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transaction: move cc.codecs.result -> cc.local

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Prepare for CSD where this will be used too.

Related: OS#4394
Change-Id: Iaf954be0455625faa06a64c19905b79b7045f8e4
This commit is contained in:
Oliver Smith
2023-05-24 10:48:07 +02:00
parent 593cd88535
commit c63c3a0cc5
7 changed files with 41 additions and 46 deletions
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14
include/osmocom/msc/codec_filter.h
View File

@@ -42,20 +42,16 @@ struct codec_filter {
/* 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_set_ran(struct codec_filter *codec_filter, enum osmo_rat_type ran_type);
void codec_filter_set_bss(struct codec_filter *codec_filter,
const struct gsm0808_speech_codec_list *codec_list_bss_supported);
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, const struct sdp_msg *remote);
int codec_filter_run(struct codec_filter *codec_filter, struct sdp_msg *result, const struct sdp_msg *remote);
int codec_filter_to_str_buf(char *buf, size_t buflen, const struct codec_filter *codec_filter,
const struct sdp_msg *result, const struct sdp_msg *remote);
char *codec_filter_to_str_c(void *ctx, const struct codec_filter *codec_filter, const struct sdp_msg *result,
const struct sdp_msg *remote);
const char *codec_filter_to_str(const struct codec_filter *codec_filter, const struct sdp_msg *result,
const struct sdp_msg *remote);
char *codec_filter_to_str_c(void *ctx, const struct codec_filter *codec_filter, const struct sdp_msg *remote);
const char *codec_filter_to_str(const struct codec_filter *codec_filter, const struct sdp_msg *remote);

3
include/osmocom/msc/transaction.h
View File

@@ -109,6 +109,9 @@ struct gsm_trans {
struct sdp_msg remote;
/* Track codec choices from BSS and remote call leg */
struct codec_filter codecs;
/* Resulting choice from codecs/bearer services and the
* local RTP address to be sent to the remote call leg. */
struct sdp_msg local;
} cc;
struct {
struct gsm411_smc_inst smc_inst;

26
src/libmsc/codec_filter.c
View File

@@ -84,19 +84,11 @@ void codec_filter_set_bss(struct codec_filter *codec_filter,
sdp_audio_codecs_from_speech_codec_list(&codec_filter->bss, codec_list_bss_supported);
}
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, const struct sdp_msg *remote)
int codec_filter_run(struct codec_filter *codec_filter, struct sdp_msg *result, const struct sdp_msg *remote)
{
struct sdp_audio_codecs *r = &codec_filter->result.audio_codecs;
struct sdp_audio_codecs *r = &result->audio_codecs;
struct sdp_audio_codec *a = &codec_filter->assignment;
*r = codec_filter->ran;
if (codec_filter->ms.count)
@@ -150,10 +142,10 @@ int codec_filter_run(struct codec_filter *codec_filter, const struct sdp_msg *re
}
int codec_filter_to_str_buf(char *buf, size_t buflen, const struct codec_filter *codec_filter,
const struct sdp_msg *remote)
const struct sdp_msg *result, const struct sdp_msg *remote)
{
struct osmo_strbuf sb = { .buf = buf, .len = buflen };
OSMO_STRBUF_APPEND(sb, sdp_msg_to_str_buf, &codec_filter->result);
OSMO_STRBUF_APPEND(sb, sdp_msg_to_str_buf, result);
OSMO_STRBUF_PRINTF(sb, " (from:");
if (sdp_audio_codec_is_set(&codec_filter->assignment)) {
@@ -188,12 +180,14 @@ int codec_filter_to_str_buf(char *buf, size_t buflen, const struct codec_filter
return sb.chars_needed;
}
char *codec_filter_to_str_c(void *ctx, const struct codec_filter *codec_filter, const struct sdp_msg *remote)
char *codec_filter_to_str_c(void *ctx, const struct codec_filter *codec_filter, const struct sdp_msg *result,
const struct sdp_msg *remote)
{
OSMO_NAME_C_IMPL(ctx, 128, "codec_filter_to_str_c-ERROR", codec_filter_to_str_buf, codec_filter, remote)
OSMO_NAME_C_IMPL(ctx, 128, "codec_filter_to_str_c-ERROR", codec_filter_to_str_buf, codec_filter, result, remote)
}
const char *codec_filter_to_str(const struct codec_filter *codec_filter, const struct sdp_msg *remote)
const char *codec_filter_to_str(const struct codec_filter *codec_filter, const struct sdp_msg *result,
const struct sdp_msg *remote)
{
return codec_filter_to_str_c(OTC_SELECT, codec_filter, remote);
return codec_filter_to_str_c(OTC_SELECT, codec_filter, result, remote);
}

21
src/libmsc/gsm_04_08_cc.c
View File

@@ -731,9 +731,9 @@ void gsm48_cc_rx_setup_cn_local_rtp_port_known(struct gsm_trans *trans)
trans_free(trans);
return;
}
codec_filter_set_local_rtp(&trans->cc.codecs, rtp_cn_local);
trans->cc.local.rtp = *rtp_cn_local;
sdp = trans->cc.codecs.result.audio_codecs.count ? &trans->cc.codecs.result : NULL;
sdp = trans->cc.local.audio_codecs.count ? &trans->cc.local : NULL;
rc = sdp_msg_to_sdp_str_buf(setup.sdp, sizeof(setup.sdp), sdp);
if (rc >= sizeof(setup.sdp)) {
LOG_TRANS(trans, LOGL_ERROR, "MNCC_SETUP_IND: SDP too long (%d > %zu bytes)\n", rc, sizeof(setup.sdp));
@@ -829,7 +829,7 @@ static int gsm48_cc_tx_setup(struct gsm_trans *trans, void *arg)
bearer_cap = (struct gsm_mncc_bearer_cap){
.speech_ver = { -1 },
};
sdp_audio_codecs_to_bearer_cap(&bearer_cap, &trans->cc.codecs.result.audio_codecs);
sdp_audio_codecs_to_bearer_cap(&bearer_cap, &trans->cc.local.audio_codecs);
rc = bearer_cap_set_radio(&bearer_cap);
if (rc) {
LOG_TRANS(trans, LOGL_ERROR, "Error composing Bearer Capability for CC Setup\n");
@@ -844,7 +844,8 @@ static int gsm48_cc_tx_setup(struct gsm_trans *trans, void *arg)
* finding a matching codec. */
if (bearer_cap.speech_ver[0] == -1) {
LOG_TRANS(trans, LOGL_ERROR, "%s: no codec match possible: %s\n",
get_mncc_name(setup->msg_type), codec_filter_to_str(&trans->cc.codecs, &trans->cc.remote));
get_mncc_name(setup->msg_type),
codec_filter_to_str(&trans->cc.codecs, &trans->cc.local, &trans->cc.remote));
/* incompatible codecs */
rc = mncc_release_ind(trans->net, trans, trans->callref,
@@ -978,7 +979,7 @@ static int gsm48_cc_mt_rtp_port_and_codec_known(struct gsm_trans *trans)
trans_free(trans);
return -EINVAL;
}
codec_filter_set_local_rtp(&trans->cc.codecs, rtp_cn_local);
trans->cc.local.rtp = *rtp_cn_local;
trans_cc_filter_run(trans);
@@ -989,7 +990,7 @@ static int gsm48_cc_mt_rtp_port_and_codec_known(struct gsm_trans *trans)
}
return mncc_recv_rtp(msc_a_net(msc_a), trans, trans->callref, MNCC_RTP_CREATE, rtp_cn_local, 0, 0,
&trans->cc.codecs.result);
&trans->cc.local);
}
static int gsm48_cc_tx_call_proc_and_assign(struct gsm_trans *trans, void *arg)
@@ -1060,7 +1061,7 @@ static int gsm48_cc_rx_alerting(struct gsm_trans *trans, struct msgb *msg)
new_cc_state(trans, GSM_CSTATE_CALL_RECEIVED);
trans_cc_filter_run(trans);
rc = sdp_msg_to_sdp_str_buf(alerting.sdp, sizeof(alerting.sdp), &trans->cc.codecs.result);
rc = sdp_msg_to_sdp_str_buf(alerting.sdp, sizeof(alerting.sdp), &trans->cc.local);
if (rc >= sizeof(alerting.sdp)) {
LOG_TRANS(trans, LOGL_ERROR, "MNCC_ALERT_IND: SDP too long (%d > %zu bytes)\n",
rc, sizeof(alerting.sdp));
@@ -1206,7 +1207,7 @@ static int gsm48_cc_rx_connect(struct gsm_trans *trans, struct msgb *msg)
rate_ctr_inc(rate_ctr_group_get_ctr(trans->net->msc_ctrs, MSC_CTR_CALL_MT_CONNECT));
trans_cc_filter_run(trans);
sdp_msg_to_sdp_str_buf(connect.sdp, sizeof(connect.sdp), &trans->cc.codecs.result);
sdp_msg_to_sdp_str_buf(connect.sdp, sizeof(connect.sdp), &trans->cc.local);
return mncc_recvmsg(trans->net, trans, MNCC_SETUP_CNF, &connect);
}
@@ -2044,7 +2045,7 @@ int gsm48_tch_rtp_create(struct gsm_trans *trans)
}
trans_cc_filter_run(trans);
codecs = &trans->cc.codecs.result.audio_codecs;
codecs = &trans->cc.local.audio_codecs;
if (!codecs->count) {
LOG_TRANS_CAT(trans, DMNCC, LOGL_ERROR,
"Cannot RTP CREATE to MNCC, there is no codec available\n");
@@ -2063,7 +2064,7 @@ int gsm48_tch_rtp_create(struct gsm_trans *trans)
}
return mncc_recv_rtp(net, trans, trans->callref, MNCC_RTP_CREATE, rtp_cn_local,
codec->payload_type, mncc_payload_msg_type, &trans->cc.codecs.result);
codec->payload_type, mncc_payload_msg_type, &trans->cc.local);
}
static int tch_rtp_connect(struct gsm_network *net, const struct gsm_mncc_rtp *rtp)

14
src/libmsc/msc_a.c
View File

@@ -640,18 +640,18 @@ static void msc_a_call_leg_ran_local_addr_available(struct msc_a *msc_a)
trans_cc_filter_run(cc_trans);
LOG_TRANS(cc_trans, LOGL_DEBUG, "Sending Assignment Command\n");
if (!cc_trans->cc.codecs.result.audio_codecs.count) {
if (!cc_trans->cc.local.audio_codecs.count) {
LOG_TRANS(cc_trans, LOGL_ERROR, "Assignment not possible, no matching codec: %s\n",
codec_filter_to_str(&cc_trans->cc.codecs, &cc_trans->cc.remote));
codec_filter_to_str(&cc_trans->cc.codecs, &cc_trans->cc.local, &cc_trans->cc.remote));
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.codecs.result.audio_codecs)) {
if (sdp_audio_codecs_to_gsm0808_channel_type(&channel_type, &cc_trans->cc.local.audio_codecs)) {
LOG_MSC_A(msc_a, LOGL_ERROR, "Cannot compose Channel Type (Permitted Speech) from codecs: %s\n",
codec_filter_to_str(&cc_trans->cc.codecs, &cc_trans->cc.remote));
codec_filter_to_str(&cc_trans->cc.codecs, &cc_trans->cc.local, &cc_trans->cc.remote));
trans_free(cc_trans);
return;
}
@@ -1455,7 +1455,7 @@ static void msc_a_up_call_assignment_complete(struct msc_a *msc_a, const struct
trans_cc_filter_run(cc_trans);
LOG_TRANS(cc_trans, LOGL_INFO, "Assignment Complete: RAN: %s, CN: %s\n",
sdp_audio_codecs_to_str(&rtps_to_ran->codecs),
sdp_audio_codecs_to_str(&cc_trans->cc.codecs.result.audio_codecs));
sdp_audio_codecs_to_str(&cc_trans->cc.local.audio_codecs));
if (cc_on_assignment_done(cc_trans)) {
/* If an error occurred, it was logged in cc_assignment_done() */
@@ -1874,13 +1874,13 @@ static int msc_a_start_assignment(struct msc_a *msc_a, struct gsm_trans *cc_tran
* 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,
&cc_trans->cc.codecs.result.audio_codecs, NULL);
&cc_trans->cc.local.audio_codecs, NULL);
if (!ran_rtp_available) {
struct sdp_audio_codecs *codecs;
if (msc_a->c.ran->force_mgw_codecs_to_ran.count)
codecs = &msc_a->c.ran->force_mgw_codecs_to_ran;
else
codecs = &cc_trans->cc.codecs.result.audio_codecs;
codecs = &cc_trans->cc.local.audio_codecs;
return call_leg_ensure_ci(cl, RTP_TO_RAN, cc_trans->callref, cc_trans, codecs, NULL);
}

4
src/libmsc/msc_ho.c
View File

@@ -416,14 +416,14 @@ static void msc_ho_send_handover_request(struct msc_a *msc_a)
if (cc_trans) {
if (sdp_audio_codecs_to_gsm0808_channel_type(&channel_type,
&cc_trans->cc.codecs.result.audio_codecs)) {
&cc_trans->cc.local.audio_codecs)) {
msc_ho_failed(msc_a, GSM0808_CAUSE_EQUIPMENT_FAILURE,
"Failed to determine Channel Type for Handover Request message\n");
return;
}
ran_enc_msg.handover_request.geran.channel_type = &channel_type;
sdp_audio_codecs_to_speech_codec_list(&scl, &cc_trans->cc.codecs.result.audio_codecs);
sdp_audio_codecs_to_speech_codec_list(&scl, &cc_trans->cc.local.audio_codecs);
if (!scl.len) {
msc_ho_failed(msc_a, GSM0808_CAUSE_EQUIPMENT_FAILURE, "Failed to compose"
" Codec List (MSC Preferred) for Handover Request message\n");

5
src/libmsc/transaction_cc.c
View File

@@ -42,8 +42,9 @@ void trans_cc_filter_set_bss(struct gsm_trans *trans, struct msc_a *msc_a)
void trans_cc_filter_run(struct gsm_trans *trans)
{
codec_filter_run(&trans->cc.codecs, &trans->cc.remote);
LOG_TRANS(trans, LOGL_DEBUG, "codecs: %s\n", codec_filter_to_str(&trans->cc.codecs, &trans->cc.remote));
codec_filter_run(&trans->cc.codecs, &trans->cc.local, &trans->cc.remote);
LOG_TRANS(trans, LOGL_DEBUG, "codecs: %s\n",
codec_filter_to_str(&trans->cc.codecs, &trans->cc.local, &trans->cc.remote));
}
void trans_cc_filter_set_ms_from_bc(struct gsm_trans *trans, const struct gsm_mncc_bearer_cap *bcap)