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Harald Welte gerrit-no-reply at lists.osmocom.orgHarald Welte has submitted this change and it was merged. ( https://gerrit.osmocom.org/10367 ) Change subject: mgcp: use codec information returned with ASSIGNMENT COMPL. ...................................................................... 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 --- M include/osmocom/msc/gsm_data.h M src/libmsc/a_iface_bssap.c M src/libmsc/gsm_04_08_cc.c M src/libmsc/msc_mgcp.c 4 files changed, 76 insertions(+), 14 deletions(-) Approvals: Harald Welte: Looks good to me, approved Jenkins Builder: Verified diff --git a/include/osmocom/msc/gsm_data.h b/include/osmocom/msc/gsm_data.h index c79adde..27f7fc5 100644 --- a/include/osmocom/msc/gsm_data.h +++ b/include/osmocom/msc/gsm_data.h @@ -143,10 +143,13 @@ char local_addr_ran[INET_ADDRSTRLEN]; uint16_t remote_port_ran; char remote_addr_ran[INET_ADDRSTRLEN]; + enum mgcp_codecs codec_ran; + uint16_t local_port_cn; char local_addr_cn[INET_ADDRSTRLEN]; uint16_t remote_port_cn; char remote_addr_cn[INET_ADDRSTRLEN]; + enum mgcp_codecs codec_cn; } rtp; /* which Iu-CS connection, if any. */ 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) { -- To view, visit https://gerrit.osmocom.org/10367 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmo-msc Gerrit-Branch: master Gerrit-MessageType: merged Gerrit-Change-Id: I9d5b1cd016d9a058b22a367d0e5e9f2ef447931a Gerrit-Change-Number: 10367 Gerrit-PatchSet: 2 Gerrit-Owner: dexter <pmaier at sysmocom.de> Gerrit-Reviewer: Harald Welte <laforge at gnumonks.org> Gerrit-Reviewer: Jenkins Builder -------------- next part -------------- An HTML attachment was scrubbed... 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