November 2023 - gerrit-log - lists.osmocom.org

[S] Change in osmo-msc[master]: add sdp_audio_codec_to_speech_codec_list()

by neels

neels has uploaded this change for review. ( https://gerrit.osmocom.org/c/osmo-msc/+/35050?usp=email ) Change subject: add sdp_audio_codec_to_speech_codec_list() ...................................................................... add sdp_audio_codec_to_speech_codec_list() Used by I8760feaa8598047369ef8c3ab2673013bac8ac8a to add just a single codec to a speech codec list, instead of a list. Change-Id: I6ac23c54bc26939e048ff2df06eb987421cfb1c5 --- M include/osmocom/msc/codec_mapping.h M src/libmsc/codec_mapping.c 2 files changed, 33 insertions(+), 9 deletions(-) git pull ssh://gerrit.osmocom.org:29418/osmo-msc refs/changes/50/35050/1 diff --git a/include/osmocom/msc/codec_mapping.h b/include/osmocom/msc/codec_mapping.h index 1232d2d..3b502a9 100644 --- a/include/osmocom/msc/codec_mapping.h +++ b/include/osmocom/msc/codec_mapping.h @@ -56,6 +56,7 @@ struct sdp_audio_codec *sdp_audio_codecs_add_mgcp_codec(struct sdp_audio_codecs *ac, enum mgcp_codecs mgcp_codec); void sdp_audio_codecs_from_bearer_cap(struct sdp_audio_codecs *ac, const struct gsm_mncc_bearer_cap *bc); +int sdp_audio_codec_to_speech_codec_list(struct gsm0808_speech_codec_list *scl, const struct sdp_audio_codec *codec); void sdp_audio_codecs_to_speech_codec_list(struct gsm0808_speech_codec_list *cl, const struct sdp_audio_codecs *ac); void sdp_audio_codecs_from_speech_codec_list(struct sdp_audio_codecs *ac, const struct gsm0808_speech_codec_list *cl); diff --git a/src/libmsc/codec_mapping.c b/src/libmsc/codec_mapping.c index 114abaa..4063514 100644 --- a/src/libmsc/codec_mapping.c +++ b/src/libmsc/codec_mapping.c @@ -414,6 +414,24 @@ } } +/* Append an entry for the given sdp_audio_codec to the gsm0808_speech_codec_list. + * Return 0 if an entry was added, -ENOENT when there is no mapping to gsm0808_speech_codec for the given + * sdp_audio_codec, and -ENOSPC when scl is full and nothing could be added. */ +int sdp_audio_codec_to_speech_codec_list(struct gsm0808_speech_codec_list *scl, const struct sdp_audio_codec *codec) +{ + const struct codec_mapping *m = codec_mapping_by_subtype_name(codec->subtype_name); + if (!m) + return -ENOENT; + if (!m->has_gsm0808_speech_codec) + return -ENOENT; + if (scl->len >= ARRAY_SIZE(scl->codec)) + return -ENOSPC; + scl->codec[scl->len] = m->gsm0808_speech_codec; + /* FIXME: apply AMR configuration according to codec->fmtp */ + scl->len++; + return 0; +} + void sdp_audio_codecs_to_speech_codec_list(struct gsm0808_speech_codec_list *scl, const struct sdp_audio_codecs *ac) { const struct sdp_audio_codec *codec; @@ -421,16 +439,9 @@ *scl = (struct gsm0808_speech_codec_list){}; foreach_sdp_audio_codec(codec, ac) { - const struct codec_mapping *m = codec_mapping_by_subtype_name(codec->subtype_name); - if (!m) - continue; - if (!m->has_gsm0808_speech_codec) - continue; - if (scl->len >= ARRAY_SIZE(scl->codec)) + int rc = sdp_audio_codec_to_speech_codec_list(scl, codec); + if (rc == -ENOSPC) break; - scl->codec[scl->len] = m->gsm0808_speech_codec; - /* FIXME: apply AMR configuration according to codec->fmtp */ - scl->len++; } } -- To view, visit https://gerrit.osmocom.org/c/osmo-msc/+/35050?usp=email To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmo-msc Gerrit-Branch: master Gerrit-Change-Id: I6ac23c54bc26939e048ff2df06eb987421cfb1c5 Gerrit-Change-Number: 35050 Gerrit-PatchSet: 1 Gerrit-Owner: neels <nhofmeyr(a)sysmocom.de> Gerrit-MessageType: newchange

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[M] Change in osmo-msc[master]: implement re-assignment to match codecs

by neels

neels has uploaded this change for review. ( https://gerrit.osmocom.org/c/osmo-msc/+/35051?usp=email ) Change subject: implement re-assignment to match codecs ...................................................................... implement re-assignment to match codecs This is the last missing piece that allows osmo-msc to make good TFO codecs choices. Since the codec_filter, osmo-msc properly gathers codec options and limitations. But the MO call leg still assigns a voice channel before getting a response from the MT call leg, and is then stuck with that. Add the capability to adjust the MO call leg's codec in case the MT side needs a different codec for TFO. This is only relevant for 2G; on 3G we always have AMR/IuUP. Related: OS#6258 Related: osmo-ttcn3-hacks I402ed0523a2a87b83f29c5577b2c828102005d53 Change-Id: I8760feaa8598047369ef8c3ab2673013bac8ac8a --- M include/osmocom/msc/msc_a.h M src/libmsc/call_leg.c M src/libmsc/codec_filter.c M src/libmsc/gsm_04_08_cc.c M src/libmsc/msc_a.c M src/libmsc/msc_ho.c M tests/msc_vlr/msc_vlr_test_call.c M tests/msc_vlr/msc_vlr_test_call.err 8 files changed, 106 insertions(+), 44 deletions(-) git pull ssh://gerrit.osmocom.org:29418/osmo-msc refs/changes/51/35051/1 diff --git a/include/osmocom/msc/msc_a.h b/include/osmocom/msc/msc_a.h index 0276d62..5e1672a 100644 --- a/include/osmocom/msc/msc_a.h +++ b/include/osmocom/msc/msc_a.h @@ -216,6 +216,7 @@ int msc_a_ensure_cn_local_rtp(struct msc_a *msc_a, struct gsm_trans *cc_trans); int msc_a_try_call_assignment(struct gsm_trans *cc_trans); +void msc_a_tx_assignment(struct msc_a *msc_a); const char *msc_a_cm_service_type_to_use(struct msc_a *msc_a, enum osmo_cm_service_type cm_service_type); diff --git a/src/libmsc/call_leg.c b/src/libmsc/call_leg.c index b797322..5720417 100644 --- a/src/libmsc/call_leg.c +++ b/src/libmsc/call_leg.c @@ -158,7 +158,8 @@ } if (!established) break; - call_leg_state_chg(cl, CALL_LEG_ST_ESTABLISHED); + if (cl->fi->state != CALL_LEG_ST_ESTABLISHED) + call_leg_state_chg(cl, CALL_LEG_ST_ESTABLISHED); break; case CALL_LEG_EV_RTP_STREAM_ADDR_AVAILABLE: diff --git a/src/libmsc/codec_filter.c b/src/libmsc/codec_filter.c index a9d93a7..1a163ae 100644 --- a/src/libmsc/codec_filter.c +++ b/src/libmsc/codec_filter.c @@ -98,7 +98,7 @@ if (remote->audio_codecs.count) sdp_audio_codecs_intersection(r, &remote->audio_codecs, true); -#if 0 +#if 1 /* Future: If osmo-msc were able to trigger a re-assignment after the remote side has picked a codec mismatching * the initial Assignment, then this code here would make sense: keep the other codecs as available to choose * from, but put the currently assigned codec in the first position. So far we only offer the single assigned @@ -108,24 +108,11 @@ /* Assignment has completed, the chosen codec should be the first of the resulting SDP. * Make sure this is actually listed in the result SDP and move to first place. */ struct sdp_audio_codec *select = sdp_audio_codecs_by_descr(r, a); - - if (!select) { - /* Not present. Add. */ - if (sdp_audio_codec_by_payload_type(r, a->payload_type, false)) { - /* Oh crunch, that payload type number is already in use. - * Find an unused one. */ - for (a->payload_type = 96; a->payload_type <= 127; a->payload_type++) { - if (!sdp_audio_codec_by_payload_type(r, a->payload_type, false)) - break; - } - - if (a->payload_type > 127) - return -ENOSPC; - } - select = sdp_audio_codecs_add_copy(r, a); - } - - sdp_audio_codecs_select(r, select); + if (select) + sdp_audio_codecs_select(r, select); + /* If 'select' is NULL, the assigned codec is not present in the intersection of possible choices for + * TFO. We could add it, but it would taint the filter result. Just omit the assigned codec, and it is + * the CC code's responsibility to detect this and assign a working codec. */ } #else /* Currently, osmo-msc does not trigger re-assignment if the remote side has picked a codec that is different diff --git a/src/libmsc/gsm_04_08_cc.c b/src/libmsc/gsm_04_08_cc.c index 31fcb23..6b8819f 100644 --- a/src/libmsc/gsm_04_08_cc.c +++ b/src/libmsc/gsm_04_08_cc.c @@ -270,7 +270,16 @@ break; } - if (sdp && sdp[0] && (sdp_msg_from_sdp_str(&sdp_msg, sdp) == 0)) { + if (sdp && sdp[0]) { + int rc = sdp_msg_from_sdp_str(&sdp_msg, sdp); + if (rc != 0) { + LOG_TRANS_CAT_SRC(trans, DMNCC, LOGL_ERROR, file, line, "%s %s: invalid SDP message (trying anyway)\n", + rx_tx, + get_mncc_name(mncc->msg_type)); + LOG_TRANS_CAT_SRC(trans, DMNCC, LOGL_DEBUG, file, line, "erratic SDP: %s\n", + osmo_quote_cstr_c(OTC_SELECT, sdp, -1)); + return; + } LOG_TRANS_CAT_SRC(trans, DMNCC, LOGL_DEBUG, file, line, "%s %s (RTP=%s)\n", rx_tx, get_mncc_name(mncc->msg_type), @@ -1131,6 +1140,7 @@ struct gsm_mncc *alerting = arg; struct msgb *msg = gsm48_msgb_alloc_name("GSM 04.08 CC ALERT"); struct gsm48_hdr *gh = (struct gsm48_hdr *) msgb_put(msg, sizeof(*gh)); + struct codec_filter *codecs = &trans->cc.codecs; int rc; gh->msg_type = GSM48_MT_CC_ALERTING; @@ -1163,7 +1173,23 @@ } } - return trans_tx_gsm48(trans, msg); + /* First handle the MNCC event */ + rc = trans_tx_gsm48(trans, msg); + + /* Now see if the codecs are fine for TFO: + * This is the first time we are told the MT call leg's codec capabilities, via the MNCC_ALERT_REQ from MT to + * MO. Here, at MO, we have already assigned a specific codec. If the MT call leg does not support this codec, + * but the MO does support one of MT's codecs, we need to re-assign our assigned codec to match MT. */ + if (sdp_audio_codec_is_set(&codecs->assignment) + && trans->cc.remote.audio_codecs.count + && !sdp_audio_codecs_by_descr(&trans->cc.remote.audio_codecs, &codecs->assignment)) { + LOG_TRANS(trans, LOGL_ERROR, "Remote call leg mismatches assigned codec: %s\n", + codec_filter_to_str(&trans->cc.codecs, &trans->cc.local, &trans->cc.remote)); + + msc_a_tx_assignment(trans->msc_a); + } + + return rc; } static int gsm48_cc_tx_progress(struct gsm_trans *trans, void *arg) @@ -2056,17 +2082,23 @@ switch (trans->cc.state) { case GSM_CSTATE_INITIATED: case GSM_CSTATE_MO_CALL_PROC: - /* MO call */ + /* MO call, send ACK in form of an MNCC_RTP_CREATE (below) */ break; + case GSM_CSTATE_CALL_DELIVERED: + case GSM_CSTATE_CONNECT_IND: + /* MO re-assignment after MT codec mismatched MO codecs */ + LOG_TRANS(trans, LOGL_DEBUG, "Re-Assignment complete\n"); + return 0; + case GSM_CSTATE_CALL_RECEIVED: case GSM_CSTATE_MO_TERM_CALL_CONF: - /* MT call */ + /* MT call, send ACK in form of an MNCC_RTP_CREATE (below) */ break; case GSM_CSTATE_ACTIVE: /* already active. MNCC finished before Abis completed the Assignment. */ - break; + return 0; default: LOG_TRANS(trans, LOGL_ERROR, "Assignment done in unexpected CC state: %d\n", trans->cc.state); diff --git a/src/libmsc/msc_a.c b/src/libmsc/msc_a.c index e64b54d..c10afb8 100644 --- a/src/libmsc/msc_a.c +++ b/src/libmsc/msc_a.c @@ -636,7 +636,7 @@ } /* The MGW has given us a local IP address for the RAN side. Ready to start the Assignment of a voice channel. */ -static void msc_a_call_leg_ran_local_addr_available(struct msc_a *msc_a) +void msc_a_tx_assignment(struct msc_a *msc_a) { struct ran_msg msg; struct gsm_trans *cc_trans = msc_a->cc.active_trans; @@ -804,7 +804,7 @@ rtps->use_osmux ? "yes" : "no", rtps->local_osmux_cid); switch (rtps->dir) { case RTP_TO_RAN: - msc_a_call_leg_ran_local_addr_available(msc_a); + msc_a_tx_assignment(msc_a); return; case RTP_TO_CN: cc_on_cn_local_rtp_port_known(rtps->for_trans); diff --git a/src/libmsc/msc_ho.c b/src/libmsc/msc_ho.c index f826975..47f000b 100644 --- a/src/libmsc/msc_ho.c +++ b/src/libmsc/msc_ho.c @@ -380,7 +380,7 @@ struct vlr_subscr *vsub = msc_a_vsub(msc_a); struct gsm_network *net = msc_a_net(msc_a); struct gsm0808_channel_type channel_type; - struct gsm0808_speech_codec_list scl; + struct gsm0808_speech_codec_list scl = {}; struct gsm_trans *cc_trans = msc_a->cc.active_trans; struct ran_msg ran_enc_msg = { .msg_type = RAN_MSG_HANDOVER_REQUEST, @@ -442,7 +442,13 @@ ran_enc_msg.handover_request.call_id_present = true; ran_enc_msg.handover_request.call_id = cc_trans->call_id; - sdp_audio_codecs_to_speech_codec_list(&scl, &cc_trans->cc.local.audio_codecs); + /* Call assignment is now capable of re-assigning to overcome a codec mismatch with the remote call leg. + * But for inter-MSC handover, that is not supported yet. So keep here the old limitation of only + * offering the assigned codec. */ + if (sdp_audio_codec_is_set(&cc_trans->cc.codecs.assignment)) + sdp_audio_codec_to_speech_codec_list(&scl, &cc_trans->cc.codecs.assignment); + else + sdp_audio_codecs_to_speech_codec_list(&scl, &cc_trans->cc.local.audio_codecs); if (!scl.len) { msc_ho_failed(msc_a, GSM0808_CAUSE_EQUIPMENT_FAILURE, "Failed to compose" " Codec List (MSC Preferred) for Handover Request message\n"); diff --git a/tests/msc_vlr/msc_vlr_test_call.c b/tests/msc_vlr/msc_vlr_test_call.c index cb3c77b..74bdfcf 100644 --- a/tests/msc_vlr/msc_vlr_test_call.c +++ b/tests/msc_vlr/msc_vlr_test_call.c @@ -1083,6 +1083,9 @@ return false; } expect_pos++; + + // only match first codec + return true; } if (*expect_pos) { BTW("%s: %s: ERROR: mismatch: expected %s to be listed, but not found", func, desc, *expect_pos); diff --git a/tests/msc_vlr/msc_vlr_test_call.err b/tests/msc_vlr/msc_vlr_test_call.err index a1da0f7..18bc2c1 100644 --- a/tests/msc_vlr/msc_vlr_test_call.err +++ b/tests/msc_vlr/msc_vlr_test_call.err @@ -2634,19 +2634,22 @@ DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-6:RTP_TO_RAN:no-CI){UNINITIALIZED}: setting codecs to AMR:octet-align=1#112 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-6:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: setting remote addr to 1.2.3.4:1234 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-6:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Not committing: no MGW endpoint CI set up -DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) -DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) Assignment Complete: RAN: AMR:octet-align=1#112, CN: AMR:octet-align=1#112 -DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) -DMNCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) tx MNCC_RTP_CREATE (RTP=10.23.23.1:23{AMR:octet-align=1#112}) +DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) +DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) Assignment Complete: RAN: AMR:octet-align=1#112, CN: AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111 +DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) +DMNCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) tx MNCC_RTP_CREATE (RTP=10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) MSC --> MNCC: callref 0x80000004: MNCC_RTP_CREATE v=0 o=OsmoMSC 0 0 IN IP4 10.23.23.1 s=GSM Call c=IN IP4 10.23.23.1 t=0 0 -m=audio 23 RTP/AVP 112 +m=audio 23 RTP/AVP 112 110 3 111 a=rtpmap:112 AMR/8000 a=fmtp:112 octet-align=1 +a=rtpmap:110 GSM-EFR/8000 +a=rtpmap:3 GSM/8000 +a=rtpmap:111 GSM-HR-08/8000 a=ptime:20 - VALIDATE_SDP OK: cc_to_mncc_tx_last_sdp == t->mo_tx_sdp_mncc_rtp_create == AMR @@ -4458,19 +4461,22 @@ DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-12:RTP_TO_RAN:no-CI){UNINITIALIZED}: setting codecs to AMR:octet-align=1#112 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-12:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: setting remote addr to 1.2.3.4:1234 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-12:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Not committing: no MGW endpoint CI set up -DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) -DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) Assignment Complete: RAN: AMR:octet-align=1#112, CN: AMR:octet-align=1#112 -DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) -DMNCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) tx MNCC_RTP_CREATE (RTP=10.23.23.1:23{AMR:octet-align=1#112}) +DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) +DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) Assignment Complete: RAN: AMR:octet-align=1#112, CN: AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111 +DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) +DMNCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) tx MNCC_RTP_CREATE (RTP=10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) MSC --> MNCC: callref 0x80000007: MNCC_RTP_CREATE v=0 o=OsmoMSC 0 0 IN IP4 10.23.23.1 s=GSM Call c=IN IP4 10.23.23.1 t=0 0 -m=audio 23 RTP/AVP 112 +m=audio 23 RTP/AVP 112 110 3 111 a=rtpmap:112 AMR/8000 a=fmtp:112 octet-align=1 +a=rtpmap:110 GSM-EFR/8000 +a=rtpmap:3 GSM/8000 +a=rtpmap:111 GSM-HR-08/8000 a=ptime:20 - VALIDATE_SDP OK: cc_to_mncc_tx_last_sdp == t->mo_tx_sdp_mncc_rtp_create == AMR @@ -4859,19 +4865,22 @@ DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-14:RTP_TO_RAN:no-CI){UNINITIALIZED}: setting codecs to AMR:octet-align=1#112 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-14:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: setting remote addr to 1.2.3.4:1234 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-14:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Not committing: no MGW endpoint CI set up -DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) -DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) Assignment Complete: RAN: AMR:octet-align=1#112, CN: AMR:octet-align=1#112 -DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) -DMNCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) tx MNCC_RTP_CREATE (RTP=10.23.23.1:23{AMR:octet-align=1#112}) +DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) +DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) Assignment Complete: RAN: AMR:octet-align=1#112, CN: AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111 +DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) +DMNCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) tx MNCC_RTP_CREATE (RTP=10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111}) MSC --> MNCC: callref 0x80000008: MNCC_RTP_CREATE v=0 o=OsmoMSC 0 0 IN IP4 10.23.23.1 s=GSM Call c=IN IP4 10.23.23.1 t=0 0 -m=audio 23 RTP/AVP 112 +m=audio 23 RTP/AVP 112 110 3 111 a=rtpmap:112 AMR/8000 a=fmtp:112 octet-align=1 +a=rtpmap:110 GSM-EFR/8000 +a=rtpmap:3 GSM/8000 +a=rtpmap:111 GSM-HR-08/8000 a=ptime:20 - VALIDATE_SDP OK: cc_to_mncc_tx_last_sdp == t->mo_tx_sdp_mncc_rtp_create == AMR -- To view, visit https://gerrit.osmocom.org/c/osmo-msc/+/35051?usp=email To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmo-msc Gerrit-Branch: master Gerrit-Change-Id: I8760feaa8598047369ef8c3ab2673013bac8ac8a Gerrit-Change-Number: 35051 Gerrit-PatchSet: 1 Gerrit-Owner: neels <nhofmeyr(a)sysmocom.de> Gerrit-MessageType: newchange

1 year, 7 months

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[S] Change in libosmocore[master]: LAPD: Indicate sequence error after indicating received data

by jolly

jolly has uploaded this change for review. ( https://gerrit.osmocom.org/c/libosmocore/+/35048?usp=email ) Change subject: LAPD: Indicate sequence error after indicating received data ...................................................................... LAPD: Indicate sequence error after indicating received data First indicated the received data in an I frame, if possible. Then indicate the sequence error using MDL-ERROR-INDICATION. This way the data is delivered before the error is handled by BSC. Also there is no reason to indicate sequence error on supervisory frames. See §8.7.4 of 3GPP TS 44.006. Related: OS#5968 Change-Id: I535c18018bf0df4124a5e9618238028fa31be289 --- M src/isdn/lapd_core.c 1 file changed, 34 insertions(+), 5 deletions(-) git pull ssh://gerrit.osmocom.org:29418/libosmocore refs/changes/48/35048/1 diff --git a/src/isdn/lapd_core.c b/src/isdn/lapd_core.c index 925f209..516d0d1 100644 --- a/src/isdn/lapd_core.c +++ b/src/isdn/lapd_core.c @@ -796,8 +796,9 @@ lapd_start_t200(dl); } -/* 5.5.3.1: Common function to acknowlege frames up to the given N(R) value */ -static void lapd_acknowledge(struct lapd_msg_ctx *lctx) +/* 5.5.3.1: Common function to acknowlege frames up to the given N(R) value + * In case of a sequence error, the cause is returned with negative sign. */ +static int lapd_acknowledge(struct lapd_msg_ctx *lctx) { struct lapd_datalink *dl = lctx->dl; uint8_t nr = lctx->n_recv; @@ -838,7 +839,7 @@ */ if (sub_mod(nr, dl->v_ack, dl->v_range) > sub_mod(dl->v_send, dl->v_ack, dl->v_range)) { LOGDL(dl, LOGL_NOTICE, "N(R) sequence error\n"); - mdl_error(MDL_CAUSE_SEQ_ERR, lctx); + return -MDL_CAUSE_SEQ_ERR; } } @@ -861,6 +862,8 @@ /* Start T203, if T200 is not running in MF EST state, if enabled */ if (!lapd_is_t200_started(dl) && (dl->t203_sec || dl->t203_usec) && (dl->state == LAPD_STATE_MF_EST)) lapd_start_t203(dl); + + return 0; } /* L1 -> L2 */ @@ -1576,6 +1579,7 @@ int length = lctx->length; int rc; bool i_frame_in_queue = false; + int mdl_cause = 0; LOGDL(dl, LOGL_INFO, "I received in state %s on SAPI(%u)\n", lapd_state_name(dl->state), lctx->sapi); @@ -1652,7 +1656,9 @@ } /* Even if N(s) sequence error, acknowledge to N(R)-1 */ /* 5.5.3.1: Acknowlege all transmitted frames up the N(R)-1 */ - lapd_acknowledge(lctx); /* V(A) is also set here */ + mdl_cause = lapd_acknowledge(lctx); /* V(A) is also set here */ + if (mdl_cause < 0) + mdl_error(-mdl_cause, lctx); /* Send message, if possible due to acknowledged data */ lapd_send_i(dl, __LINE__, false); @@ -1673,7 +1679,7 @@ } /* 5.5.3.1: Acknowlege all transmitted frames up the the N(R)-1 */ - lapd_acknowledge(lctx); /* V(A) is also set here */ + mdl_cause = lapd_acknowledge(lctx); /* V(A) is also set here */ /* Only if we are not in own receiver busy condition */ if (!dl->own_busy) { @@ -1718,6 +1724,10 @@ } else LOGDL(dl, LOGL_INFO, "I frame ignored during own receiver busy condition\n"); + /* Indicate sequence error, if exists. */ + if (mdl_cause < 0) + mdl_error(-mdl_cause, lctx); + /* Check for P bit */ if (lctx->p_f) { /* 5.5.2.1 */ -- To view, visit https://gerrit.osmocom.org/c/libosmocore/+/35048?usp=email To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: libosmocore Gerrit-Branch: master Gerrit-Change-Id: I535c18018bf0df4124a5e9618238028fa31be289 Gerrit-Change-Number: 35048 Gerrit-PatchSet: 1 Gerrit-Owner: jolly <andreas(a)eversberg.eu> Gerrit-MessageType: newchange

1 year, 7 months

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[S] Change in osmo-ttcn3-hacks[master]: BTS_Tests_LAPDm.TC_nr_seq_error: Fix test

by jolly

jolly has uploaded this change for review. ( https://gerrit.osmocom.org/c/osmo-ttcn3-hacks/+/35047?usp=email ) Change subject: BTS_Tests_LAPDm.TC_nr_seq_error: Fix test ...................................................................... BTS_Tests_LAPDm.TC_nr_seq_error: Fix test Related: OS:5968 Change-Id: I4cdff6ab607e158d4956a1bd87f74e9e75579cff --- M bts/BTS_Tests_LAPDm.ttcn 1 file changed, 15 insertions(+), 0 deletions(-) git pull ssh://gerrit.osmocom.org:29418/osmo-ttcn3-hacks refs/changes/47/35047/1 diff --git a/bts/BTS_Tests_LAPDm.ttcn b/bts/BTS_Tests_LAPDm.ttcn index b4c2c33..2e984d1 100644 --- a/bts/BTS_Tests_LAPDm.ttcn +++ b/bts/BTS_Tests_LAPDm.ttcn @@ -801,6 +801,11 @@ /* SAPI = 0, C = 1, P = 0, M = 0, L = N201, N(S) = 0, N(R) = 1. * */ LAPDM.send(t_PH_DATA(0, false, ts_LAPDm_I(sapi, c_r:=cr_MO_CMD, p:=false, nr:=1, ns:=0, l3:=l3_mo))); + /* LAPD indicates the data and then indicates error with sequence error cause. */ + RSL.receive(tr_RSL_DATA_IND(g_chan_nr, link_id, l3_mo)); + RSL.receive(tr_RSL_ERROR_IND(g_chan_nr, link_id, '07'O)); + /* Release RSL link. */ + RSL.send(ts_RSL_REL_REQ(g_chan_nr, link_id)); /* The BTS may: a) send a DISC frame within N200×T200; */ /* DISC SAPI = 0, C = 0, P = 1, M = 0, L = 0. */ -- To view, visit https://gerrit.osmocom.org/c/osmo-ttcn3-hacks/+/35047?usp=email To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmo-ttcn3-hacks Gerrit-Branch: master Gerrit-Change-Id: I4cdff6ab607e158d4956a1bd87f74e9e75579cff Gerrit-Change-Number: 35047 Gerrit-PatchSet: 1 Gerrit-Owner: jolly <andreas(a)eversberg.eu> Gerrit-MessageType: newchange

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[L] Change in libosmocore[master]: soft_uart: add unit tests for the receiver and transmitter

by laforge

Attention is currently required from: fixeria. laforge has posted comments on this change. ( https://gerrit.osmocom.org/c/libosmocore/+/35046?usp=email ) Change subject: soft_uart: add unit tests for the receiver and transmitter ...................................................................... Patch Set 2: Code-Review+1 (1 comment) Patchset: PS2: please squash -- To view, visit https://gerrit.osmocom.org/c/libosmocore/+/35046?usp=email To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: libosmocore Gerrit-Branch: master Gerrit-Change-Id: Icdfa0c644548964d37940c32dc9dcfcfc53c3a19 Gerrit-Change-Number: 35046 Gerrit-PatchSet: 2 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: laforge <laforge(a)osmocom.org> Gerrit-Attention: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-Comment-Date: Thu, 16 Nov 2023 15:30:49 +0000 Gerrit-HasComments: Yes Gerrit-Has-Labels: Yes Gerrit-MessageType: comment

1 year, 7 months

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[M] Change in libosmocore[master]: [WIP] soft_uart: implement OSMO_SUART_RX_MODE_N_FRAMES

by laforge

Attention is currently required from: fixeria. laforge has posted comments on this change. ( https://gerrit.osmocom.org/c/libosmocore/+/35045?usp=email ) Change subject: [WIP] soft_uart: implement OSMO_SUART_RX_MODE_N_FRAMES ...................................................................... Patch Set 2: (1 comment) File include/osmocom/core/soft_uart.h: https://gerrit.osmocom.org/c/libosmocore/+/35045/comment/feaa2f0c_67944841 PS2, Line 78: number of received UART frames for OSMO_SUART_RX_MODE_N_FRAMES; UART : * will flush receive buffer via the receive call-back after indicated number : * of UART frames had been received. * how is this different from the existing behavior if you'd simply make the rx buffer size == rx_n_uart_frames? The original code was written in a way that *normally* you would call the call-back whenever the buffer is full, but in casese that takes too long, we use the timeout as a fall-back. -- To view, visit https://gerrit.osmocom.org/c/libosmocore/+/35045?usp=email To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: libosmocore Gerrit-Branch: master Gerrit-Change-Id: Ib3249a06c84f3ddb2723d0787db51873c4707d81 Gerrit-Change-Number: 35045 Gerrit-PatchSet: 2 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-CC: laforge <laforge(a)osmocom.org> Gerrit-Attention: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-Comment-Date: Thu, 16 Nov 2023 15:30:27 +0000 Gerrit-HasComments: Yes Gerrit-Has-Labels: No Gerrit-MessageType: comment

1 year, 7 months

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[M] Change in libosmocore[master]: [WIP] soft_uart: implement OSMO_SUART_RX_MODE_N_FRAMES

by laforge

Attention is currently required from: fixeria. laforge has posted comments on this change. ( https://gerrit.osmocom.org/c/libosmocore/+/35045?usp=email ) Change subject: [WIP] soft_uart: implement OSMO_SUART_RX_MODE_N_FRAMES ...................................................................... Patch Set 2: (1 comment) Patchset: PS2: might be worth explaining the usage pattern. Also, if it does what I think it does, it might be easier to understand it as a "rx buffer fill threshold expressed in bytes" or the like? -- To view, visit https://gerrit.osmocom.org/c/libosmocore/+/35045?usp=email To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: libosmocore Gerrit-Branch: master Gerrit-Change-Id: Ib3249a06c84f3ddb2723d0787db51873c4707d81 Gerrit-Change-Number: 35045 Gerrit-PatchSet: 2 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-CC: laforge <laforge(a)osmocom.org> Gerrit-Attention: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-Comment-Date: Thu, 16 Nov 2023 15:22:21 +0000 Gerrit-HasComments: Yes Gerrit-Has-Labels: No Gerrit-MessageType: comment

1 year, 7 months

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[M] Change in libosmocore[master]: soft_uart: implement the transmitter

by laforge

Attention is currently required from: fixeria. laforge has posted comments on this change. ( https://gerrit.osmocom.org/c/libosmocore/+/35044?usp=email ) Change subject: soft_uart: implement the transmitter ...................................................................... Patch Set 2: Code-Review-1 (1 comment) File src/core/soft_uart.c: https://gerrit.osmocom.org/c/libosmocore/+/35044/comment/805a3c8e_44104bba PS2, Line 209: size_t osmo_soft_uart_tx(struct osmo_soft_uart *suart, how does this reject any buffer overflows? It is difficult to manage transmission with a buffer, IMHO. I would suggest to let the user of this API manage the data and simply call a function every time the transmitter wants to pull data out of the application-side buffer. This way it's up to the application to manage the size of the buffer, any kind of prioritization of different data, the overflow/overrun semantics, etc. Yes, that would mean one function call-back for each byte the transmitter pulls out of that buffer. If that really turns out to be a performance problem (probably with hundreds of parallel uarts, if we ever need that) we might want to pull multiple bytes at the same time. But in any case, I think we hve a fixed bit-rate consumer of data (the transmitter) and it should pull data. -- To view, visit https://gerrit.osmocom.org/c/libosmocore/+/35044?usp=email To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: libosmocore Gerrit-Branch: master Gerrit-Change-Id: Ibcd9643227e5616efd8bbd7a1430feda6fcef45c Gerrit-Change-Number: 35044 Gerrit-PatchSet: 2 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: laforge <laforge(a)osmocom.org> Gerrit-Attention: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-Comment-Date: Thu, 16 Nov 2023 15:16:40 +0000 Gerrit-HasComments: Yes Gerrit-Has-Labels: Yes Gerrit-MessageType: comment

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[S] Change in libosmocore[master]: soft_uart: implement parity checking for the receiver

by laforge

Attention is currently required from: fixeria. laforge has posted comments on this change. ( https://gerrit.osmocom.org/c/libosmocore/+/35023?usp=email ) Change subject: soft_uart: implement parity checking for the receiver ...................................................................... Patch Set 2: Code-Review+1 (1 comment) Patchset: PS2: can be squashed -- To view, visit https://gerrit.osmocom.org/c/libosmocore/+/35023?usp=email To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: libosmocore Gerrit-Branch: master Gerrit-Change-Id: I28be2ca19d423447a718fb518566d52ae1967ec7 Gerrit-Change-Number: 35023 Gerrit-PatchSet: 2 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: laforge <laforge(a)osmocom.org> Gerrit-Attention: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-Comment-Date: Thu, 16 Nov 2023 15:12:45 +0000 Gerrit-HasComments: Yes Gerrit-Has-Labels: Yes Gerrit-MessageType: comment

1 year, 7 months

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[M] Change in libosmocore[master]: soft_uart: rework osmo_uart_rx_bit() to use flow state

by laforge

Attention is currently required from: fixeria. laforge has posted comments on this change. ( https://gerrit.osmocom.org/c/libosmocore/+/35022?usp=email ) Change subject: soft_uart: rework osmo_uart_rx_bit() to use flow state ...................................................................... Patch Set 2: Code-Review+1 (1 comment) Patchset: PS2: can be squashed -- To view, visit https://gerrit.osmocom.org/c/libosmocore/+/35022?usp=email To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: libosmocore Gerrit-Branch: master Gerrit-Change-Id: I40ab5d12b6f7087daa51405468f5c4ea639561ea Gerrit-Change-Number: 35022 Gerrit-PatchSet: 2 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: laforge <laforge(a)osmocom.org> Gerrit-Attention: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-Comment-Date: Thu, 16 Nov 2023 15:12:26 +0000 Gerrit-HasComments: Yes Gerrit-Has-Labels: Yes Gerrit-MessageType: comment

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