diff options
author | Philipp Maier <pmaier@sysmocom.de> | 2018-08-07 13:00:14 +0200 |
---|---|---|
committer | Harald Welte <laforge@gnumonks.org> | 2018-08-07 16:51:30 +0000 |
commit | 8ad3dacebb2ec1c160c2cc3b0ec0a8eaec4332f0 (patch) | |
tree | ae024def16eb5b6ccf50bf8cb96a967abfdd81b5 /src | |
parent | 6cc377d359a938e59dc517a99d0e182315cdf669 (diff) |
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
Diffstat (limited to 'src')
-rw-r--r-- | src/libmsc/a_iface_bssap.c | 48 | ||||
-rw-r--r-- | src/libmsc/gsm_04_08_cc.c | 19 | ||||
-rw-r--r-- | src/libmsc/msc_mgcp.c | 20 |
3 files changed, 73 insertions, 14 deletions
diff --git a/src/libmsc/a_iface_bssap.c b/src/libmsc/a_iface_bssap.c index 1ace43dae..11d367383 100644 --- a/src/libmsc/a_iface_bssap.c +++ b/src/libmsc/a_iface_bssap.c @@ -502,11 +502,50 @@ static int bssmap_rx_sapi_n_rej(struct gsm_subscriber_connection *conn, struct m 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 @@ static int bssmap_rx_ass_compl(struct gsm_subscriber_connection *conn, struct ms 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 8becd0553..2c17e22f5 100644 --- a/src/libmsc/gsm_04_08_cc.c +++ b/src/libmsc/gsm_04_08_cc.c @@ -319,6 +319,15 @@ static int tch_bridge(struct gsm_network *net, struct gsm_mncc_bridge *bridge) /* 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 @@ static int tch_rtp_connect(struct gsm_network *net, void *arg) 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 f5bdeb7aa..e58b24903 100644 --- a/src/libmsc/msc_mgcp.c +++ b/src/libmsc/msc_mgcp.c @@ -277,22 +277,16 @@ static void fsm_crcx_ran_cb(struct osmo_fsm_inst *fi, uint32_t event, void *data 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 @@ static void fsm_crcx_cn_cb(struct osmo_fsm_inst *fi, uint32_t event, void *data) 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 @@ static void fsm_mdcx_cn_cb(struct osmo_fsm_inst *fi, uint32_t event, void *data) .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 @@ static void fsm_mdcx_ran_cb(struct osmo_fsm_inst *fi, uint32_t event, void *data .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) { |