speech_codec_type = GSM0808_SCT_HR3, + .perm_speech = GSM0808_PERM_HR3, + .frhr = CODEC_FRHR_HR, + }, + { + .sdp = { + .payload_type = 113, + .subtype_name = "AMR-WB", + .rate = 16000, + .fmtp = "octet-align=1", + }, + .mgcp = CODEC_AMRWB_16000_1, + .speech_ver_count = 2, + .speech_ver = { GSM48_BCAP_SV_AMR_OFW, GSM48_BCAP_SV_AMR_FW }, + .mncc_payload_msg_type = GSM_TCH_FRAME_AMR, + .has_gsm0808_speech_codec_type = true, + .gsm0808_speech_codec_type = GSM0808_SCT_FR5, + .perm_speech = GSM0808_PERM_FR5, + .frhr = CODEC_FRHR_FR, + }, + { + .sdp = { + .payload_type = 113, + .subtype_name = "AMR-WB", + .rate = 16000, + .fmtp = "octet-align=1;mode-set=0,1,2,3", /* TODO: does this make sense?? */ + }, + .mgcp = CODEC_AMRWB_16000_1, + .speech_ver_count = 1, + .speech_ver = { GSM48_BCAP_SV_AMR_OHW }, + .mncc_payload_msg_type = GSM_TCH_FRAME_AMR, + .has_gsm0808_speech_codec_type = true, + .gsm0808_speech_codec_type = GSM0808_SCT_HR4, + .perm_speech = GSM0808_PERM_HR4, + .frhr = CODEC_FRHR_HR, + }, +}; + +const struct gsm_mncc_bearer_cap bearer_cap_empty = { + .speech_ver = { -1 }, + }; + +const struct codec_mapping *codec_mapping_by_speech_ver(enum gsm48_bcap_speech_ver speech_ver) +{ + const struct codec_mapping *m; + foreach_codec_mapping(m) { + int i; + for (i = 0; i < m->speech_ver_count; i++) + if (m->speech_ver[i] == speech_ver) + return m; + } + return NULL; +} + + +const struct codec_mapping *codec_mapping_by_gsm0808_speech_codec_type(enum gsm0808_speech_codec_type sct, uint16_t cfg) +{ + const struct codec_mapping *m; + foreach_codec_mapping(m) { + if (!m->has_gsm0808_speech_codec_type) + continue; + if (m->gsm0808_speech_codec_type == sct) + return m; + /* TODO: evaluate cfg bits? */ + } + return NULL; +} + +const struct codec_mapping *codec_mapping_by_perm_speech(enum gsm0808_permitted_speech perm_speech) +{ + const struct codec_mapping *m; + foreach_codec_mapping(m) { + if (m->perm_speech == perm_speech) + return m; + } + return NULL; +} + +const struct codec_mapping *codec_mapping_by_subtype_name(const char *subtype_name) +{ + const struct codec_mapping *m; + foreach_codec_mapping(m) { + if (!strcmp(m->sdp.subtype_name, subtype_name)) + return m; + } + return NULL; +} + +const struct codec_mapping *codec_mapping_by_mgcp_codec(enum mgcp_codecs mgcp) +{ + const struct codec_mapping *m; + foreach_codec_mapping(m) { + if (m->mgcp == mgcp) + return m; + } + return NULL; +} + +/* Append given Speech Version to the end of the Bearer Capabilities Speech Version array. Return 1 if added, zero + * otherwise (as in, return the number of items added). */ +int bearer_cap_add_speech_ver(struct gsm_mncc_bearer_cap *bearer_cap, enum gsm48_bcap_speech_ver speech_ver) +{ + int i; + for (i = 0; i < ARRAY_SIZE(bearer_cap->speech_ver) - 1; i++) { + if (bearer_cap->speech_ver[i] == speech_ver) + return 0; + if (bearer_cap->speech_ver[i] == -1) { + bearer_cap->speech_ver[i] = speech_ver; + bearer_cap->speech_ver[i+1] = -1; + return 1; + } + } + return 0; +} + +/* From the current speech_ver list present in the bearer_cap, set the bearer_cap.radio. + * If a HR speech_ver is present, set to GSM48_BCAP_RRQ_DUAL_FR, otherwise set to GSM48_BCAP_RRQ_FR_ONLY. */ +int bearer_cap_set_radio(struct gsm_mncc_bearer_cap *bearer_cap) +{ + bool hr_present; + int i; + for (i = 0; i < ARRAY_SIZE(bearer_cap->speech_ver) - 1; i++) { + const struct codec_mapping *m = codec_mapping_by_speech_ver(bearer_cap->speech_ver[i]); + + if (!m) + continue; + + if (m->frhr == CODEC_FRHR_HR) + hr_present = true; + } + + if (hr_present) + bearer_cap->radio = GSM48_BCAP_RRQ_DUAL_FR; + else + bearer_cap->radio = GSM48_BCAP_RRQ_FR_ONLY; + + return 0; +} + +/* Try to convert the SDP audio codec name to Speech Versions to append to Bearer Capabilities. + * Return the number of Speech Version entries added (some may add more than one, others may be unknown/unapplicable and + * return 0). */ +int sdp_audio_codec_add_to_bearer_cap(struct gsm_mncc_bearer_cap *bearer_cap, const struct sdp_audio_codec *codec) +{ + const struct codec_mapping *m; + int added = 0; + foreach_codec_mapping(m) { + int i; + if (strcmp(m->sdp.subtype_name, codec->subtype_name)) + continue; + /* TODO also match rate and fmtp? */ + for (i = 0; i < m->speech_ver_count; i++) { + added += bearer_cap_add_speech_ver(bearer_cap, m->speech_ver[i]); + } + } + return added; +} + +/* Append all audio codecs found in given sdp_msg to Bearer Capability, by traversing all codec entries with + * sdp_audio_codec_add_to_bearer_cap(). Return the number of Speech Version entries added. + * Note that Speech Version entries are only appended, no previous entries are removed. + * Note that only the Speech Version entries are modified; to make a valid Bearer Capabiliy, at least bearer_cap->radio + * must also be set (before or after this function); see also bearer_cap_set_radio(). */ +int sdp_audio_codecs_to_bearer_cap(struct gsm_mncc_bearer_cap *bearer_cap, const struct sdp_audio_codecs *ac) +{ + const struct sdp_audio_codec *codec; + int added = 0; + + foreach_sdp_audio_codec(codec, ac) { + added += sdp_audio_codec_add_to_bearer_cap(bearer_cap, codec); + } + + return added; +} + +/* Convert Speech Version to SDP audio codec and append to SDP message struct. */ +struct sdp_audio_codec *sdp_audio_codecs_add_speech_ver(struct sdp_audio_codecs *ac, + enum gsm48_bcap_speech_ver speech_ver) +{ + const struct codec_mapping *m; + struct sdp_audio_codec *ret = NULL; + foreach_codec_mapping(m) { + int i; + for (i = 0; i < m->speech_ver_count; i++) { + if (m->speech_ver[i] == speech_ver) { + ret = sdp_audio_codec_add_copy(ac, &m->sdp); + break; + } + } + } + return ret; +} + +struct sdp_audio_codec *sdp_audio_codecs_add_mgcp_codec(struct sdp_audio_codecs *ac, enum mgcp_codecs mgcp_codec) +{ + const struct codec_mapping *m = codec_mapping_by_mgcp_codec(mgcp_codec); + if (!m) + return NULL; + return sdp_audio_codec_add_copy(ac, &m->sdp); +} + +void sdp_audio_codecs_from_bearer_cap(struct sdp_audio_codecs *ac, const struct gsm_mncc_bearer_cap *bc) +{ + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(bc->speech_ver); i++) { + if (bc->speech_ver[i] == -1) + break; + sdp_audio_codecs_add_speech_ver(ac, bc->speech_ver[i]); + } +} + +void sdp_audio_codecs_from_speech_codec_list(struct sdp_audio_codecs *ac, const struct gsm0808_speech_codec_list *cl) +{ + int i; + for (i = 0; i < cl->len; i++) { + const struct gsm0808_speech_codec *sc = &cl->codec[i]; + const struct codec_mapping *m = codec_mapping_by_gsm0808_speech_codec_type(sc->type, sc->cfg); + if (!m) + continue; + sdp_audio_codec_add_copy(ac, &m->sdp); + } +} + +int sdp_audio_codecs_to_gsm0808_channel_type(struct gsm0808_channel_type *ct, const struct sdp_audio_codecs *ac) +{ + const struct sdp_audio_codec *codec; + bool fr_present = false; + int first_fr_idx = -1; + bool hr_present = false; + int first_hr_idx = -1; + int idx = -1; + + *ct = (struct gsm0808_channel_type){ + .ch_indctr = GSM0808_CHAN_SPEECH, + }; + + foreach_sdp_audio_codec(codec, ac) { + const struct codec_mapping *m; + int i; + bool dup; + idx++; + foreach_codec_mapping(m) { + if (strcmp(m->sdp.subtype_name, codec->subtype_name)) + continue; + + switch (m->perm_speech) { + default: + continue; + + case GSM0808_PERM_FR1: + case GSM0808_PERM_FR2: + case GSM0808_PERM_FR3: + case GSM0808_PERM_FR4: + case GSM0808_PERM_FR5: + fr_present = true; + if (first_fr_idx < 0) + first_fr_idx = idx; + break; + + case GSM0808_PERM_HR1: + case GSM0808_PERM_HR2: + case GSM0808_PERM_HR3: + case GSM0808_PERM_HR4: + case GSM0808_PERM_HR6: + hr_present = true; + if (first_hr_idx < 0) + first_hr_idx = idx; + break; + } + + /* Avoid duplicates */ + dup = false; + for (i = 0; i < ct->perm_spch_len; i++) { + if (ct->perm_spch[i] == m->perm_speech) { + dup = true; + break; + } + } + if (dup) + continue; + + ct->perm_spch[ct->perm_spch_len] = m->perm_speech; + ct->perm_spch_len++; + } + } + + if (fr_present && hr_present) { + if (first_fr_idx <= first_hr_idx) + ct->ch_rate_type = GSM0808_SPEECH_FULL_PREF; + else + ct->ch_rate_type = GSM0808_SPEECH_HALF_PREF; + } else if (fr_present && !hr_present) + ct->ch_rate_type = GSM0808_SPEECH_FULL_BM; + else if (!fr_present && hr_present) + ct->ch_rate_type = GSM0808_SPEECH_HALF_LM; + else + return -EINVAL; + return 0; +} + +enum mgcp_codecs sdp_audio_codec_to_mgcp_codec(const struct sdp_audio_codec *codec) +{ + const struct codec_mapping *m; + foreach_codec_mapping(m) { + if (!sdp_audio_codec_cmp(&m->sdp, codec, false, false)) + return m->mgcp; + } + return NO_MGCP_CODEC; +} diff --git a/src/libmsc/gsm_04_08_cc.c b/src/libmsc/gsm_04_08_cc.c index 8cfb6117d..c92e3676b 100644 --- a/src/libmsc/gsm_04_08_cc.c +++ b/src/libmsc/gsm_04_08_cc.c @@ -55,6 +55,9 @@ #include <osmocom/msc/rtp_stream.h> #include <osmocom/msc/mncc_call.h> #include <osmocom/msc/msc_t.h> +#include <osmocom/msc/sdp_msg.h> +#include <osmocom/msc/cc_sdp.h> +#include <osmocom/msc/codec_sdp_cc_t9n.h> #include <osmocom/gsm/gsm48.h> #include <osmocom/gsm/gsm0480.h> @@ -254,8 +257,12 @@ static int mncc_recvmsg(struct gsm_network *net, struct gsm_trans *trans, int mncc_release_ind(struct gsm_network *net, struct gsm_trans *trans, uint32_t callref, int location, int value) { + /* BEWARE: trans may be passed as NULL to reply to invalid MNCC requests */ struct gsm_mncc rel; + if (trans && trans->cc.mncc_release_sent) + return 0; + memset(&rel, 0, sizeof(rel)); rel.callref = callref; mncc_set_cause(&rel, location, value); @@ -498,6 +505,8 @@ static int gsm48_cc_rx_setup(struct gsm_trans *trans, struct msgb *msg) memset(&setup, 0, sizeof(struct gsm_mncc)); setup.callref = trans->callref; + OSMO_ASSERT(trans->msc_a); + tlv_parse(&tp, &gsm48_att_tlvdef, gh->data, payload_len, 0, 0); /* emergency setup is identified by msg_type */ if (msg_type == GSM48_MT_CC_EMERG_SETUP) { @@ -567,25 +576,77 @@ static int gsm48_cc_rx_setup(struct gsm_trans *trans, struct msgb *msg) TLVP_VAL(&tp, GSM48_IE_CC_CAP)-1); } - new_cc_state(trans, GSM_CSTATE_INITIATED); + cc_sdp_init(&trans->cc.sdp, trans->msc_a->c.ran->type, + setup.fields & MNCC_F_BEARER_CAP ? &trans->bearer_cap : NULL, + &trans->msc_a->cc.codec_list_bss_supported); + cc_sdp_filter(&trans->cc.sdp); LOG_TRANS(trans, setup.emergency ? LOGL_NOTICE : LOGL_INFO, "%sSETUP to %s\n", setup.emergency ? "EMERGENCY_" : "", setup.called.number); + LOG_TRANS(trans, LOGL_DEBUG, "codecs: %s\n", cc_sdp_name(&trans->cc.sdp)); rate_ctr_inc(&trans->net->msc_ctrs->ctr[MSC_CTR_CALL_MO_SETUP]); + new_cc_state(trans, GSM_CSTATE_INITIATED); + + /* To complete the MNCC_SETUP_IND, we need to provide an RTP address and port. First instruct the MGW to create + * a CN-side RTP conn, and continue with MNCC_SETUP_IND once that is done. Leave trans.cc in GSM_CSTATE_NULL and + * note down the msg_type to indicate that we indeed composed an MNCC_SETUP_IND for later. */ + setup.msg_type = MNCC_SETUP_IND; + trans->cc.msg = setup; + return msc_a_try_call_assignment(trans); + /* continue in gsm48_cc_rx_setup_cn_local_rtp_port_known() */ +} + +/* Callback for MNCC_SETUP_IND waiting for the core network RTP port to be established by the MGW (via msc_a) */ +void gsm48_cc_rx_setup_cn_local_rtp_port_known(struct gsm_trans *trans) +{ + struct msc_a *msc_a = trans->msc_a; + struct gsm_mncc setup = trans->cc.msg; + struct osmo_sockaddr_str *rtp_cn_local; + struct sdp_msg *sdp; + + if (trans->cc.state != GSM_CSTATE_INITIATED + || setup.msg_type != MNCC_SETUP_IND) { + LOG_TRANS(trans, LOGL_ERROR, + "Unexpected CC state. Expected GSM_CSTATE_NULL and a buffered MNCC_SETUP_IND message," + " found CC state %d and msg_type %s\n", + trans->cc.state, get_mncc_name(setup.msg_type)); + trans->callref = 0; + trans_free(trans); + return; + } + + if (!msc_a) { + LOG_TRANS(trans, LOGL_ERROR, "No connection for CC trans\n"); + trans->callref = 0; + trans_free(trans); + return; + } + + /* 'setup' above has taken the value of trans->cc.msg, we can now clear that. */ + trans->cc.msg = (struct gsm_mncc){}; + + /* Insert the CN side RTP port now available into SDP and compose SDP string */ + rtp_cn_local = call_leg_local_ip(msc_a->cc.call_leg, RTP_TO_CN); + if (!osmo_sockaddr_str_is_nonzero(rtp_cn_local)) { + LOG_TRANS(trans, LOGL_ERROR, "Cannot compose SDP for MNCC_SETUP_IND: no RTP set up for the CN side\n"); + trans_free(trans); + return; + } + + cc_sdp_filter(&trans->cc.sdp); + sdp = &trans->cc.sdp.result; + sdp->rtp = *rtp_cn_local; + sdp_msg_to_str(setup.sdp, sizeof(setup.sdp), sdp); + /* indicate setup to MNCC */ mncc_recvmsg(trans->net, trans, MNCC_SETUP_IND, &setup); - - /* MNCC code will modify the channel asynchronously, we should - * ipaccess-bind only after the modification has been made to the - * lchan->tch_mode */ - return 0; } static int gsm48_cc_tx_setup(struct gsm_trans *trans, void *arg) { - struct msgb *msg = gsm48_msgb_alloc_name("GSM 04.08 CC STUP"); + struct msgb *msg = gsm48_msgb_alloc_name("GSM 04.08 CC SETUP"); struct gsm48_hdr *gh; struct gsm_mncc *setup = arg; int rc, trans_id; @@ -622,15 +683,65 @@ static int gsm48_cc_tx_setup(struct gsm_trans *trans, void *arg) gh->msg_type = GSM48_MT_CC_SETUP; - gsm48_start_cc_timer(trans, 0x303, GSM48_T303); + /* We must not pass bearer_cap to cc_sdp_init(), because we haven't received the MS's Bearer Capabilities yet; + * the Bearer Capabilities handled here are actually the remote call leg's Bearer Capabilities to be passed on + * during the CC Setup. */ + cc_sdp_init(&trans->cc.sdp, trans->msc_a->c.ran->type, NULL, + &trans->msc_a->cc.codec_list_bss_supported); + + /* sdp.remote: if SDP is included in the MNCC, take that as definitive list of remote audio codecs. */ + if (setup->sdp[0]) { + rc = sdp_msg_from_str(&trans->cc.sdp.remote, setup->sdp); + if (rc) + LOG_TRANS(trans, LOGL_ERROR, "Failed to parse remote call leg SDP: %d\n", rc); + } + + /* sdp.remote: if there is no SDP information or we failed to parse it, try using the Bearer Capability from + * MNCC, if any. */ + if (!trans->cc.sdp.remote.audio_codecs.count && (setup->fields & MNCC_F_BEARER_CAP)) { + trans->cc.sdp.remote = (struct sdp_msg){}; + sdp_audio_codecs_from_bearer_cap(&trans->cc.sdp.remote.audio_codecs, + &setup->bearer_cap); + } + + if (!trans->cc.sdp.remote.audio_codecs.count) + LOG_TRANS(trans, LOGL_ERROR, + "Got no information of remote audio codecs: neither SDP nor Bearer Capability. Trying anyway.\n"); + + /* Translate SDP to bearer capability Speech Version entries. + * If we supported transcoding, this could add arbitrary speech versions. + * For now add speech_ver entries for each codec in the SDP that matches a GSM speech_ver constant. */ + cc_sdp_filter(&trans->cc.sdp); + LOG_TRANS(trans, LOGL_DEBUG, "codecs: %s\n", cc_sdp_name(&trans->cc.sdp)); + trans->bearer_cap = (struct gsm_mncc_bearer_cap){ + .speech_ver = { -1 }, + }; + sdp_audio_codecs_to_bearer_cap(&trans->bearer_cap, &trans->cc.sdp.result.audio_codecs); + rc = bearer_cap_set_radio(&trans->bearer_cap); + if (rc) { + LOG_TRANS(trans, LOGL_ERROR, "Error composing Bearer Capability for CC Setup\n"); + trans_free(trans); + msgb_free(msg); + return rc; + } - /* bearer capability */ - if (setup->fields & MNCC_F_BEARER_CAP) { - /* Create a copy of the bearer capability in the transaction struct, so we - * can use this information later */ - memcpy(&trans->bearer_cap, &setup->bearer_cap, sizeof(trans->bearer_cap)); - gsm48_encode_bearer_cap(msg, 0, &setup->bearer_cap); + /* If no resulting codecs remain, error out. If the MGW were able to transcode, we would just use unidentical + * codecs on each conn of the MGW endpoint. */ + if (trans->bearer_cap.speech_ver[0] == -1) { + LOG_TRANS(trans, LOGL_ERROR, "%s: no codec match possible: %s\n", + get_mncc_name(setup->msg_type), cc_sdp_name(&trans->cc.sdp)); + + /* incompatible codecs */ + rc = mncc_release_ind(trans->net, trans, trans->callref, + GSM48_CAUSE_LOC_PRN_S_LU, + GSM48_CC_CAUSE_INCOMPAT_DEST /* TODO: correct cause code? */); + trans->cc.mncc_release_sent = true; + trans_free(trans); + msgb_free(msg); + return rc; } + gsm48_encode_bearer_cap(msg, 0, &trans->bearer_cap); + /* facility */ if (setup->fields & MNCC_F_FACILITY) gsm48_encode_facility(msg, 0, &setup->facility); @@ -657,6 +768,8 @@ static int gsm48_cc_tx_setup(struct gsm_trans *trans, void *arg) rate_ctr_inc(&trans->net->msc_ctrs->ctr[MSC_CTR_CALL_MT_SETUP]); + gsm48_start_cc_timer(trans, 0x303, GSM48_T303); + return trans_tx_gsm48(trans, msg); } @@ -691,9 +804,14 @@ static int gsm48_cc_rx_call_conf(struct gsm_trans *trans, struct msgb *msg) /* Create a copy of the bearer capability * in the transaction struct, so we can use * this information later */ - memcpy(&trans->bearer_cap,&call_conf.bearer_cap, + memcpy(&trans->bearer_cap, &call_conf.bearer_cap, sizeof(trans->bearer_cap)); + + /* Note MS codec capabilities for codec negotiation */ + trans->cc.sdp.ms = (struct sdp_audio_codecs){}; + sdp_audio_codecs_from_bearer_cap(&trans->cc.sdp.ms, &call_conf.bearer_cap); } + /* cause */ if (TLVP_PRESENT(&tp, GSM48_IE_CAUSE)) { call_conf.fields |= MNCC_F_CAUSE; @@ -710,8 +828,6 @@ static int gsm48_cc_rx_call_conf(struct gsm_trans *trans, struct msgb *msg) /* IMSI of called subscriber */ OSMO_STRLCPY_ARRAY(call_conf.imsi, trans->vsub->imsi); - new_cc_state(trans, GSM_CSTATE_MO_TERM_CALL_CONF); - /* Assign call (if not done yet) */ rc = msc_a_try_call_assignment(trans); @@ -720,8 +836,53 @@ static int gsm48_cc_rx_call_conf(struct gsm_trans *trans, struct msgb *msg) if (rc) return rc; - return mncc_recvmsg(trans->net, trans, MNCC_CALL_CONF_IND, - &call_conf); + /* Directly ack with MNCC_CALL_CONF_IND, not yet containing SDP or RTP IP:port information. */ + new_cc_state(trans, GSM_CSTATE_MO_TERM_CALL_CONF); + return mncc_recvmsg(trans->net, trans, MNCC_CALL_CONF_IND, &call_conf); +} + +static int mncc_recv_rtp(struct gsm_network *net, struct gsm_trans *trans, uint32_t callref, + int cmd, struct osmo_sockaddr_str *rtp_addr, uint32_t payload_type, + uint32_t payload_msg_type, const struct sdp_msg *sdp); + +int gsm48_cc_mt_rtp_port_and_codec_known(struct gsm_trans *trans) +{ + struct msc_a *msc_a = trans->msc_a; + struct osmo_sockaddr_str *rtp_cn_local; + struct rtp_stream *rtp_ran; + struct gsm_mncc_rtp; + + if (!msc_a) { + LOG_TRANS(trans, LOGL_ERROR, "No connection for CC trans\n"); + trans->callref = 0; + trans_free(trans); + return -EINVAL; + } + + /* Set chosen codec in SDP. This is the result of the Assignment, the actual codec the BSS has chosen for this + * MT side. */ + rtp_ran = msc_a->cc.call_leg->rtp[RTP_TO_RAN]; + if (!rtp_ran->codecs_known) { + LOG_TRANS(trans, LOGL_ERROR, "RAN codecs not known but should be, cannot continue.\n"); + trans_free(trans); + return -EINVAL; + } + trans->cc.sdp.assignment = rtp_ran->codecs.codec[0]; + + /* Insert the CN side RTP port now available into SDP */ + rtp_cn_local = call_leg_local_ip(msc_a->cc.call_leg, RTP_TO_CN); + if (!rtp_cn_local) { + LOG_TRANS(trans, LOGL_ERROR, "Cannot compose SDP for MNCC_RTP_CREATE: no RTP set up for the CN side\n"); + trans_free(trans); + return -EINVAL; + } + trans->cc.sdp.result.rtp = *rtp_cn_local; + + cc_sdp_filter(&trans->cc.sdp); + LOG_TRANS(trans, LOGL_DEBUG, "codecs: %s\n", cc_sdp_name(&trans->cc.sdp)); + + return mncc_recv_rtp(msc_a_net(msc_a), trans, trans->callref, MNCC_RTP_CREATE, rtp_cn_local, 0, 0, + &trans->cc.sdp.result); } static int gsm48_cc_tx_call_proc_and_assign(struct gsm_trans *trans, void *arg) @@ -790,6 +951,10 @@ static int gsm48_cc_rx_alerting(struct gsm_trans *trans, struct msgb *msg) new_cc_state(trans, GSM_CSTATE_CALL_RECEIVED); + cc_sdp_filter(&trans->cc.sdp); + LOG_TRANS(trans, LOGL_DEBUG, "codecs: %s\n", cc_sdp_name(&trans->cc.sdp)); + sdp_msg_to_str(alerting.sdp, sizeof(alerting.sdp), &trans->cc.sdp.result); + return mncc_recvmsg(trans->net, trans, MNCC_ALERT_IND, &alerting); } @@ -814,6 +979,19 @@ static int gsm48_cc_tx_alerting(struct gsm_trans *trans, void *arg) new_cc_state(trans, GSM_CSTATE_CALL_DELIVERED); + if (alerting->sdp[0]) { + struct call_leg *cl = trans->msc_a->cc.call_leg; + struct rtp_stream *rtp_cn = cl ? cl->rtp[RTP_TO_CN] : NULL; + sdp_msg_from_str(&trans->cc.sdp.remote, alerting->sdp); + LOG_TRANS(trans, LOGL_DEBUG, "%s codecs: %s\n", + get_mncc_name(alerting->msg_type), + cc_sdp_name(&trans->cc.sdp)); + if (rtp_cn) { + rtp_stream_set_remote_addr_and_codecs(rtp_cn, &trans->cc.sdp.remote); + rtp_stream_commit(rtp_cn); + } + } + return trans_tx_gsm48(trans, msg); } @@ -860,6 +1038,20 @@ static int gsm48_cc_tx_connect(struct gsm_trans *trans, void *arg) new_cc_state(trans, GSM_CSTATE_CONNECT_IND); + /* Received an MNCC_SETUP_RSP with the remote leg's SDP information. Apply codec choice. */ + if (connect->sdp[0]) { + struct call_leg *cl = trans->msc_a->cc.call_leg; + struct rtp_stream *rtp_cn = cl ? cl->rtp[RTP_TO_CN] : NULL; + sdp_msg_from_str(&trans->cc.sdp.remote, connect->sdp); + LOG_TRANS(trans, LOGL_DEBUG, "%s codecs: %s\n", + get_mncc_name(connect->msg_type), + cc_sdp_name(&trans->cc.sdp)); + if (rtp_cn) { + rtp_stream_set_remote_addr_and_codecs(rtp_cn, &trans->cc.sdp.remote); + rtp_stream_commit(rtp_cn); + } + } + return trans_tx_gsm48(trans, msg); } @@ -902,6 +1094,8 @@ static int gsm48_cc_rx_connect(struct gsm_trans *trans, struct msgb *msg) new_cc_state(trans, GSM_CSTATE_CONNECT_REQUEST); rate_ctr_inc(&trans->net->msc_ctrs->ctr[MSC_CTR_CALL_MT_CONNECT]); + cc_sdp_filter(&trans->cc.sdp); + sdp_msg_to_str(connect.sdp, sizeof(connect.sdp), &trans->cc.sdp.result); return mncc_recvmsg(trans->net, trans, MNCC_SETUP_CNF, &connect); } @@ -1027,7 +1221,6 @@ static int gsm48_cc_rx_release(struct gsm_trans *trans, struct msgb *msg) unsigned int payload_len = msgb_l3len(msg) - sizeof(*gh); struct tlv_parsed tp; struct gsm_mncc rel; - int rc; gsm48_stop_cc_timer(trans); @@ -1059,14 +1252,16 @@ static int gsm48_cc_rx_release(struct gsm_trans *trans, struct msgb *msg) TLVP_VAL(&tp, GSM48_IE_SS_VERS)-1); } - if (trans->cc.state == GSM_CSTATE_RELEASE_REQ) { - /* release collision 5.4.5 */ - rc = mncc_recvmsg(trans->net, trans, MNCC_REL_CNF, &rel); - } else { - rc = gsm48_tx_simple(trans->msc_a, - GSM48_PDISC_CC | (trans->transaction_id << 4), - GSM48_MT_CC_RELEASE_COMPL); - rc = mncc_recvmsg(trans->net, trans, MNCC_REL_IND, &rel); + if (!trans->cc.mncc_release_sent) { + if (trans->cc.state == GSM_CSTATE_RELEASE_REQ) { + /* release collision 5.4.5 */ + mncc_recvmsg(trans->net, trans, MNCC_REL_CNF, &rel); + } else { + gsm48_tx_simple(trans->msc_a, + GSM48_PDISC_CC | (trans->transaction_id << 4), + GSM48_MT_CC_RELEASE_COMPL); + mncc_recvmsg(trans->net, trans, MNCC_REL_IND, &rel); + } } new_cc_state(trans, GSM_CSTATE_NULL); @@ -1074,7 +1269,7 @@ static int gsm48_cc_rx_release(struct gsm_trans *trans, struct msgb *msg) trans->callref = 0; trans_free(trans); - return rc; + return 0; } static int gsm48_cc_tx_release(struct gsm_trans *trans, void *arg) @@ -1153,19 +1348,21 @@ static int gsm48_cc_rx_release_compl(struct gsm_trans *trans, struct msgb *msg) TLVP_VAL(&tp, GSM48_IE_SS_VERS)-1); } - if (trans->callref) { - switch (trans->cc.state) { - case GSM_CSTATE_CALL_PRESENT: - rc = mncc_recvmsg(trans->net, trans, - MNCC_REJ_IND, &rel); - break; - case GSM_CSTATE_RELEASE_REQ: - rc = mncc_recvmsg(trans->net, trans, - MNCC_REL_CNF, &rel); - break; - default: - rc = mncc_recvmsg(trans->net, trans, - MNCC_REL_IND, &rel); + if (!trans->cc.mncc_release_sent) { + if (trans->callref) { + switch (trans->cc.state) { + case GSM_CSTATE_CALL_PRESENT: + rc = mncc_recvmsg(trans->net, trans, + MNCC_REJ_IND, &rel); + break; + case GSM_CSTATE_RELEASE_REQ: + rc = mncc_recvmsg(trans->net, trans, + MNCC_REL_CNF, &rel); + break; + default: + rc = mncc_recvmsg(trans->net, trans, + MNCC_REL_IND, &rel); + } } } @@ -1612,7 +1809,7 @@ static int gsm48_cc_rx_userinfo(struct gsm_trans *trans, struct msgb *msg) static int mncc_recv_rtp(struct gsm_network *net, struct gsm_trans *trans, uint32_t callref, int cmd, struct osmo_sockaddr_str *rtp_addr, uint32_t payload_type, - uint32_t payload_msg_type) + uint32_t payload_msg_type, const struct sdp_msg *sdp) { uint8_t data[sizeof(struct gsm_mncc)]; struct gsm_mncc_rtp *rtp; @@ -1628,12 +1825,18 @@ static int mncc_recv_rtp(struct gsm_network *net, struct gsm_trans *trans, uint3 } rtp->payload_type = payload_type; rtp->payload_msg_type = payload_msg_type; + if (sdp) { + LOG_TRANS(trans, LOGL_DEBUG, "%s SDP: %s\n", + get_mncc_name(rtp->msg_type), + sdp_msg_name(sdp)); + sdp_msg_to_str(rtp->sdp, sizeof(rtp->sdp), sdp); + } return mncc_recvmsg(net, trans, cmd, (struct gsm_mncc *)data); } static void mncc_recv_rtp_err(struct gsm_network *net, struct gsm_trans *trans, uint32_t callref, int cmd) { - mncc_recv_rtp(net, trans, callref, cmd, NULL, 0, 0); + mncc_recv_rtp(net, trans, callref, cmd, NULL, 0, 0, NULL); } static int tch_rtp_create(struct gsm_network *net, uint32_t callref) @@ -1659,6 +1862,57 @@ static int tch_rtp_create(struct gsm_network *net, uint32_t callref) return msc_a_try_call_assignment(trans); } +int cc_cn_local_rtp_port_known(struct gsm_trans *cc_trans) +{ + switch(cc_trans->cc.state) { + case GSM_CSTATE_INITIATED: + if (cc_trans->cc.msg.msg_type != MNCC_SETUP_IND) { + LOG_TRANS(cc_trans, LOGL_ERROR, "Assuming MO call, expected MNCC_SETUP_IND to be prepared\n"); + return -EINVAL; + } + /* This is the MO call leg, waiting for a CN RTP be able to send initial MNCC_SETUP_IND. */ + gsm48_cc_rx_setup_cn_local_rtp_port_known(cc_trans); + return 0; + + case GSM_CSTATE_MO_TERM_CALL_CONF: + /* This is the MT call leg, waiting for a CN RTP to be able to send MNCC_CALL_CONF_IND. */ + return gsm48_cc_mt_rtp_port_and_codec_known(cc_trans); + + default: + LOG_TRANS(cc_trans, LOGL_ERROR, "CN RTP address available, but in unexpected state %d\n", + cc_trans->cc.state); + return -EINVAL; + } +} + +int cc_assignment_done(struct gsm_trans *trans) +{ + struct msc_a *msc_a = trans->msc_a; + + switch (trans->cc.state) { + case GSM_CSTATE_INITIATED: + case GSM_CSTATE_MO_CALL_PROC: + /* MO call */ + break; + + case GSM_CSTATE_CALL_RECEIVED: + case GSM_CSTATE_MO_TERM_CALL_CONF: + /* MT call */ + break; + + default: + LOG_TRANS(trans, LOGL_ERROR, "Assignment done in unexpected CC state: %d\n", trans->cc.state); + return -EINVAL; + } + + if (!call_leg_local_ip(msc_a->cc.call_leg, RTP_TO_CN)) { + LOG_TRANS(trans, LOGL_DEBUG, + "Assignment complete, but still waiting for the CRCX OK on the CN side RTP\n"); + return 0; + } + return gsm48_tch_rtp_create(trans); +} + /* Trigger TCH_RTP_CREATE acknowledgement */ int gsm48_tch_rtp_create(struct gsm_trans *trans) { @@ -1670,30 +1924,32 @@ int gsm48_tch_rtp_create(struct gsm_trans *trans) struct call_leg *cl = msc_a->cc.call_leg; struct osmo_sockaddr_str *rtp_cn_local; struct rtp_stream *rtp_cn = cl ? cl->rtp[RTP_TO_CN] : NULL; - uint32_t payload_type; - int payload_msg_type; - const struct mgcp_conn_peer *mgcp_info; + int mncc_payload_msg_type; + struct sdp_audio_codec *codec; + const struct codec_mapping *m; if (!rtp_cn) { LOG_TRANS_CAT(trans, DMNCC, LOGL_ERROR, "Cannot RTP CREATE to MNCC, no RTP set up for the CN side\n"); return -EINVAL; } - if (!rtp_cn->codec_known) { + cc_sdp_filter(&trans->cc.sdp); + LOG_TRANS(trans, LOGL_DEBUG, "codecs: %s\n", cc_sdp_name(&trans->cc.sdp)); + + if (!trans->cc.sdp.result.audio_codecs.count) { LOG_TRANS_CAT(trans, DMNCC, LOGL_ERROR, - "Cannot RTP CREATE to MNCC, no codec set up for the RTP CN side\n"); + "Cannot RTP CREATE to MNCC, there is no codec available\n"); return -EINVAL; } - /* Codec */ - payload_msg_type = mgcp_codec_to_mncc_payload_msg_type(rtp_cn->codec); + /* Modify the MGW endpoint if necessary, usually this should already match and not cause MGCP. */ + rtp_stream_set_codecs(rtp_cn, &trans->cc.sdp.result.audio_codecs); + rtp_stream_commit(rtp_cn); - /* Payload Type number */ - mgcp_info = osmo_mgcpc_ep_ci_get_rtp_info(rtp_cn->ci); - if (mgcp_info && mgcp_info->ptmap_len) - payload_type = map_codec_to_pt(mgcp_info->ptmap, mgcp_info->ptmap_len, rtp_cn->codec); - else - payload_type = rtp_cn->codec; + /* Populate the legacy MNCC codec elements: payload_type and payload_msg_type */ + codec = &rtp_cn->codecs.codec[0]; + m = codec_mapping_by_subtype_name(codec->subtype_name); + mncc_payload_msg_type = m ? m->mncc_payload_msg_type : 0; rtp_cn_local = call_leg_local_ip(cl, RTP_TO_CN); if (!rtp_cn_local) { @@ -1701,7 +1957,9 @@ int gsm48_tch_rtp_create(struct gsm_trans *trans) return -EINVAL; } - return mncc_recv_rtp(net, trans, trans->callref, MNCC_RTP_CREATE, rtp_cn_local, payload_type, payload_msg_type); + return mncc_recv_rtp(net, trans, trans->callref, MNCC_RTP_CREATE, rtp_cn_local, + codec->payload_type, mncc_payload_msg_type, + &trans->cc.sdp.result); } static int tch_rtp_connect(struct gsm_network *net, const struct gsm_mncc_rtp *rtp) @@ -1709,7 +1967,6 @@ static int tch_rtp_connect(struct gsm_network *net, const struct gsm_mncc_rtp *r struct gsm_trans *trans; struct call_leg *cl; struct rtp_stream *rtps; - struct osmo_sockaddr_str rtp_addr; /* FIXME: in *rtp we should get the codec information of the remote * leg. We will have to populate trans->conn->rtp.codec_cn with a @@ -1735,7 +1992,7 @@ static int tch_rtp_connect(struct gsm_network *net, const struct gsm_mncc_rtp *r return -EIO; } - LOG_TRANS_CAT(trans, DMNCC, LOGL_DEBUG, "rx %s\n", get_mncc_name(MNCC_RTP_CONNECT)); + LOG_TRANS_CAT(trans, DMNCC, LOGL_DEBUG, "rx %s\n", get_mncc_name(rtp->msg_type)); cl = trans->msc_a->cc.call_leg; rtps = cl ? cl->rtp[RTP_TO_CN] : NULL; @@ -1746,8 +2003,21 @@ static int tch_rtp_connect(struct gsm_network *net, const struct gsm_mncc_rtp *r return -EINVAL; } - osmo_sockaddr_str_from_32n(&rtp_addr, rtp->ip, rtp->port); - rtp_stream_set_remote_addr(rtps, &rtp_addr); + if (rtp->sdp[0]) { + sdp_msg_from_str(&trans->cc.sdp.remote, rtp->sdp); + LOG_TRANS(trans, LOGL_DEBUG, "%s contained SDP %s\n", + get_mncc_name(rtp->msg_type), + sdp_msg_name(&trans->cc.sdp.remote)); + } + rtp_stream_set_remote_addr_and_codecs(rtps, &trans->cc.sdp.remote); + + if (!osmo_sockaddr_str_is_nonzero(&rtps->remote)) { + /* Didn't get an IP address from SDP. Try legacy MNCC IP address */ + struct osmo_sockaddr_str rtp_addr; + osmo_sockaddr_str_from_32n(&rtp_addr, rtp->ip, rtp->port); + rtp_stream_set_remote_addr(rtps, &rtp_addr); + } + rtp_stream_commit(rtps); return 0; } @@ -1929,6 +2199,19 @@ static int mncc_tx_to_gsm_cc(struct gsm_network *net, const union mncc_msg *msg) return -ENOMEM; } + /* Remember remote SDP, if any */ + if (data->sdp[0]) { + if (sdp_msg_from_str(&trans->cc.sdp.remote, data->sdp)) { + LOG_TRANS(trans, LOGL_ERROR, "Failed to parse incoming SDP: %s\n", + osmo_quote_str(data->sdp, -1)); + vlr_subscr_put(vsub, __func__); + mncc_release_ind(net, NULL, data->callref, + GSM48_CAUSE_LOC_PRN_S_LU, + GSM48_CC_CAUSE_NORMAL_UNSPEC); + return -EINVAL; + } + } + /* If subscriber has no conn */ if (!msc_a) { /* This condition will return before the common logging of the received MNCC message below, so @@ -1976,6 +2259,7 @@ static int mncc_tx_to_gsm_cc(struct gsm_network *net, const union mncc_msg *msg) LOG_TRANS(trans, LOGL_DEBUG, "rx %s in paging state\n", get_mncc_name(msg->msg_type)); mncc_set_cause(&rel, GSM48_CAUSE_LOC_PRN_S_LU, GSM48_CC_CAUSE_NORM_CALL_CLEAR); + trans->cc.mncc_release_sent = true; if (msg->msg_type == MNCC_REL_REQ) rc = mncc_recvmsg(net, trans, MNCC_REL_CNF, &rel); else diff --git a/src/libmsc/mncc_call.c b/src/libmsc/mncc_call.c index 0deb9037f..12da197d2 100644 --- a/src/libmsc/mncc_call.c +++ b/src/libmsc/mncc_call.c @@ -35,6 +35,7 @@ #include <osmocom/msc/rtp_stream.h> #include <osmocom/msc/msub.h> #include <osmocom/msc/vlr.h> +#include <osmocom/msc/codec_sdp_cc_t9n.h> struct osmo_fsm mncc_call_fsm; static bool mncc_call_tx_rtp_create(struct mncc_call *mncc_call); @@ -261,37 +262,18 @@ static bool mncc_call_rx_rtp_create(struct mncc_call *mncc_call) return true; } - if (!mncc_call->rtps->codec_known) { + if (!mncc_call->rtps->codecs_known) { LOG_MNCC_CALL(mncc_call, LOGL_DEBUG, "Got RTP_CREATE, but RTP stream has no codec set\n"); return true; } LOG_MNCC_CALL(mncc_call, LOGL_DEBUG, "Got RTP_CREATE, responding with " OSMO_SOCKADDR_STR_FMT " %s\n", OSMO_SOCKADDR_STR_FMT_ARGS(&mncc_call->rtps->local), - osmo_mgcpc_codec_name(mncc_call->rtps->codec)); + sdp_audio_codecs_name(&mncc_call->rtps->codecs)); /* Already know what RTP IP:port to tell the MNCC. Send it. */ return mncc_call_tx_rtp_create(mncc_call); } -/* Convert enum mgcp_codecs to an gsm_mncc_rtp->payload_msg_type value. */ -uint32_t mgcp_codec_to_mncc_payload_msg_type(enum mgcp_codecs codec) -{ - switch (codec) { - default: - /* disclaimer: i have no idea what i'm doing. */ - case CODEC_GSM_8000_1: - return GSM_TCHF_FRAME; - case CODEC_GSMEFR_8000_1: - return GSM_TCHF_FRAME_EFR; - case CODEC_GSMHR_8000_1: - return GSM_TCHH_FRAME; - case CODEC_AMR_8000_1: - case CODEC_AMRWB_16000_1: - //return GSM_TCHF_FRAME; - return GSM_TCH_FRAME_AMR; - } -} - static bool mncc_call_tx_rtp_create(struct mncc_call *mncc_call) { if (!mncc_call->rtps || !osmo_sockaddr_str_is_nonzero(&mncc_call->rtps->local)) { @@ -313,9 +295,16 @@ static bool mncc_call_tx_rtp_create(struct mncc_call *mncc_call) return false; } - if (mncc_call->rtps->codec_known) { - mncc_msg.rtp.payload_type = 0; /* ??? */ - mncc_msg.rtp.payload_msg_type = mgcp_codec_to_mncc_payload_msg_type(mncc_call->rtps->codec); + if (mncc_call->rtps->codecs_known) { + struct sdp_audio_codec *codec = &mncc_call->rtps->codecs.codec[0]; + const struct codec_mapping *m = codec_mapping_by_subtype_name(codec->subtype_name); + + if (!m) { + mncc_call_error(mncc_call, "Failed to resolve audio codec '%s'\n", sdp_audio_codec_name(codec)); + return false; + } + mncc_msg.rtp.payload_type = codec->payload_type; + mncc_msg.rtp.payload_msg_type = m->mncc_payload_msg_type; } if (mncc_call_tx(mncc_call, &mncc_msg)) diff --git a/src/libmsc/msc_a.c b/src/libmsc/msc_a.c index bc5b7ea93..ee153410e 100644 --- a/src/libmsc/msc_a.c +++ b/src/libmsc/msc_a.c @@ -46,6 +46,7 @@ #include <osmocom/msc/call_leg.h> #include <osmocom/msc/rtp_stream.h> #include <osmocom/msc/msc_ho.h> +#include <osmocom/msc/codec_sdp_cc_t9n.h> #define MSC_A_USE_WAIT_CLEAR_COMPLETE "wait-Clear-Complete" @@ -511,12 +512,87 @@ static void msc_a_fsm_authenticated(struct osmo_fsm_inst *fi, uint32_t event, vo } } +static struct call_leg *msc_a_ensure_call_leg(struct msc_a *msc_a, struct gsm_trans *for_cc_trans) +{ + struct call_leg *cl = msc_a->cc.call_leg; + struct gsm_network *net = msc_a_net(msc_a); + + /* Ensure that events about RTP endpoints coming from the msc_a->cc.call_leg know which gsm_trans to abort on + * error */ + if (!msc_a->cc.active_trans) + msc_a->cc.active_trans = for_cc_trans; + if (msc_a->cc.active_trans != for_cc_trans) { + LOG_TRANS(for_cc_trans, LOGL_ERROR, + "Cannot create call leg, another trans is already active for this conn\n"); + return NULL; + } + + if (!cl) { + cl = msc_a->cc.call_leg = call_leg_alloc(msc_a->c.fi, + MSC_EV_CALL_LEG_TERM, + MSC_EV_CALL_LEG_RTP_LOCAL_ADDR_AVAILABLE, + MSC_EV_CALL_LEG_RTP_COMPLETE); + OSMO_ASSERT(cl); + + /* HACK: We put the connection in loopback mode at the beginnig to + * trick the hNodeB into doing the IuUP negotiation with itself. + * This is a hack we need because osmo-mgw does not support IuUP yet, see OS#2459. */ + if (msc_a->c.ran->type == OSMO_RAT_UTRAN_IU) + cl->crcx_conn_mode[RTP_TO_RAN] = MGCP_CONN_LOOPBACK; + + if (net->use_osmux != OSMUX_USAGE_OFF) { + struct msc_i *msc_i = msc_a_msc_i(msc_a); + if (msc_i->c.remote_to) { + /* TODO: investigate what to do in this case */ + LOG_MSC_A(msc_a, LOGL_ERROR, "Osmux not yet supported for inter-MSC"); + } else { + cl->ran_peer_supports_osmux = msc_i->ran_conn->ran_peer->remote_supports_osmux; + } + } + + } + return cl; +} + +int msc_a_ensure_cn_local_rtp(struct msc_a *msc_a, struct gsm_trans *cc_trans) +{ + struct call_leg *cl; + struct rtp_stream *rtp_to_ran; + + cl = msc_a_ensure_call_leg(msc_a, cc_trans); + if (!cl) + return -EINVAL; + rtp_to_ran = cl->rtp[RTP_TO_RAN]; + + if (call_leg_local_ip(cl, RTP_TO_CN)) { + /* Already has an RTP address and port towards the CN, continue right away. */ + return osmo_fsm_inst_dispatch(msc_a->c.fi, MSC_EV_CALL_LEG_RTP_LOCAL_ADDR_AVAILABLE, cl->rtp[RTP_TO_CN]); + } + + /* No CN RTP address available yet, ask the MGW to create one. + * Set a codec to be used: if Assignment on the RAN side is already done, take the same codec as the RTP_TO_RAN. + * If no RAN side RTP is established, try to guess a preliminary codec from SDP -- before Assignment, picking a + * codec from the SDP is more politeness/avoiding confusion than necessity. The actual codec to be used would be + * determined later. If no codec could be determined, pass none for the time being. */ + return call_leg_ensure_ci(cl, RTP_TO_CN, cc_trans->callref, cc_trans, + rtp_to_ran->codecs_known ? &rtp_to_ran->codecs : NULL, NULL); +} + +static void msc_a_call_leg_cn_local_addr_available(struct msc_a *msc_a, struct gsm_trans *cc_trans) +{ + cc_cn_local_rtp_port_known(cc_trans); +} + + /* The MGW has given us a local IP address for the RAN side. Ready to start the Assignment of a voice channel. */ static void msc_a_call_leg_ran_local_addr_available(struct msc_a *msc_a) { struct ran_msg msg; struct gsm_trans *cc_trans = msc_a->cc.active_trans; - struct gsm0808_channel_type channel_type; + struct gsm0808_channel_type channel_type = { + .ch_indctr = GSM0808_CHAN_SPEECH, + .ch_rate_type = GSM0808_SPEECH_FULL_PREF, + }; if (!cc_trans) { LOG_MSC_A(msc_a, LOGL_ERROR, "No CC transaction active\n"); @@ -526,9 +602,21 @@ static void msc_a_call_leg_ran_local_addr_available(struct msc_a *msc_a) /* Once a CI is known, we could also CRCX the CN side of the MGW endpoint, but it makes sense to wait for the * codec to be determined by the Assignment Complete message, first. */ + cc_sdp_filter(&cc_trans->cc.sdp); + LOG_TRANS(cc_trans, LOGL_DEBUG, "Sending Assignment Command with codecs: %s\n", cc_sdp_name(&cc_trans->cc.sdp)); - if (mncc_bearer_cap_to_channel_type(&channel_type, &cc_trans->bearer_cap)) { - LOG_MSC_A(msc_a, LOGL_ERROR, "Cannot compose Channel Type from bearer capabilities\n"); + if (!cc_trans->cc.sdp.result.audio_codecs.count) { + LOG_TRANS(cc_trans, LOGL_ERROR, "Assignment not possible, no matching codec: %s\n", + cc_sdp_name(&cc_trans->cc.sdp)); + call_leg_release(msc_a->cc.call_leg); + return; + } + + /* Compose 48.008 Channel Type from the current set of codecs determined from both local and remote codec + * capabilities. */ + if (sdp_audio_codecs_to_gsm0808_channel_type(&channel_type, &cc_trans->cc.sdp.result.audio_codecs)) { + LOG_MSC_A(msc_a, LOGL_ERROR, "Cannot compose Channel Type (Permitted Speech) from codecs: %s\n", + cc_sdp_name(&cc_trans->cc.sdp)); trans_free(cc_trans); return; } @@ -550,15 +638,6 @@ static void msc_a_call_leg_ran_local_addr_available(struct msc_a *msc_a) } } -static void msc_a_call_leg_cn_local_addr_available(struct msc_a *msc_a, struct gsm_trans *cc_trans) -{ - if (gsm48_tch_rtp_create(cc_trans)) { - LOG_MSC_A(msc_a, LOGL_ERROR, "Cannot inform MNCC of RTP address\n"); - trans_free(cc_trans); - return; - } -} - static struct gsm_trans *find_waiting_call(struct msc_a *msc_a) { struct gsm_trans *trans; @@ -1264,6 +1343,7 @@ static void msc_a_up_call_assignment_complete(struct msc_a *msc_a, const struct { struct gsm_trans *cc_trans = msc_a->cc.active_trans; struct rtp_stream *rtps_to_ran = msc_a->cc.call_leg ? msc_a->cc.call_leg->rtp[RTP_TO_RAN] : NULL; + const struct codec_mapping *m; if (!rtps_to_ran) { LOG_MSC_A(msc_a, LOGL_ERROR, "Rx Assignment Complete, but no RTP stream is set up\n"); @@ -1281,24 +1361,30 @@ static void msc_a_up_call_assignment_complete(struct msc_a *msc_a, const struct return; } + m = codec_mapping_by_mgcp_codec(ac->assignment_complete.codec); + if (!m) { + LOG_TRANS(cc_trans, LOGL_ERROR, "Unknown codec in Assignment Complete: %s\n", + osmo_mgcpc_codec_name(ac->assignment_complete.codec)); + call_leg_release(msc_a->cc.call_leg); + return; + } + /* Update RAN-side endpoint CI: */ - rtp_stream_set_codec(rtps_to_ran, ac->assignment_complete.codec); + rtp_stream_set_one_codec(rtps_to_ran, &m->sdp); rtp_stream_set_remote_addr(rtps_to_ran, &ac->assignment_complete.remote_rtp); if (rtps_to_ran->use_osmux) rtp_stream_set_remote_osmux_cid(rtps_to_ran, ac->assignment_complete.osmux_cid); - rtp_stream_commit(rtps_to_ran); - /* Setup CN side endpoint CI: - * Now that - * - the first CI has been created and a definitive endpoint name is assigned to the call_leg's MGW - * endpoint, - * - the Assignment has chosen a speech codec - * go on to create the CN side RTP stream's CI. */ - if (call_leg_ensure_ci(msc_a->cc.call_leg, RTP_TO_CN, cc_trans->callref, cc_trans, - &ac->assignment_complete.codec, NULL)) { - LOG_MSC_A_CAT(msc_a, DCC, LOGL_ERROR, "Error creating MGW CI towards CN\n"); + /* Remember the Codec List (BSS Supported) */ + if (ac->assignment_complete.codec_list_bss_supported) + cc_sdp_set_cell(&cc_trans->cc.sdp, ac->assignment_complete.codec_list_bss_supported); + + cc_trans->cc.sdp.assignment = m->sdp; + + if (cc_assignment_done(cc_trans)) { + /* If an error occured, it was logged in cc_assignment_done() */ call_leg_release(msc_a->cc.call_leg); return; } @@ -1377,6 +1463,15 @@ int msc_a_ran_dec_from_msc_i(struct msc_a *msc_a, struct msc_a_ran_dec_data *d) .lai.plmn = msc_a_net(msc_a)->plmn, }; gsm0808_cell_id_to_cgi(&msc_a->via_cell, msg->compl_l3.cell_id); + if (msg->compl_l3.codec_list_bss_supported) { + msc_a->cc.codec_list_bss_supported = *msg->compl_l3.codec_list_bss_supported; + if (log_check_level(msc_a->c.ran->log_subsys, LOGL_DEBUG)) { + struct sdp_audio_codecs ac = {}; + sdp_audio_codecs_from_speech_codec_list(&ac, &msc_a->cc.codec_list_bss_supported); + LOG_MSC_A(msc_a, LOGL_DEBUG, "Complete Layer 3: Codec List (BSS Supported): %s\n", + sdp_audio_codecs_name(&ac)); + } + } rc = msc_a_up_l3(msc_a, msg->compl_l3.msg); if (!rc) { struct ran_conn *conn = msub_ran_conn(msc_a->c.msub); @@ -1643,45 +1738,42 @@ int msc_tx_common_id(struct msc_a *msc_a, enum msc_role to_role) static int msc_a_start_assignment(struct msc_a *msc_a, struct gsm_trans *cc_trans) { - struct call_leg *cl = msc_a->cc.call_leg; - struct msc_i *msc_i = msc_a_msc_i(msc_a); - struct gsm_network *net = msc_a_net(msc_a); + struct call_leg *cl; + bool cn_rtp_available; + bool ran_rtp_available; + struct sdp_audio_codecs *codecs; OSMO_ASSERT(!msc_a->cc.active_trans); msc_a->cc.active_trans = cc_trans; OSMO_ASSERT(cc_trans && cc_trans->type == TRANS_CC); + cl = msc_a_ensure_call_leg(msc_a, cc_trans); + if (!cl) + return -EINVAL; - if (!cl) { - cl = msc_a->cc.call_leg = call_leg_alloc(msc_a->c.fi, - MSC_EV_CALL_LEG_TERM, - MSC_EV_CALL_LEG_RTP_LOCAL_ADDR_AVAILABLE, - MSC_EV_CALL_LEG_RTP_COMPLETE); - OSMO_ASSERT(cl); + /* See if we can set a preliminary codec. If not, pass none for the time being. */ + cc_sdp_filter(&cc_trans->cc.sdp); + codecs = cc_trans->cc.sdp.result.audio_codecs.count ? &cc_trans->cc.sdp.result.audio_codecs : NULL; - /* HACK: We put the connection in loopback mode at the beginning to - * trick the hNodeB into doing the IuUP negotiation with itself. - * This is a hack we need because osmo-mgw does not support IuUP yet, see OS#2459. */ - if (msc_a->c.ran->type == OSMO_RAT_UTRAN_IU) - cl->crcx_conn_mode[RTP_TO_RAN] = MGCP_CONN_LOOPBACK; - } + cn_rtp_available = call_leg_local_ip(cl, RTP_TO_CN); + ran_rtp_available = call_leg_local_ip(cl, RTP_TO_RAN); - if (net->use_osmux != OSMUX_USAGE_OFF) { - msc_i = msc_a_msc_i(msc_a); - if (msc_i->c.remote_to) { - /* TODO: investigate what to do in this case */ - LOG_MSC_A(msc_a, LOGL_ERROR, "Osmux not yet supported for inter-MSC"); - } else { - cl->ran_peer_supports_osmux = msc_i->ran_conn->ran_peer->remote_supports_osmux; - } - } + /* Set up RTP ports for both RAN and CN side. Even though we ask for both at the same time, the + * osmo_mgcpc_ep_fsm automagically waits for the first CRCX to complete before firing the second CRCX. The one + * issued first here will also be the first CRCX sent to the MGW. Usually both still need to be set up. */ + if (!cn_rtp_available) + call_leg_ensure_ci(cl, RTP_TO_CN, cc_trans->callref, cc_trans, codecs, NULL); + if (!ran_rtp_available) + call_leg_ensure_ci(cl, RTP_TO_RAN, cc_trans->callref, cc_trans, codecs, NULL); - /* This will lead to either MSC_EV_CALL_LEG_LOCAL_ADDR_AVAILABLE or MSC_EV_CALL_LEG_TERM. - * If the local address is already known, then immediately trigger. */ - if (call_leg_local_ip(cl, RTP_TO_RAN)) + /* Should these already be set up, immediately continue by retriggering the events signalling that the RTP + * ports are available. The ordering is: first CN, then RAN. */ + if (cn_rtp_available && ran_rtp_available) return osmo_fsm_inst_dispatch(msc_a->c.fi, MSC_EV_CALL_LEG_RTP_LOCAL_ADDR_AVAILABLE, cl->rtp[RTP_TO_RAN]); - else - return call_leg_ensure_ci(msc_a->cc.call_leg, RTP_TO_RAN, cc_trans->callref, cc_trans, NULL, NULL); + else if (cn_rtp_available) + return osmo_fsm_inst_dispatch(msc_a->c.fi, MSC_EV_CALL_LEG_RTP_LOCAL_ADDR_AVAILABLE, cl->rtp[RTP_TO_CN]); + /* Otherwise wait for MGCP response and continue from there. */ + return 0; } int msc_a_try_call_assignment(struct gsm_trans *cc_trans) diff --git a/src/libmsc/msc_ho.c b/src/libmsc/msc_ho.c index aa513eb58..0db971793 100644 --- a/src/libmsc/msc_ho.c +++ b/src/libmsc/msc_ho.c @@ -43,6 +43,7 @@ #include <osmocom/msc/call_leg.h> #include <osmocom/msc/rtp_stream.h> #include <osmocom/msc/mncc_call.h> +#include <osmocom/msc/codec_sdp_cc_t9n.h> struct osmo_fsm msc_ho_fsm; @@ -563,7 +564,7 @@ static int msc_ho_start_inter_msc_call_forwarding(struct msc_a *msc_a, struct ms /* Backup old cell's RTP IP:port and codec data */ msc_a->ho.old_cell.ran_remote_rtp = rtp_to_ran->remote; - msc_a->ho.old_cell.codec = rtp_to_ran->codec; + msc_a->ho.old_cell.codecs = rtp_to_ran->codecs; /* Blindly taken over from an MNCC trace of existing code: send an all-zero CCCAP: */ outgoing_call_req.fields |= MNCC_F_CCCAP; @@ -700,7 +701,7 @@ static void msc_ho_rtp_switch_to_new_cell(struct msc_a *msc_a) /* Backup old cell's RTP IP:port and codec data */ msc_a->ho.old_cell.ran_remote_rtp = rtp_to_ran->remote; - msc_a->ho.old_cell.codec = rtp_to_ran->codec; + msc_a->ho.old_cell.codecs = rtp_to_ran->codecs; LOG_HO(msc_a, LOGL_DEBUG, "Switching RTP stream to new cell: from " OSMO_SOCKADDR_STR_FMT " to " OSMO_SOCKADDR_STR_FMT "\n", OSMO_SOCKADDR_STR_FMT_ARGS(&msc_a->ho.old_cell.ran_remote_rtp), @@ -719,9 +720,14 @@ static void msc_ho_rtp_switch_to_new_cell(struct msc_a *msc_a) /* Switch over to the new peer */ rtp_stream_set_remote_addr(rtp_to_ran, &msc_a->ho.new_cell.ran_remote_rtp); - if (msc_a->ho.new_cell.codec_present) - rtp_stream_set_codec(rtp_to_ran, msc_a->ho.new_cell.codec); - else + if (msc_a->ho.new_cell.codec_present) { + struct sdp_audio_codecs codecs = {}; + if (!sdp_audio_codecs_add_mgcp_codec(&codecs, msc_a->ho.new_cell.codec)) { + LOG_HO(msc_a, LOGL_ERROR, + "Cannot resolve codec: %s\n", osmo_mgcpc_codec_name(msc_a->ho.new_cell.codec)); + } else + rtp_stream_set_codecs(rtp_to_ran, &codecs); + } else LOG_HO(msc_a, LOGL_ERROR, "No codec is set\n"); rtp_stream_commit(rtp_to_ran); } @@ -761,7 +767,7 @@ static void msc_ho_rtp_rollback_to_old_cell(struct msc_a *msc_a) /* Switch back to the old cell */ rtp_stream_set_remote_addr(rtp_to_ran, &msc_a->ho.old_cell.ran_remote_rtp); - rtp_stream_set_codec(rtp_to_ran, msc_a->ho.old_cell.codec); + rtp_stream_set_codecs(rtp_to_ran, &msc_a->ho.old_cell.codecs); rtp_stream_commit(rtp_to_ran); } diff --git a/src/libmsc/msc_t.c b/src/libmsc/msc_t.c index af0ddaaef..413ffd10d 100644 --- a/src/libmsc/msc_t.c +++ b/src/libmsc/msc_t.c @@ -454,9 +454,12 @@ static int msc_t_patch_and_send_ho_request_ack(struct msc_t *msc_t, const struct if (r->codec_present) { LOG_MSC_T(msc_t, LOGL_DEBUG, "From Handover Request Ack, got %s\n", osmo_mgcpc_codec_name(r->codec)); - rtp_stream_set_codec(rtp_ran, r->codec); + if (!rtp_stream_set_codecs_from_mgcp_codec(rtp_ran, r->codec)) { + LOG_MSC_T(msc_t, LOGL_ERROR, "Cannot resolve codec in Handover Request Ack: %s\n", + osmo_mgcpc_codec_name(r->codec)); + } if (rtp_cn) - rtp_stream_set_codec(rtp_cn, r->codec); + rtp_stream_set_codecs_from_mgcp_codec(rtp_cn, r->codec); } else { LOG_MSC_T(msc_t, LOGL_DEBUG, "No codec in Handover Request Ack\n"); } diff --git a/src/libmsc/rtp_stream.c b/src/libmsc/rtp_stream.c index 29025204f..66c25f890 100644 --- a/src/libmsc/rtp_stream.c +++ b/src/libmsc/rtp_stream.c @@ -28,6 +28,7 @@ #include <osmocom/msc/transaction.h> #include <osmocom/msc/call_leg.h> #include <osmocom/msc/rtp_stream.h> +#include <osmocom/msc/codec_sdp_cc_t9n.h> #define LOG_RTPS(rtps, level, fmt, args...) \ LOGPFSML(rtps->fi, level, fmt, ##args) @@ -78,10 +79,10 @@ void rtp_stream_update_id(struct rtp_stream *rtps) OSMO_STRBUF_PRINTF(sb, ":no-remote-port"); else if (!rtps->remote_sent_to_mgw) OSMO_STRBUF_PRINTF(sb, ":remote-port-not-sent"); - if (!rtps->codec_known) - OSMO_STRBUF_PRINTF(sb, ":no-codec"); - else if (!rtps->codec_sent_to_mgw) - OSMO_STRBUF_PRINTF(sb, ":codec-not-sent"); + if (!rtps->codecs_known) + OSMO_STRBUF_PRINTF(sb, ":no-codecs"); + else if (!rtps->codecs_sent_to_mgw) + OSMO_STRBUF_PRINTF(sb, ":codecs-not-sent"); if (rtps->use_osmux) { if (rtps->remote_osmux_cid < 0) OSMO_STRBUF_PRINTF(sb, ":no-remote-osmux-cid"); @@ -141,7 +142,7 @@ static void check_established(struct rtp_stream *rtps) && osmo_sockaddr_str_is_nonzero(&rtps->remote) && rtps->remote_sent_to_mgw && (!rtps->use_osmux || rtps->remote_osmux_cid_sent_to_mgw) - && rtps->codec_known) + && rtps->codecs_known) rtp_stream_state_chg(rtps, RTP_STREAM_ST_ESTABLISHED); } @@ -171,14 +172,14 @@ static void rtp_stream_fsm_establishing_established(struct osmo_fsm_inst *fi, ui osmo_fsm_inst_dispatch(fi->proc.parent, CALL_LEG_EV_RTP_STREAM_ADDR_AVAILABLE, rtps); check_established(rtps); - if ((!rtps->remote_sent_to_mgw || !rtps->codec_sent_to_mgw) + if ((!rtps->remote_sent_to_mgw || !rtps->codecs_sent_to_mgw) && osmo_sockaddr_str_is_nonzero(&rtps->remote) && (!rtps->use_osmux || rtps->remote_osmux_cid_sent_to_mgw) - && rtps->codec_known) { + && rtps->codecs_known) { LOG_RTPS(rtps, LOGL_DEBUG, "local ip:port set;%s%s%s triggering MDCX to send the new settings\n", (!rtps->remote_sent_to_mgw)? " remote ip:port not yet sent," : "", - (!rtps->codec_sent_to_mgw)? " codec not yet sent," : "", + (!rtps->codecs_sent_to_mgw)? " codecs not yet sent," : "", (rtps->use_osmux && !rtps->remote_osmux_cid_sent_to_mgw) ? "Osmux CID not yet sent,": ""); rtp_stream_do_mdcx(rtps); } @@ -192,7 +193,7 @@ static void rtp_stream_fsm_establishing_established(struct osmo_fsm_inst *fi, ui case RTP_STREAM_EV_CRCX_FAIL: case RTP_STREAM_EV_MDCX_FAIL: rtps->remote_sent_to_mgw = false; - rtps->codec_sent_to_mgw = false; + rtps->codecs_sent_to_mgw = false; rtps->remote_osmux_cid_sent_to_mgw = false; rtp_stream_update_id(rtps); rtp_stream_state_chg(rtps, RTP_STREAM_ST_DISCARDING); @@ -310,10 +311,25 @@ static int rtp_stream_do_mgcp_verb(struct rtp_stream *rtps, enum mgcp_verb verb, if (verb == MGCP_VERB_CRCX) verb_info.conn_mode = rtps->crcx_conn_mode; - if (rtps->codec_known) { - verb_info.codecs[0] = rtps->codec; - verb_info.codecs_len = 1; - rtps->codec_sent_to_mgw = true; + if (rtps->codecs_known) { + /* Send the list of codecs to the MGW. Ideally we would just feed the SDP directly, but for legacy + * reasons we still need to translate to a struct mgcp_conn_peer representation to send it. */ + struct sdp_audio_codec *codec; + int i = 0; + foreach_sdp_audio_codec(codec, &rtps->codecs) { + const struct codec_mapping *m = codec_mapping_by_subtype_name(codec->subtype_name); + if (!m) + continue; + verb_info.codecs[i] = m->mgcp; + verb_info.ptmap[i] = (struct ptmap){ + .codec = m->mgcp, + .pt = codec->payload_type, + }; + i++; + verb_info.codecs_len = i; + verb_info.ptmap_len = i; + } + rtps->codecs_sent_to_mgw = true; } if (osmo_sockaddr_str_is_nonzero(&rtps->remote)) { int rc = osmo_strlcpy(verb_info.addr, rtps->remote.ip, sizeof(verb_info.addr)); @@ -361,43 +377,75 @@ void rtp_stream_release(struct rtp_stream *rtps) * least one of them has not yet been sent to the MGW in a previous CRCX or MDCX. */ int rtp_stream_commit(struct rtp_stream *rtps) { - if (!rtps->ci) { - LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: no MGW endpoint CI set up\n"); - return -1; - } if (!osmo_sockaddr_str_is_nonzero(&rtps->remote)) { LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: no remote RTP address known\n"); return -1; } - if (!rtps->codec_known) { - LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: no codec known\n"); + if (!rtps->codecs_known) { + LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: no codecs known\n"); return -1; } - if (rtps->remote_sent_to_mgw && rtps->codec_sent_to_mgw) { - LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: both remote RTP address and codec already set up at MGW\n"); + if (rtps->remote_sent_to_mgw && rtps->codecs_sent_to_mgw) { + LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: both remote RTP address and codecs already set up at MGW\n"); return 0; } + if (!rtps->ci) { + LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: no MGW endpoint CI set up\n"); + return -1; + } LOG_RTPS(rtps, LOGL_DEBUG, "Committing: Tx MDCX to update the MGW: updating%s%s%s\n", rtps->remote_sent_to_mgw ? "" : " remote-RTP-IP-port", - rtps->codec_sent_to_mgw ? "" : " codec", + rtps->codecs_sent_to_mgw ? "" : " codecs", (!rtps->use_osmux || rtps->remote_osmux_cid_sent_to_mgw) ? "" : " remote-Osmux-CID"); return rtp_stream_do_mdcx(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) { + if (!codecs || !codecs->count) + return; + if (sdp_audio_codecs_cmp(&rtps->codecs, codecs, false, true) == 0) { + LOG_RTPS(rtps, LOGL_DEBUG, "no change: codecs already set to %s\n", + sdp_audio_codecs_name(&rtps->codecs)); + return; + } if (rtps->fi->state == RTP_STREAM_ST_ESTABLISHED) rtp_stream_state_chg(rtps, RTP_STREAM_ST_ESTABLISHING); - LOG_RTPS(rtps, LOGL_DEBUG, "setting codec to %s\n", osmo_mgcpc_codec_name(codec)); - rtps->codec = codec; - rtps->codec_known = true; - rtps->codec_sent_to_mgw = false; + LOG_RTPS(rtps, LOGL_DEBUG, "setting codecs to %s\n", sdp_audio_codecs_name(codecs)); + rtps->codecs = *codecs; + rtps->codecs_known = true; + rtps->codecs_sent_to_mgw = false; rtp_stream_update_id(rtps); } +/* Convenience shortcut to call rtp_stream_set_codecs() with a list of only one sdp_audio_codec record. */ +void rtp_stream_set_one_codec(struct rtp_stream *rtps, const struct sdp_audio_codec *codec) +{ + struct sdp_audio_codecs codecs = {}; + sdp_audio_codec_add_copy(&codecs, codec); + rtp_stream_set_codecs(rtps, &codecs); +} + +/* For legacy, rather use rtp_stream_set_codecs() with a full codecs list. */ +bool rtp_stream_set_codecs_from_mgcp_codec(struct rtp_stream *rtps, enum mgcp_codecs codec) +{ + struct sdp_audio_codecs codecs = {}; + if (!sdp_audio_codecs_add_mgcp_codec(&codecs, codec)) + return false; + rtp_stream_set_codecs(rtps, &codecs); + return true; +} + void rtp_stream_set_remote_addr(struct rtp_stream *rtps, const struct osmo_sockaddr_str *r) { + if (!strcmp(rtps->remote.ip, r->ip) + && rtps->remote.port == r->port + && rtps->remote.af == r->af) { + LOG_RTPS(rtps, LOGL_DEBUG, "remote addr already " OSMO_SOCKADDR_STR_FMT ", no change\n", + OSMO_SOCKADDR_STR_FMT_ARGS(r)); + return; + } if (rtps->fi->state == RTP_STREAM_ST_ESTABLISHED) rtp_stream_state_chg(rtps, RTP_STREAM_ST_ESTABLISHING); LOG_RTPS(rtps, LOGL_DEBUG, "setting remote addr to " OSMO_SOCKADDR_STR_FMT "\n", OSMO_SOCKADDR_STR_FMT_ARGS(r)); @@ -406,6 +454,13 @@ void rtp_stream_set_remote_addr(struct rtp_stream *rtps, const struct osmo_socka rtp_stream_update_id(rtps); } +void rtp_stream_set_remote_addr_and_codecs(struct rtp_stream *rtps, const struct sdp_msg *sdp) +{ + rtp_stream_set_codecs(rtps, &sdp->audio_codecs); + if (osmo_sockaddr_str_is_nonzero(&sdp->rtp)) + rtp_stream_set_remote_addr(rtps, &sdp->rtp); +} + void rtp_stream_set_remote_osmux_cid(struct rtp_stream *rtps, uint8_t osmux_cid) { if (rtps->fi->state == RTP_STREAM_ST_ESTABLISHED) @@ -425,7 +480,7 @@ bool rtp_stream_is_established(struct rtp_stream *rtps) if (rtps->fi->state != RTP_STREAM_ST_ESTABLISHED) return false; if (!rtps->remote_sent_to_mgw - || !rtps->codec_sent_to_mgw + || !rtps->codecs_sent_to_mgw || (rtps->use_osmux && !rtps->remote_osmux_cid_sent_to_mgw)) return false; return true; diff --git a/src/libmsc/sdp_msg.c b/src/libmsc/sdp_msg.c index 7880978a0..45a8e2c11 100644 --- a/src/libmsc/sdp_msg.c +++ b/src/libmsc/sdp_msg.c @@ -30,31 +30,78 @@ #include <osmocom/msc/debug.h> #include <osmocom/msc/sdp_msg.h> +#define CMP(a,b) (a < b? -1 : (a > b? 1 : 0)) + /* Compare name, rate and fmtp, returning typical cmp result: 0 on match, and -1 / 1 on mismatch. - * Do *not* compare the payload_type number. + * If cmp_fmtp is false, do *not* compare the fmtp string; if true, compare fmtp 1:1 as strings. + * If cmp_payload_type is false, do *not* compare the payload_type number. * The fmtp is only string-compared -- e.g. if AMR parameters appear in a different order, it amounts to a mismatch even * though all parameters are the same. */ -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 rc; + int cmp; if (a == b) return 0; if (!a) return -1; if (!b) return 1; - rc = strncmp(a->subtype_name, b->subtype_name, sizeof(a->subtype_name)); - if (rc) - return rc; + cmp = strncmp(a->subtype_name, b->subtype_name, sizeof(a->subtype_name)); + if (cmp) + return cmp; + cmp = CMP(a->rate, b->rate); + if (cmp) + return cmp; + if (cmp_fmtp) { + cmp = strncmp(a->fmtp, b->fmtp, sizeof(a->fmtp)); + if (cmp) + return cmp; + } + if (cmp_payload_type) { + cmp = CMP(a->payload_type, b->payload_type); + if (cmp) + return cmp; + } + return 0; +} - if (a->rate < b->rate) +int sdp_audio_codecs_cmp(const struct sdp_audio_codecs *a, const struct sdp_audio_codecs *b, + bool cmp_fmtp, bool cmp_payload_type) +{ + const struct sdp_audio_codec *codec_a; + const struct sdp_audio_codec *codec_b; + int cmp; + if (a == b) + return 0; + if (!a) return -1; - if (a->rate > b->rate) + if (!b) return 1; - rc = strncmp(a->fmtp, b->fmtp, sizeof(a->fmtp)); - if (rc) - return rc; + /* The first codec is the "chosen" codec and should match. The others may appear in different order. */ + if (a->count && b->count) { + cmp = sdp_audio_codec_cmp(&a->codec[0], &b->codec[0], cmp_fmtp, cmp_payload_type); + if (cmp) + return cmp; + } + + cmp = CMP(a->count, b->count); + if (cmp) + return cmp; + + /* See if each codec in a is also present in b */ + foreach_sdp_audio_codec(codec_a, a) { + bool match_found = false; + foreach_sdp_audio_codec(codec_b, b) { + if (!sdp_audio_codec_cmp(codec_a, codec_b, cmp_fmtp, cmp_payload_type)) { + match_found = true; + break; + } + } + if (!match_found) + return -1; + } return 0; } @@ -130,13 +177,13 @@ struct sdp_audio_codec *sdp_audio_codec_by_payload_type(struct sdp_audio_codecs return codec; } -/* Return a given sdp_msg's codec entry that matches the subtype_name, rate and fmtp of the given codec, or NULL if no - * match is found. Comparison is made by sdp_audio_codec_cmp(). */ +/* Return a given sdp_msg's codec entry that matches the subtype_name and rate of the given codec, or NULL if no + * match is found. Comparison is made by sdp_audio_codec_cmp(cmp_payload_type=false). */ struct sdp_audio_codec *sdp_audio_codec_by_descr(struct sdp_audio_codecs *ac, const struct sdp_audio_codec *codec) { struct sdp_audio_codec *i; foreach_sdp_audio_codec(i, ac) { - if (!sdp_audio_codec_cmp(i, codec)) + if (!sdp_audio_codec_cmp(i, codec, false, false)) return i; } return NULL; @@ -451,8 +498,8 @@ next_line: } /* Leave only those codecs in 'ac_dest' that are also present in 'ac_other'. - * The matching is made by sdp_audio_codec_cmp(), i.e. payload_type numbers are not compared and fmtp parameters are - * compared 1:1 as plain strings. + * The matching is made by sdp_audio_codec_cmp(cmp_payload_type=false), i.e. payload_type numbers are not compared and + * fmtp parameters are compared 1:1 as plain strings. * If translate_payload_type_numbers has an effect if ac_dest and ac_other have mismatching payload_type numbers for the * same SDP codec descriptions. If translate_payload_type_numbers is true, take the payload_type numbers from ac_other. * If false, keep payload_type numbers in ac_dest unchanged. */ @@ -508,8 +555,11 @@ int sdp_audio_codec_name_buf(char *buf, size_t buflen, const struct sdp_audio_co { struct osmo_strbuf sb = { .buf = buf, .len = buflen }; OSMO_STRBUF_PRINTF(sb, "%s", codec->subtype_name); + if (codec->rate != 8000) + OSMO_STRBUF_PRINTF(sb, "/%u", codec->rate); if (codec->fmtp[0]) OSMO_STRBUF_PRINTF(sb, ":%s", codec->fmtp); + OSMO_STRBUF_PRINTF(sb, "#%d", codec->payload_type); return sb.chars_needed; } |