osmith has uploaded this change for review. (
https://gerrit.osmocom.org/c/osmo-msc/+/33366 )
Change subject: transaction: move cc.codecs.result -> cc.local
......................................................................
transaction: move cc.codecs.result -> cc.local
Prepare for CSD where this will be used too.
Related: OS#4394
Change-Id: Iaf954be0455625faa06a64c19905b79b7045f8e4
---
M include/osmocom/msc/codec_filter.h
M include/osmocom/msc/transaction.h
M src/libmsc/codec_filter.c
M src/libmsc/gsm_04_08_cc.c
M src/libmsc/msc_a.c
M src/libmsc/msc_ho.c
M src/libmsc/transaction_cc.c
7 files changed, 53 insertions(+), 46 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmo-msc refs/changes/66/33366/1
diff --git a/include/osmocom/msc/codec_filter.h
b/include/osmocom/msc/codec_filter.h
index c0d8f32..da4a67e 100644
--- a/include/osmocom/msc/codec_filter.h
+++ b/include/osmocom/msc/codec_filter.h
@@ -42,20 +42,16 @@
/* 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);
-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);
+const char *codec_filter_to_str(const struct codec_filter *codec_filter, const
struct sdp_msg *result,
+ const struct sdp_msg *remote);
diff --git a/include/osmocom/msc/transaction.h
b/include/osmocom/msc/transaction.h
index bfb32ad..36aab78 100644
--- a/include/osmocom/msc/transaction.h
+++ b/include/osmocom/msc/transaction.h
@@ -109,6 +109,9 @@
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;
diff --git a/src/libmsc/codec_filter.c b/src/libmsc/codec_filter.c
index 38a1246..a9d93a7 100644
--- a/src/libmsc/codec_filter.c
+++ b/src/libmsc/codec_filter.c
@@ -84,19 +84,11 @@
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_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 @@
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);
}
diff --git a/src/libmsc/gsm_04_08_cc.c b/src/libmsc/gsm_04_08_cc.c
index 70a0b77..77090ca 100644
--- a/src/libmsc/gsm_04_08_cc.c
+++ b/src/libmsc/gsm_04_08_cc.c
@@ -731,9 +731,9 @@
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 @@
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 @@
* 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 @@
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 @@
}
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 @@
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 @@
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 @@
}
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 @@
}
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)
diff --git a/src/libmsc/msc_a.c b/src/libmsc/msc_a.c
index b8a5de1..ca38206 100644
--- a/src/libmsc/msc_a.c
+++ b/src/libmsc/msc_a.c
@@ -640,18 +640,18 @@
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 @@
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 @@
* 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);
}
diff --git a/src/libmsc/msc_ho.c b/src/libmsc/msc_ho.c
index 7467287..2a5891a 100644
--- a/src/libmsc/msc_ho.c
+++ b/src/libmsc/msc_ho.c
@@ -416,14 +416,14 @@
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");
diff --git a/src/libmsc/transaction_cc.c b/src/libmsc/transaction_cc.c
index 4c30f84..d221d7c 100644
--- a/src/libmsc/transaction_cc.c
+++ b/src/libmsc/transaction_cc.c
@@ -42,8 +42,9 @@
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)
--
To view, visit https://gerrit.osmocom.org/c/osmo-msc/+/33366
To unsubscribe, or for help writing mail filters, visit
https://gerrit.osmocom.org/settings
Gerrit-Project: osmo-msc
Gerrit-Branch: master
Gerrit-Change-Id: Iaf954be0455625faa06a64c19905b79b7045f8e4
Gerrit-Change-Number: 33366
Gerrit-PatchSet: 1
Gerrit-Owner: osmith <[email protected]>
Gerrit-MessageType: newchange
