mgcp: use codec information returned with ASSIGNMENT COMPL.
When the assignment completes a choosen codec is returned. At the
moment we do not use this information.
- add struct members for codec info (both, RAN and CN)
- parse codec info in BSSMAP ASSIGNMENT COMPLETE
- use codec info on mgcp
Since the MNCC API is not complete yet, we currently only use the
codec info only on the internal MNCC yet.
Change-Id: I9d5b1cd016d9a058b22a367d0e5e9f2ef447931a
Related: OS#2728
diff --git a/src/libmsc/a_iface_bssap.c b/src/libmsc/a_iface_bssap.c
index 1ace43d..11d3673 100644
--- a/src/libmsc/a_iface_bssap.c
+++ b/src/libmsc/a_iface_bssap.c
@@ -502,11 +502,50 @@
return 0;
}
+/* Use the speech codec info we go with the assignment complete to dtermine
+ * which codec we will signal to the MGW */
+static enum mgcp_codecs mgcp_codec_from_sc(struct gsm0808_speech_codec *sc)
+{
+ switch (sc->type) {
+ case GSM0808_SCT_FR1:
+ return CODEC_GSM_8000_1;
+ break;
+ case GSM0808_SCT_FR2:
+ return CODEC_GSMEFR_8000_1;
+ break;
+ case GSM0808_SCT_FR3:
+ return CODEC_AMR_8000_1;
+ break;
+ case GSM0808_SCT_FR4:
+ return CODEC_AMRWB_16000_1;
+ break;
+ case GSM0808_SCT_FR5:
+ return CODEC_AMRWB_16000_1;
+ break;
+ case GSM0808_SCT_HR1:
+ return CODEC_GSMHR_8000_1;
+ break;
+ case GSM0808_SCT_HR3:
+ return CODEC_AMR_8000_1;
+ break;
+ case GSM0808_SCT_HR4:
+ return CODEC_AMRWB_16000_1;
+ break;
+ case GSM0808_SCT_HR6:
+ return CODEC_AMRWB_16000_1;
+ break;
+ default:
+ return CODEC_PCMU_8000_1;
+ break;
+ }
+}
+
/* Endpoint to handle assignment complete */
static int bssmap_rx_ass_compl(struct gsm_subscriber_connection *conn, struct msgb *msg,
struct tlv_parsed *tp)
{
struct sockaddr_storage rtp_addr;
+ struct gsm0808_speech_codec sc;
struct sockaddr_in *rtp_addr_in;
int rc;
@@ -525,6 +564,15 @@
return -EINVAL;
}
+ /* Decode speech codec (choosen) element */
+ rc = gsm0808_dec_speech_codec(&sc, TLVP_VAL(tp, GSM0808_IE_SPEECH_CODEC),
+ TLVP_LEN(tp, GSM0808_IE_SPEECH_CODEC));
+ if (rc < 0) {
+ LOGPCONN(conn, LOGL_ERROR, "Unable to decode speech codec (choosen).\n");
+ return -EINVAL;
+ }
+ conn->rtp.codec_ran = mgcp_codec_from_sc(&sc);
+
/* use address / port supplied with the AoIP
* transport address element */
if (rtp_addr.ss_family == AF_INET) {
diff --git a/src/libmsc/gsm_04_08_cc.c b/src/libmsc/gsm_04_08_cc.c
index 8becd05..2c17e22 100644
--- a/src/libmsc/gsm_04_08_cc.c
+++ b/src/libmsc/gsm_04_08_cc.c
@@ -319,6 +319,15 @@
/* Which subscriber do we want to track trans1 or trans2? */
log_set_context(LOG_CTX_VLR_SUBSCR, trans1->vsub);
+ /* This call briding mechanism is only used with the internal MNCC.
+ * functionality (with external MNCC briding would be done by the PBX)
+ * This means 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. */
+ trans1->conn->rtp.codec_cn = trans1->conn->rtp.codec_ran;
+ trans2->conn->rtp.codec_cn = trans1->conn->rtp.codec_ran;
+
/* Bridge RTP streams */
rc = msc_mgcp_call_complete(trans1, trans2->conn->rtp.local_port_cn,
trans2->conn->rtp.local_addr_cn);
@@ -1716,6 +1725,16 @@
struct gsm_mncc_rtp *rtp = arg;
struct in_addr 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
+ * meaningful value based on this information but unfortunately we
+ * can't do that yet because the mncc API can not signal dynamic
+ * payload types yet. This must be fixed first. Also there may be
+ * additional members necessary in trans->conn->rtp because we
+ * somehow need to deal with dynamic payload types that do not
+ * comply to 3gpp's assumptions of payload type numbers on the A
+ * interface. See also related tickets: OS#3399 and OS1683 */
+
/* Find callref */
trans = trans_find_by_callref(net, rtp->callref);
if (!trans) {
diff --git a/src/libmsc/msc_mgcp.c b/src/libmsc/msc_mgcp.c
index f5bdeb7..e58b249 100644
--- a/src/libmsc/msc_mgcp.c
+++ b/src/libmsc/msc_mgcp.c
@@ -277,22 +277,16 @@
struct mgcp_msg mgcp_msg;
struct msgb *msg;
int rc;
-
-#ifdef BUILD_IU
struct gsm_trans *trans;
struct gsm_subscriber_connection *conn;
-#endif
OSMO_ASSERT(mgcp_ctx);
mgcp = mgcp_ctx->mgcp;
OSMO_ASSERT(mgcp);
-
-#ifdef BUILD_IU
trans = mgcp_ctx->trans;
OSMO_ASSERT(trans);
conn = trans->conn;
OSMO_ASSERT(conn);
-#endif
/* NOTE: In case of error, we will not be able to perform any DLCX
* operation because until this point we do not have requested any
@@ -396,22 +390,16 @@
struct mgcp_msg mgcp_msg;
struct msgb *msg;
int rc;
-
-#ifdef BUILD_IU
struct gsm_trans *trans;
struct gsm_subscriber_connection *conn;
-#endif
OSMO_ASSERT(mgcp_ctx);
mgcp = mgcp_ctx->mgcp;
OSMO_ASSERT(mgcp);
-
-#ifdef BUILD_IU
trans = mgcp_ctx->trans;
OSMO_ASSERT(trans);
conn = trans->conn;
OSMO_ASSERT(conn);
-#endif
switch (event) {
case EV_CRCX_RAN_RESP:
@@ -593,7 +581,9 @@
.conn_id = mgcp_ctx->conn_id_cn,
.conn_mode = MGCP_CONN_RECV_SEND,
.audio_ip = conn->rtp.remote_addr_cn,
- .audio_port = conn->rtp.remote_port_cn
+ .audio_port = conn->rtp.remote_port_cn,
+ .codecs[0] = conn->rtp.codec_cn,
+ .codecs_len = 1
};
if (osmo_strlcpy(mgcp_msg.endpoint, mgcp_ctx->rtp_endpoint, sizeof(mgcp_msg.endpoint)) >=
MGCP_ENDPOINT_MAXLEN) {
@@ -710,7 +700,9 @@
.conn_id = mgcp_ctx->conn_id_ran,
.conn_mode = MGCP_CONN_RECV_SEND,
.audio_ip = conn->rtp.remote_addr_ran,
- .audio_port = conn->rtp.remote_port_ran
+ .audio_port = conn->rtp.remote_port_ran,
+ .codecs[0] = conn->rtp.codec_ran,
+ .codecs_len = 1
};
if (osmo_strlcpy(mgcp_msg.endpoint, mgcp_ctx->rtp_endpoint, sizeof(mgcp_msg.endpoint)) >=
MGCP_ENDPOINT_MAXLEN) {