neels has submitted this change. ( https://gerrit.osmocom.org/c/osmo-msc/+/30115 )

Change subject: MNCC: use codec_mapping, drop mgcp_codec_to_mncc_payload_msg_type() ......................................................................

MNCC: use codec_mapping, drop mgcp_codec_to_mncc_payload_msg_type()

Change-Id: I8995ef43b9f79bc1db5672362c6433e4d96dd9e0 --- M src/libmsc/gsm_04_08_cc.c M src/libmsc/mncc_call.c 2 files changed, 29 insertions(+), 22 deletions(-)

Approvals: laforge: Looks good to me, approved Jenkins Builder: Verified

diff --git a/src/libmsc/gsm_04_08_cc.c b/src/libmsc/gsm_04_08_cc.c index 5715429..685bf60 100644 --- a/src/libmsc/gsm_04_08_cc.c +++ b/src/libmsc/gsm_04_08_cc.c @@ -56,6 +56,7 @@ #include <osmocom/msc/mncc_call.h> #include <osmocom/msc/msc_t.h> #include <osmocom/msc/sdp_msg.h> +#include <osmocom/msc/codec_mapping.h>

#include <osmocom/gsm/gsm48.h> #include <osmocom/gsm/gsm0480.h> @@ -1770,6 +1771,7 @@ uint32_t payload_type; int payload_msg_type; const struct mgcp_conn_peer *mgcp_info; + const struct codec_mapping *m;

if (!rtp_cn) { LOG_TRANS_CAT(trans, DMNCC, LOGL_ERROR, "Cannot RTP CREATE to MNCC, no RTP set up for the CN side\n"); @@ -1783,7 +1785,14 @@ }

/* Codec */ - payload_msg_type = mgcp_codec_to_mncc_payload_msg_type(rtp_cn->codec); + m = codec_mapping_by_mgcp_codec(rtp_cn->codec); + if (!m) { + LOG_TRANS_CAT(trans, DMNCC, LOGL_ERROR, + "Cannot RTP CREATE to MNCC, cannot resolve codec '%s'\n", + osmo_mgcpc_codec_name(rtp_cn->codec)); + return -EINVAL; + } + payload_msg_type = m->mncc_payload_msg_type;

/* Payload Type number */ mgcp_info = osmo_mgcpc_ep_ci_get_rtp_info(rtp_cn->ci); diff --git a/src/libmsc/mncc_call.c b/src/libmsc/mncc_call.c index c9a6d56..fbf96f3 100644 --- a/src/libmsc/mncc_call.c +++ b/src/libmsc/mncc_call.c @@ -35,6 +35,7 @@ #include <osmocom/msc/rtp_stream.h> #include <osmocom/msc/msub.h> #include <osmocom/msc/vlr.h> +#include <osmocom/msc/codec_mapping.h>

struct osmo_fsm mncc_call_fsm; static bool mncc_call_tx_rtp_create(struct mncc_call *mncc_call); @@ -274,25 +275,6 @@ return mncc_call_tx_rtp_create(mncc_call); }

-/* Convert enum mgcp_codecs to an gsm_mncc_rtp->payload_msg_type value. */ -uint32_t mgcp_codec_to_mncc_payload_msg_type(enum mgcp_codecs codec) -{ - switch (codec) { - default: - /* disclaimer: i have no idea what i'm doing. */ - case CODEC_GSM_8000_1: - return GSM_TCHF_FRAME; - case CODEC_GSMEFR_8000_1: - return GSM_TCHF_FRAME_EFR; - case CODEC_GSMHR_8000_1: - return GSM_TCHH_FRAME; - case CODEC_AMR_8000_1: - case CODEC_AMRWB_16000_1: - //return GSM_TCHF_FRAME; - return GSM_TCH_FRAME_AMR; - } -} - static bool mncc_call_tx_rtp_create(struct mncc_call *mncc_call) { if (!mncc_call->rtps || !osmo_sockaddr_str_is_nonzero(&mncc_call->rtps->local)) { @@ -314,8 +296,15 @@ }

if (mncc_call->rtps->codec_known) { - mncc_msg.rtp.payload_type = 0; /* ??? */ - mncc_msg.rtp.payload_msg_type = mgcp_codec_to_mncc_payload_msg_type(mncc_call->rtps->codec); + const struct codec_mapping *m = codec_mapping_by_mgcp_codec(mncc_call->rtps->codec); + + if (!m) { + mncc_call_error(mncc_call, "Failed to resolve audio codec '%s'\n", + osmo_mgcpc_codec_name(mncc_call->rtps->codec)); + return false; + } + mncc_msg.rtp.payload_type = m->sdp.payload_type; + mncc_msg.rtp.payload_msg_type = m->mncc_payload_msg_type; }

if (mncc_call_tx(mncc_call, &mncc_msg))