diff options
author | Neels Hofmeyr <nhofmeyr@sysmocom.de> | 2023-11-17 04:12:29 +0100 |
---|---|---|
committer | neels <nhofmeyr@sysmocom.de> | 2023-12-13 01:52:22 +0000 |
commit | d767c73a1f93253a54d6a8650a4cf2143353bba0 (patch) | |
tree | a6bedd5f75c91f728f6df48fff51868b97e4e224 | |
parent | cefe594c72330d70a3efaf05172dd15af36b94f8 (diff) |
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.
For inter-MSC handover, keep the behavior unchanged: offer only the
currently assigned codec to the remote side. Codec-changing HO should be
equally trivial to implement, but that is for another day.
msc_vlr_test_call's codec tests are adjusted to test the new feature in
Ib933554f826c1b4347dfa3f6c4f6fe086be8b133. For now, avoid change in
these tests by validating the first codec in SDP lists only.
Related: OS#6258
Related: osmo-ttcn3-hacks I402ed0523a2a87b83f29c5577b2c828102005d53
Change-Id: I8760feaa8598047369ef8c3ab2673013bac8ac8a
-rw-r--r-- | include/osmocom/msc/msc_a.h | 1 | ||||
-rw-r--r-- | src/libmsc/codec_filter.c | 42 | ||||
-rw-r--r-- | src/libmsc/gsm_04_08_cc.c | 50 | ||||
-rw-r--r-- | src/libmsc/msc_a.c | 4 | ||||
-rw-r--r-- | src/libmsc/msc_ho.c | 10 | ||||
-rw-r--r-- | tests/msc_vlr/msc_vlr_test_call.c | 3 | ||||
-rw-r--r-- | tests/msc_vlr/msc_vlr_test_call.err | 39 |
7 files changed, 89 insertions, 60 deletions
diff --git a/include/osmocom/msc/msc_a.h b/include/osmocom/msc/msc_a.h index 0276d62e2..4099d4cd2 100644 --- a/include/osmocom/msc/msc_a.h +++ b/include/osmocom/msc/msc_a.h @@ -216,6 +216,7 @@ bool msc_a_is_establishing_auth_ciph(const struct msc_a *msc_a); 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_cmd(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/codec_filter.c b/src/libmsc/codec_filter.c index a9d93a70d..7511f9026 100644 --- a/src/libmsc/codec_filter.c +++ b/src/libmsc/codec_filter.c @@ -98,46 +98,16 @@ int codec_filter_run(struct codec_filter *codec_filter, struct sdp_msg *result, if (remote->audio_codecs.count) sdp_audio_codecs_intersection(r, &remote->audio_codecs, true); -#if 0 - /* 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 - * codec, because we have no way to deal with the remote side picking a different codec. - * Another approach would be to postpone assignment until we know the codecs from the remote side. */ if (sdp_audio_codec_is_set(a)) { /* 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. */ + * If present, make sure this is listed in first place. + * If 'select' is NULL, the assigned codec is not present in the intersection of possible choices for + * TFO. Just omit the assigned codec from the filter result, and it is the CC code's responsibility to + * detect this and assign a working codec instead. */ 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); - } -#else - /* Currently, osmo-msc does not trigger re-assignment if the remote side has picked a codec that is different - * from the already assigned codec. - * So, if locally, Assignment has already chosen a codec, this is the single definitive result to be used - * towards the CN. */ - if (sdp_audio_codec_is_set(a)) { - /* Assignment has completed, the chosen codec should be the the only possible one. */ - *r = (struct sdp_audio_codecs){}; - sdp_audio_codecs_add_copy(r, a); + if (select) + sdp_audio_codecs_select(r, select); } -#endif return 0; } diff --git a/src/libmsc/gsm_04_08_cc.c b/src/libmsc/gsm_04_08_cc.c index fe34127ff..f6ec81b41 100644 --- a/src/libmsc/gsm_04_08_cc.c +++ b/src/libmsc/gsm_04_08_cc.c @@ -270,7 +270,16 @@ static void _log_mncc_rx_tx(const char *file, int line, 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), @@ -748,6 +757,7 @@ void gsm48_cc_rx_setup_cn_local_rtp_port_known(struct gsm_trans *trans) static void rx_mncc_sdp(struct gsm_trans *trans, uint32_t mncc_msg_type, const char *sdp, const struct gsm_mncc_bearer_cap *bcap) { + struct codec_filter *codecs = &trans->cc.codecs; struct call_leg *cl = trans->msc_a ? trans->msc_a->cc.call_leg : NULL; struct rtp_stream *rtp_cn = cl ? cl->rtp[RTP_TO_CN] : NULL; @@ -775,6 +785,30 @@ static void rx_mncc_sdp(struct gsm_trans *trans, uint32_t mncc_msg_type, const c rtp_stream_set_remote_addr_and_codecs(rtp_cn, &trans->cc.remote); rtp_stream_commit(rtp_cn); } + + /* See if we need to switch codecs to maintain TFO: has the remote side changed the codecs information? If we + * have already assigned a specific codec here, but the remote call leg has now chosen a different codec, we + * need to re-assign this call leg to match the remote leg. */ + if (!sdp_audio_codec_is_set(&codecs->assignment)) { + /* Voice channel assignment has not completed. Do not interfere. */ + return; + } + if (!trans->cc.remote.audio_codecs.count) { + /* Don't know remote codecs, nothing to do. */ + return; + } + if (sdp_audio_codecs_by_descr(&trans->cc.remote.audio_codecs, &codecs->assignment)) { + /* The assigned codec is part of the remote codec set. All is well. */ + /* TODO: maybe this should require exactly the *first* remote codec to match, because we cannot flexibly + * transcode, and assume the actual payload we will receive is listed in the first place? */ + return; + } + + /* We've already completed Assignment of a voice channel (some time ago), and now the remote side has changed + * to a mismatching codec (list). Try to re-assign this side to a matching codec. */ + LOG_TRANS(trans, LOGL_INFO, "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_cmd(trans->msc_a); } static int gsm48_cc_tx_setup(struct gsm_trans *trans, void *arg) @@ -2049,17 +2083,23 @@ int cc_on_assignment_done(struct gsm_trans *trans) 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_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; + /* already active. We decided to re-assign later on during the call - at time of writing this never + * happens. */ + case GSM_CSTATE_CALL_DELIVERED: + case GSM_CSTATE_CONNECT_IND: + /* MNCC has progressed past the initial assignment. Usually it means that this happened: after + * MNCC_ALERT_REQ, MO has triggered a re-assignment, to adjust MO's codec to MT's codec. */ + LOG_TRANS(trans, LOGL_DEBUG, "Re-Assignment complete\n"); + 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 e64b54d8f..a933bd22d 100644 --- a/src/libmsc/msc_a.c +++ b/src/libmsc/msc_a.c @@ -636,7 +636,7 @@ int msc_a_ensure_cn_local_rtp(struct msc_a *msc_a, struct gsm_trans *cc_trans) } /* 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_cmd(struct msc_a *msc_a) { struct ran_msg msg; struct gsm_trans *cc_trans = msc_a->cc.active_trans; @@ -804,7 +804,7 @@ static void msc_a_fsm_communicating(struct osmo_fsm_inst *fi, uint32_t event, vo 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_cmd(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 f82697503..47f000b2f 100644 --- a/src/libmsc/msc_ho.c +++ b/src/libmsc/msc_ho.c @@ -380,7 +380,7 @@ static void msc_ho_send_handover_request(struct msc_a *msc_a) 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 @@ static void msc_ho_send_handover_request(struct msc_a *msc_a) 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 cb3c77bb1..3b91524b1 100644 --- a/tests/msc_vlr/msc_vlr_test_call.c +++ b/tests/msc_vlr/msc_vlr_test_call.c @@ -1083,6 +1083,9 @@ static bool validate_sdp(const char *func, const char *desc, 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 4af1bcec7..57581753e 100644 --- a/tests/msc_vlr/msc_vlr_test_call.err +++ b/tests/msc_vlr/msc_vlr_test_call.err @@ -2636,19 +2636,22 @@ DBSSAP msc_a(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ){MSC_A_ST_ 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 @@ -4457,19 +4460,22 @@ DBSSAP msc_a(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ){MSC_A_ST_ 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 @@ DBSSAP msc_a(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ){MSC_A_ST_ 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 |