neels has submitted this change. ( https://gerrit.osmocom.org/c/osmo-msc/+/31696 ) ( 3 is the latest approved patch-set. No files were changed between the latest approved patch-set and the submitted one. )Change subject: in ran_msg, return gsm0808_speech_codec (inter-MSC) ...................................................................... in ran_msg, return gsm0808_speech_codec (inter-MSC) Get rid of enum mgcp_codecs in inter-MSC handover related code. Change-Id: I9c649f98738a55b8637ae600d5cdf81099fd08e5 --- M include/osmocom/msc/msc_ho.h M include/osmocom/msc/ran_msg.h M src/libmsc/msc_ho.c M src/libmsc/msc_t.c M src/libmsc/ran_msg_a.c 5 files changed, 45 insertions(+), 25 deletions(-) Approvals: pespin: Looks good to me, but someone else must approve laforge: Looks good to me, approved Jenkins Builder: Verified diff --git a/include/osmocom/msc/msc_ho.h b/include/osmocom/msc/msc_ho.h index a3f60c7..ca5618e 100644 --- a/include/osmocom/msc/msc_ho.h +++ b/include/osmocom/msc/msc_ho.h @@ -81,7 +81,7 @@ struct osmo_sockaddr_str ran_remote_rtp; /* The codec from Handover Request Acknowledge. */ bool codec_present; - enum mgcp_codecs codec; + struct gsm0808_speech_codec codec; /* Inter-MSC voice forwarding via MNCC, to the remote MSC. The Prepare Handover Response sent us the * Handover Number the remote MSC assigned. This is a call to that Handover Number, via PBX. diff --git a/include/osmocom/msc/ran_msg.h b/include/osmocom/msc/ran_msg.h index d6b1e75..3e0ac60 100644 --- a/include/osmocom/msc/ran_msg.h +++ b/include/osmocom/msc/ran_msg.h @@ -162,7 +162,8 @@ struct osmo_sockaddr_str remote_rtp; bool codec_present; - enum mgcp_codecs codec; + struct gsm0808_speech_codec codec; + bool codec_with_iuup; }; struct ran_handover_command { diff --git a/src/libmsc/msc_ho.c b/src/libmsc/msc_ho.c index d53bb9e..e11abcf 100644 --- a/src/libmsc/msc_ho.c +++ b/src/libmsc/msc_ho.c @@ -681,7 +681,7 @@ msc_a->ho.new_cell.codec = hra->ran_dec->handover_request_ack.codec; if (hra->ran_dec->handover_request_ack.codec_present) { LOG_HO(msc_a, LOGL_DEBUG, "Request Ack contains codec %s\n", - osmo_mgcpc_codec_name(msc_a->ho.new_cell.codec)); + gsm0808_speech_codec_type_name(msc_a->ho.new_cell.codec.type)); } } @@ -728,13 +728,14 @@ /* Switch over to the new peer */ rtp_stream_set_remote_addr(rtp_to_ran, &msc_a->ho.new_cell.ran_remote_rtp); if (msc_a->ho.new_cell.codec_present) { - struct sdp_audio_codecs codecs = {}; - if (!sdp_audio_codecs_add_mgcp_codec(&codecs, msc_a->ho.new_cell.codec)) { - LOG_HO(msc_a, LOGL_ERROR, - "Cannot resolve codec: %s\n", osmo_mgcpc_codec_name(msc_a->ho.new_cell.codec)); - } else { - rtp_stream_set_codecs(rtp_to_ran, &codecs); - } + const struct codec_mapping *m; + m = codec_mapping_by_gsm0808_speech_codec_type(msc_a->ho.new_cell.codec.type); + /* TODO: use codec_mapping_by_gsm0808_speech_codec() to also match on codec.cfg */ + if (!m) + LOG_HO(msc_a, LOGL_ERROR, "Cannot resolve codec: %s\n", + gsm0808_speech_codec_type_name(msc_a->ho.new_cell.codec.type)); + else + rtp_stream_set_one_codec(rtp_to_ran, &m->sdp); } else { LOG_HO(msc_a, LOGL_ERROR, "No codec is set\n"); } diff --git a/src/libmsc/msc_t.c b/src/libmsc/msc_t.c index 787d736..7f316a7 100644 --- a/src/libmsc/msc_t.c +++ b/src/libmsc/msc_t.c @@ -38,6 +38,7 @@ #include <osmocom/msc/vlr.h> #include <osmocom/msc/msc_i.h> #include <osmocom/msc/gsm_data.h> +#include <osmocom/msc/codec_mapping.h> static struct osmo_fsm msc_t_fsm; @@ -448,11 +449,20 @@ LOG_MSC_T(msc_t, LOGL_DEBUG, "No RTP IP:port in Handover Request Ack\n"); } if (r->codec_present) { - LOG_MSC_T(msc_t, LOGL_DEBUG, "From Handover Request Ack, got %s\n", - osmo_mgcpc_codec_name(r->codec)); - rtp_stream_set_codecs_from_mgcp_codec(rtp_ran, r->codec); - if (rtp_cn) - rtp_stream_set_codecs_from_mgcp_codec(rtp_cn, r->codec); + const struct codec_mapping *m = codec_mapping_by_gsm0808_speech_codec_type(r->codec.type); + /* TODO: use codec_mapping_by_gsm0808_speech_codec() to also match on codec.cfg */ + if (!m) { + LOG_MSC_T(msc_t, LOGL_ERROR, "Cannot resolve codec in Handover Request Ack: %s / %s\n", + gsm0808_speech_codec_type_name(r->codec.type), + m ? sdp_audio_codec_to_str(&m->sdp) : "(unknown)"); + } else { + LOG_MSC_T(msc_t, LOGL_DEBUG, "From Handover Request Ack, got codec %s / %s\n", + gsm0808_speech_codec_type_name(r->codec.type), + sdp_audio_codec_to_str(&m->sdp)); + rtp_stream_set_one_codec(rtp_ran, &m->sdp); + if (rtp_cn) + rtp_stream_set_one_codec(rtp_cn, &m->sdp); + } } else { LOG_MSC_T(msc_t, LOGL_DEBUG, "No codec in Handover Request Ack\n"); } diff --git a/src/libmsc/ran_msg_a.c b/src/libmsc/ran_msg_a.c index 56252f1..19402bc 100644 --- a/src/libmsc/ran_msg_a.c +++ b/src/libmsc/ran_msg_a.c @@ -671,13 +671,12 @@ } if (ie_chosen_speech_version) { - struct gsm0808_speech_codec sc; ran_dec_msg.handover_request_ack.chosen_speech_version = ie_chosen_speech_version->val[0]; /* the codec may be extrapolated from this Speech Version or below from Speech Codec */ - gsm0808_speech_codec_from_chan_type(&sc, ran_dec_msg.handover_request_ack.chosen_speech_version); - ran_dec_msg.handover_request_ack.codec_present = true; - ran_dec_msg.handover_request_ack.codec = ran_a_mgcp_codec_from_sc(&sc); + if (gsm0808_speech_codec_from_chan_type(&ran_dec_msg.handover_request_ack.codec, + ran_dec_msg.handover_request_ack.chosen_speech_version) == 0) + ran_dec_msg.handover_request_ack.codec_present = true; } if (ie_aoip_transp_addr) { @@ -692,14 +691,12 @@ } if (ie_speech_codec) { - struct gsm0808_speech_codec sc; - if (gsm0808_dec_speech_codec(&sc, ie_speech_codec->val, ie_speech_codec->len) < 0) + /* the codec may be extrapolated from above Speech Version or from this Speech Codec */ + if (gsm0808_dec_speech_codec(&ran_dec_msg.handover_request_ack.codec, + ie_speech_codec->val, ie_speech_codec->len) < 0) LOG_RAN_A_DEC_MSG(LOGL_ERROR, "unable to decode IE Speech Codec (Chosen)\n"); - else { - /* the codec may be extrapolated from above Speech Version or from this Speech Codec */ + else ran_dec_msg.handover_request_ack.codec_present = true; - ran_dec_msg.handover_request_ack.codec = ran_a_mgcp_codec_from_sc(&sc); - } } return ran_decoded(ran_dec, &ran_dec_msg); -- To view, visit https://gerrit.osmocom.org/c/osmo-msc/+/31696 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmo-msc Gerrit-Branch: master Gerrit-Change-Id: I9c649f98738a55b8637ae600d5cdf81099fd08e5 Gerrit-Change-Number: 31696 Gerrit-PatchSet: 8 Gerrit-Owner: neels <nhofmeyr@sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: laforge <laforge@osmocom.org> Gerrit-Reviewer: neels <nhofmeyr@sysmocom.de> Gerrit-Reviewer: pespin <pespin@sysmocom.de> Gerrit-MessageType: merged