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

[S] Change in osmo-msc[master]: [codecs filter] apply BSS codecs from Assignment Complete
by neels 14 Mar '23

14 Mar '23

neels has submitted this change. ( https://gerrit.osmocom.org/c/osmo-msc/+/30124 ) ( 1 is the latest approved patch-set. No files were changed between the latest approved patch-set and the submitted one. )Change subject: [codecs filter] apply BSS codecs from Assignment Complete ...................................................................... [codecs filter] apply BSS codecs from Assignment Complete Codec List (BSS Supported) is received once in Complete Layer 3 and again in Assignment Complete messages. Use the most recent one, i.e. the one from Assignment Complete, when it occurs. Related: SYS#5066 Change-Id: I5e66ecc7987fa926f39d8be8eaf5799b931ab20a --- M src/libmsc/msc_a.c 1 file changed, 19 insertions(+), 0 deletions(-) Approvals: laforge: Looks good to me, approved pespin: Looks good to me, but someone else must approve Jenkins Builder: Verified diff --git a/src/libmsc/msc_a.c b/src/libmsc/msc_a.c index d9a7341..4c7f763 100644 --- a/src/libmsc/msc_a.c +++ b/src/libmsc/msc_a.c @@ -47,6 +47,7 @@ #include <osmocom/msc/rtp_stream.h> #include <osmocom/msc/msc_ho.h> #include <osmocom/msc/codec_mapping.h> +#include <osmocom/msc/codec_filter.h> #define MSC_A_USE_WAIT_CLEAR_COMPLETE "wait-Clear-Complete" @@ -1353,6 +1354,10 @@ rtp_stream_commit(rtps_to_ran); + /* Remember the Codec List (BSS Supported) */ + if (ac->assignment_complete.codec_list_bss_supported) + codec_filter_set_bss(&cc_trans->cc.codecs, ac->assignment_complete.codec_list_bss_supported); + /* Setup CN side endpoint CI: * Now that * - the first CI has been created and a definitive endpoint name is assigned to the call_leg's MGW -- To view, visit https://gerrit.osmocom.org/c/osmo-msc/+/30124 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmo-msc Gerrit-Branch: master Gerrit-Change-Id: I5e66ecc7987fa926f39d8be8eaf5799b931ab20a Gerrit-Change-Number: 30124 Gerrit-PatchSet: 8 Gerrit-Owner: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: laforge <laforge(a)osmocom.org> Gerrit-Reviewer: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: pespin <pespin(a)sysmocom.de> Gerrit-MessageType: merged

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[M] Change in osmo-msc[master]: msc_vlr_test_call: include RAN RTP addr in ass compl
by neels 14 Mar '23

14 Mar '23

neels has submitted this change. ( https://gerrit.osmocom.org/c/osmo-msc/+/30123 ) ( 1 is the latest approved patch-set. No files were changed between the latest approved patch-set and the submitted one. )Change subject: msc_vlr_test_call: include RAN RTP addr in ass compl ...................................................................... msc_vlr_test_call: include RAN RTP addr in ass compl I noticed by chance that the Assignment Complete message generated in the test lacks a remote RTP address for the RAN side. Make the test more realistic by adding a remote RTP address and port. It doesn't have any bearing on the tests besides more accurately showing RTP stream setup in the logs. Change-Id: Ia428762a16dcc17f036d725a00e0b3767418289b --- M tests/msc_vlr/msc_vlr_test_call.err M tests/msc_vlr/msc_vlr_tests.c 2 files changed, 42 insertions(+), 25 deletions(-) Approvals: laforge: Looks good to me, approved pespin: Looks good to me, but someone else must approve Jenkins Builder: Verified diff --git a/tests/msc_vlr/msc_vlr_test_call.err b/tests/msc_vlr/msc_vlr_test_call.err index 3c31ab7..fbe2601 100644 --- a/tests/msc_vlr/msc_vlr_test_call.err +++ b/tests/msc_vlr/msc_vlr_test_call.err @@ -315,8 +315,8 @@ - Assignment succeeds, triggering CRCX to CN DIUCS msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){MSC_A_ST_COMMUNICATING}: RAN decode: ASSIGNMENT_COMPLETE DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483649:RTP_TO_RAN:no-CI){UNINITIALIZED}: setting codec to AMR/8000/1 -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483649:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: remote addr already :0, no change -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483649:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Not committing: no remote RTP address known +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483649: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-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483649: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 MGW <--CRCX to RTP_TO_CN-- MSC: callref=0x80000001 DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){UNINITIALIZED}: Allocated DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){UNINITIALIZED}: is child of call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) @@ -429,9 +429,9 @@ DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483649:RTP_TO_CN:no-CI){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483649:RTP_TO_CN:no-CI){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483649:RTP_TO_CN:no-CI){UNINITIALIZED}: Deallocated -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483649:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483649:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483649:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Deallocated +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483649:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483649:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483649:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Deallocated DCC call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){RELEASING}: Terminating (cause = OSMO_FSM_TERM_REGULAR) DCC call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){RELEASING}: Removing from parent msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) DMSC msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){MSC_A_ST_RELEASING}: Received Event MSC_EV_CALL_LEG_TERM @@ -801,8 +801,8 @@ - Assignment completes, triggering CRCX to CN DIUCS msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP){MSC_A_ST_COMMUNICATING}: RAN decode: ASSIGNMENT_COMPLETE DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI){UNINITIALIZED}: setting codec to AMR/8000/1 -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: remote addr already :0, no change -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Not committing: no remote RTP address known +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059: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-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059: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 MGW <--CRCX to RTP_TO_CN-- MSC: callref=0x423 DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP){UNINITIALIZED}: Allocated DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP){UNINITIALIZED}: is child of call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP) @@ -899,9 +899,9 @@ DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_CN:no-CI){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_CN:no-CI){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP) DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_CN:no-CI){UNINITIALIZED}: Deallocated -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP) -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Deallocated +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP) +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Deallocated DCC call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP){RELEASING}: Terminating (cause = OSMO_FSM_TERM_REGULAR) DCC call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP){RELEASING}: Removing from parent msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP) DMSC msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP){MSC_A_ST_RELEASING}: Received Event MSC_EV_CALL_LEG_TERM @@ -1271,8 +1271,8 @@ - Assignment completes, triggering CRCX to CN DIUCS msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP){MSC_A_ST_COMMUNICATING}: RAN decode: ASSIGNMENT_COMPLETE DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI){UNINITIALIZED}: setting codec to AMR/8000/1 -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: remote addr already :0, no change -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Not committing: no remote RTP address known +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059: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-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059: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 MGW <--CRCX to RTP_TO_CN-- MSC: callref=0x423 DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP){UNINITIALIZED}: Allocated DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP){UNINITIALIZED}: is child of call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP) @@ -1324,9 +1324,9 @@ DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_CN:no-CI){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_CN:no-CI){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP) DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_CN:no-CI){UNINITIALIZED}: Deallocated -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP) -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Deallocated +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP) +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP:trans-0:call-1059:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Deallocated DCC call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP){RELEASING}: Terminating (cause = OSMO_FSM_TERM_REGULAR) DCC call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP){RELEASING}: Removing from parent msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP) DMSC msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:PAGING_RESP){MSC_A_ST_RELEASING}: Received Event MSC_EV_CALL_LEG_TERM @@ -1681,8 +1681,8 @@ - Assignment succeeds, triggering CRCX to CN DIUCS msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){MSC_A_ST_COMMUNICATING}: RAN decode: ASSIGNMENT_COMPLETE DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483650:RTP_TO_RAN:no-CI){UNINITIALIZED}: setting codec to AMR/8000/1 -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483650:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: remote addr already :0, no change -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483650:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Not committing: no remote RTP address known +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483650: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-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483650: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 MGW <--CRCX to RTP_TO_CN-- MSC: callref=0x80000002 DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){UNINITIALIZED}: Allocated DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){UNINITIALIZED}: is child of call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) @@ -1752,9 +1752,9 @@ DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483650:RTP_TO_CN:no-CI){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483650:RTP_TO_CN:no-CI){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483650:RTP_TO_CN:no-CI){UNINITIALIZED}: Deallocated -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483650:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483650:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483650:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Deallocated +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483650:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483650:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483650:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Deallocated DCC call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){RELEASING}: Terminating (cause = OSMO_FSM_TERM_REGULAR) DCC call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){RELEASING}: Removing from parent msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) DMSC msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){MSC_A_ST_RELEASING}: Received Event MSC_EV_CALL_LEG_TERM @@ -2108,8 +2108,8 @@ - Assignment succeeds, triggering CRCX to CN DIUCS msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){MSC_A_ST_COMMUNICATING}: RAN decode: ASSIGNMENT_COMPLETE DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483651:RTP_TO_RAN:no-CI){UNINITIALIZED}: setting codec to AMR/8000/1 -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483651:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: remote addr already :0, no change -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483651:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Not committing: no remote RTP address known +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483651: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-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483651: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 MGW <--CRCX to RTP_TO_CN-- MSC: callref=0x80000003 DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){UNINITIALIZED}: Allocated DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){UNINITIALIZED}: is child of call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) @@ -2176,9 +2176,9 @@ DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483651:RTP_TO_CN:no-CI){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483651:RTP_TO_CN:no-CI){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483651:RTP_TO_CN:no-CI){UNINITIALIZED}: Deallocated -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483651:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483651:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) -DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483651:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: Deallocated +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483651:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Terminating (cause = OSMO_FSM_TERM_PARENT) +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483651:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Removing from parent call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) +DCC rtp_stream(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ:trans-8:call-2147483651:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Deallocated DCC call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){RELEASING}: Terminating (cause = OSMO_FSM_TERM_REGULAR) DCC call_leg(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){RELEASING}: Removing from parent msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ) DMSC msc_a(IMSI-901700000010650:MSISDN-42342:TMSI-0x03020100:UTRAN-Iu:CM_SERVICE_REQ){MSC_A_ST_RELEASING}: Received Event MSC_EV_CALL_LEG_TERM diff --git a/tests/msc_vlr/msc_vlr_tests.c b/tests/msc_vlr/msc_vlr_tests.c index 3ef94f2..6b80452 100644 --- a/tests/msc_vlr/msc_vlr_tests.c +++ b/tests/msc_vlr/msc_vlr_tests.c @@ -1011,6 +1011,7 @@ .codec = assigned_codec, }, }; + osmo_sockaddr_str_from_str(&ran_dec.assignment_complete.remote_rtp, "1.2.3.4", 1234); fake_msc_a_ran_dec(&ran_dec); if (!conn_exists(g_msub)) -- To view, visit https://gerrit.osmocom.org/c/osmo-msc/+/30123 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmo-msc Gerrit-Branch: master Gerrit-Change-Id: Ia428762a16dcc17f036d725a00e0b3767418289b Gerrit-Change-Number: 30123 Gerrit-PatchSet: 8 Gerrit-Owner: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: laforge <laforge(a)osmocom.org> Gerrit-Reviewer: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: pespin <pespin(a)sysmocom.de> Gerrit-MessageType: merged

1 0

[L] Change in osmo-msc[master]: [codecs filter] add codec_filter.h,c
by neels 14 Mar '23

14 Mar '23

neels has submitted this change. ( https://gerrit.osmocom.org/c/osmo-msc/+/30116 ) ( 4 is the latest approved patch-set. No files were changed between the latest approved patch-set and the submitted one. )Change subject: [codecs filter] add codec_filter.h,c ...................................................................... [codecs filter] add codec_filter.h,c Add the infrastructure to store and filter all codec limitiations from the different stages: MS, BSS, CN and remote call leg. Upcoming patches will properly collect these and find an optimal codec. No functional change, yet. Related: SYS#5066 Change-Id: I4d90f7ca62f2307a7b93dd164aeecbf4bd98ff0a --- M include/osmocom/msc/Makefile.am A include/osmocom/msc/codec_filter.h M src/libmsc/Makefile.am A src/libmsc/codec_filter.c 4 files changed, 299 insertions(+), 0 deletions(-) Approvals: laforge: Looks good to me, approved pespin: Looks good to me, but someone else must approve Jenkins Builder: Verified diff --git a/include/osmocom/msc/Makefile.am b/include/osmocom/msc/Makefile.am index 80f8972..34b4902 100644 --- a/include/osmocom/msc/Makefile.am +++ b/include/osmocom/msc/Makefile.am @@ -1,6 +1,7 @@ noinst_HEADERS = \ call_leg.h \ cell_id_list.h \ + codec_filter.h \ codec_mapping.h \ db.h \ debug.h \ diff --git a/include/osmocom/msc/codec_filter.h b/include/osmocom/msc/codec_filter.h new file mode 100644 index 0000000..16315c7 --- /dev/null +++ b/include/osmocom/msc/codec_filter.h @@ -0,0 +1,66 @@ +/* Filter/overlay codec selections for a voice call, across MS, RAN and CN limitations */ +/* + * (C) 2019-2022 by sysmocom - s.m.f.c. GmbH <info(a)sysmocom.de> + * All Rights Reserved + * + * Author: Neels Hofmeyr + * + * SPDX-License-Identifier: GPL-2.0+ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ +#pragma once + +#include <osmocom/gsm/gsm_utils.h> +#include <osmocom/gsm/mncc.h> +#include <osmocom/mgcp_client/mgcp_client.h> + +#include <osmocom/msc/sdp_msg.h> + +struct gsm0808_speech_codec_list; + +/* Combine various codec selections to obtain a resulting set of codecs allowed by all of them. + * Members reflect the different entities/stages that select codecs in a voice call. + * Call codec_filter_run() and obtain the resulting set of codecs in codec_filter.result. */ +struct codec_filter { + /* The fixed set of codecs available on the RAN type, per definition. */ + struct sdp_audio_codecs ran; + /* The codecs advertised by the MS Bearer Capabilities */ + struct sdp_audio_codecs ms; + /* If known, the set of codecs the current RAN cell allows / has available. + * This may not be available if the BSC does not issue this information early enough. + * Should be ignored if empty. */ + struct sdp_audio_codecs bss; + + /* SDP as last received from the remote call leg. */ + struct sdp_msg remote; + + /* After a channel was assigned, this reflects the chosen codec. */ + struct sdp_audio_codec assignment; + + /* Resulting choice of supported codecs, usually the intersection of the above, + * and the local RTP address to be sent to the remote call leg. + * The RTP address:port in result.rtp is not modified by codec_filter_run() -- set it once. */ + struct sdp_msg result; +}; + +void codec_filter_init(struct codec_filter *codec_filter); +void codec_filter_set_ran(struct codec_filter *codec_filter, enum osmo_rat_type ran_type); +void codec_filter_set_ms_from_bc(struct codec_filter *codec_filter, const struct gsm_mncc_bearer_cap *ms_bearer_cap); +void codec_filter_set_bss(struct codec_filter *codec_filter, + const struct gsm0808_speech_codec_list *codec_list_bss_supported); +int codec_filter_set_remote(struct codec_filter *codec_filter, const char *remote_sdp); +void codec_filter_set_local_rtp(struct codec_filter *codec_filter, const struct osmo_sockaddr_str *rtp); +int codec_filter_run(struct codec_filter *codec_filter); + +int codec_filter_to_str_buf(char *buf, size_t buflen, const struct codec_filter *codec_filter); +char *codec_filter_to_str_c(void *ctx, const struct codec_filter *codec_filter); +const char *codec_filter_to_str(const struct codec_filter *codec_filter); diff --git a/src/libmsc/Makefile.am b/src/libmsc/Makefile.am index c7fe4bb..bde70c9 100644 --- a/src/libmsc/Makefile.am +++ b/src/libmsc/Makefile.am @@ -27,6 +27,7 @@ libmsc_a_SOURCES = \ call_leg.c \ cell_id_list.c \ + codec_filter.c \ codec_mapping.c \ sccp_ran.c \ msc_vty.c \ diff --git a/src/libmsc/codec_filter.c b/src/libmsc/codec_filter.c new file mode 100644 index 0000000..c77aeb9 --- /dev/null +++ b/src/libmsc/codec_filter.c @@ -0,0 +1,215 @@ +/* Filter/overlay codec selections for a voice call, across MS, RAN and CN limitations */ +/* + * (C) 2019-2023 by sysmocom - s.m.f.c. GmbH <info(a)sysmocom.de> + * All Rights Reserved + * + * Author: Neels Hofmeyr + * + * SPDX-License-Identifier: AGPL-3.0+ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU Affero General Public License as published by + * the Free Software Foundation; either version 3 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU Affero General Public License for more details. + * + * You should have received a copy of the GNU Affero General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <osmocom/gsm/protocol/gsm_08_08.h> + +#include <osmocom/msc/codec_filter.h> +#include <osmocom/msc/codec_mapping.h> +#include <osmocom/msc/debug.h> + +/* Add all known payload types encountered in GSM networks */ +static void sdp_add_all_geran_codecs(struct sdp_audio_codecs *ac) +{ + /* In order of preference. TODO: make configurable */ + static const enum gsm48_bcap_speech_ver mobile_codecs[] = { + GSM48_BCAP_SV_AMR_F /*!< 4 GSM FR V3 (FR AMR) */, + GSM48_BCAP_SV_AMR_H /*!< 5 GSM HR V3 (HR_AMR) */, + GSM48_BCAP_SV_EFR /*!< 2 GSM FR V2 (GSM EFR) */, + GSM48_BCAP_SV_FR /*!< 0 GSM FR V1 (GSM FR) */, + GSM48_BCAP_SV_HR /*!< 1 GSM HR V1 (GSM HR) */, + }; + int i; + for (i = 0; i < ARRAY_SIZE(mobile_codecs); i++) + sdp_audio_codecs_add_speech_ver(ac, mobile_codecs[i]); +} + +/* Add all known AMR payload types encountered in UTRAN networks */ +static void sdp_add_all_utran_codecs(struct sdp_audio_codecs *ac) +{ + /* In order of preference. TODO: make configurable */ + static const enum gsm48_bcap_speech_ver utran_codecs[] = { + GSM48_BCAP_SV_AMR_F /*!< 4 GSM FR V3 (FR AMR) */, + GSM48_BCAP_SV_AMR_H /*!< 5 GSM HR V3 (HR_AMR) */, + GSM48_BCAP_SV_AMR_OH /*!< 11 GSM HR V6 (OHR AMR) */, + GSM48_BCAP_SV_AMR_FW /*!< 8 GSM FR V5 (FR AMR-WB) */, + GSM48_BCAP_SV_AMR_OFW /*!< 6 GSM FR V4 (OFR AMR-WB) */, + GSM48_BCAP_SV_AMR_OHW /*!< 7 GSM HR V4 (OHR AMR-WB) */, + }; + int i; + for (i = 0; i < ARRAY_SIZE(utran_codecs); i++) + sdp_audio_codecs_add_speech_ver(ac, utran_codecs[i]); +} + +void codec_filter_init(struct codec_filter *codec_filter) +{ + *codec_filter = (struct codec_filter){}; +} + +void codec_filter_set_ran(struct codec_filter *codec_filter, enum osmo_rat_type ran_type) +{ + codec_filter->ran = (struct sdp_audio_codecs){}; + + switch (ran_type) { + default: + case OSMO_RAT_GERAN_A: + sdp_add_all_geran_codecs(&codec_filter->ran); + break; + + case OSMO_RAT_UTRAN_IU: + sdp_add_all_utran_codecs(&codec_filter->ran); + break; + } +} + +void codec_filter_set_ms_from_bc(struct codec_filter *codec_filter, const struct gsm_mncc_bearer_cap *ms_bearer_cap) +{ + codec_filter->ms = (struct sdp_audio_codecs){0}; + if (ms_bearer_cap) + sdp_audio_codecs_from_bearer_cap(&codec_filter->ms, ms_bearer_cap); +} + +void codec_filter_set_bss(struct codec_filter *codec_filter, + const struct gsm0808_speech_codec_list *codec_list_bss_supported) +{ + codec_filter->bss = (struct sdp_audio_codecs){}; + if (codec_list_bss_supported) + sdp_audio_codecs_from_speech_codec_list(&codec_filter->bss, codec_list_bss_supported); +} + +int codec_filter_set_remote(struct codec_filter *codec_filter, const char *remote_sdp) +{ + return sdp_msg_from_sdp_str(&codec_filter->remote, remote_sdp); +} + +void codec_filter_set_local_rtp(struct codec_filter *codec_filter, const struct osmo_sockaddr_str *rtp) +{ + if (!rtp) + codec_filter->result.rtp = (struct osmo_sockaddr_str){0}; + else + codec_filter->result.rtp = *rtp; +} + +/* Render intersections of all known audio codec constraints to reach a resulting choice of favorite audio codec, plus + * possible set of alternative audio codecs, in codec_filter->result. (The result.rtp address remains unchanged.) */ +int codec_filter_run(struct codec_filter *codec_filter) +{ + struct sdp_audio_codecs *r = &codec_filter->result.audio_codecs; + struct sdp_audio_codec *a = &codec_filter->assignment; + *r = codec_filter->ran; + if (codec_filter->ms.count) + sdp_audio_codecs_intersection(r, &codec_filter->ms, false); + if (codec_filter->bss.count) + sdp_audio_codecs_intersection(r, &codec_filter->bss, false); + if (codec_filter->remote.audio_codecs.count) + sdp_audio_codecs_intersection(r, &codec_filter->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. */ + 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); + } +#endif + return 0; +} + +int codec_filter_to_str_buf(char *buf, size_t buflen, const struct codec_filter *codec_filter) +{ + struct osmo_strbuf sb = { .buf = buf, .len = buflen }; + OSMO_STRBUF_APPEND(sb, sdp_msg_to_str_buf, &codec_filter->result); + OSMO_STRBUF_PRINTF(sb, " (from:"); + + if (sdp_audio_codec_is_set(&codec_filter->assignment)) { + OSMO_STRBUF_PRINTF(sb, " assigned="); + OSMO_STRBUF_APPEND(sb, sdp_audio_codec_to_str_buf, &codec_filter->assignment); + } + + if (codec_filter->remote.audio_codecs.count + || osmo_sockaddr_str_is_nonzero(&codec_filter->remote.rtp)) { + OSMO_STRBUF_PRINTF(sb, " remote="); + OSMO_STRBUF_APPEND(sb, sdp_msg_to_str_buf, &codec_filter->remote); + } + + if (codec_filter->ms.count) { + OSMO_STRBUF_PRINTF(sb, " MS={"); + OSMO_STRBUF_APPEND(sb, sdp_audio_codecs_to_str_buf, &codec_filter->ms); + OSMO_STRBUF_PRINTF(sb, "}"); + } + + if (codec_filter->bss.count) { + OSMO_STRBUF_PRINTF(sb, " bss={"); + OSMO_STRBUF_APPEND(sb, sdp_audio_codecs_to_str_buf, &codec_filter->bss); + OSMO_STRBUF_PRINTF(sb, "}"); + } + + OSMO_STRBUF_PRINTF(sb, " RAN={"); + OSMO_STRBUF_APPEND(sb, sdp_audio_codecs_to_str_buf, &codec_filter->ran); + OSMO_STRBUF_PRINTF(sb, "}"); + + OSMO_STRBUF_PRINTF(sb, ")"); + + return sb.chars_needed; +} + +char *codec_filter_to_str_c(void *ctx, const struct codec_filter *codec_filter) +{ + OSMO_NAME_C_IMPL(ctx, 128, "codec_filter_to_str_c-ERROR", codec_filter_to_str_buf, codec_filter) +} + +const char *codec_filter_to_str(const struct codec_filter *codec_filter) +{ + return codec_filter_to_str_c(OTC_SELECT, codec_filter); +} -- To view, visit https://gerrit.osmocom.org/c/osmo-msc/+/30116 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmo-msc Gerrit-Branch: master Gerrit-Change-Id: I4d90f7ca62f2307a7b93dd164aeecbf4bd98ff0a Gerrit-Change-Number: 30116 Gerrit-PatchSet: 8 Gerrit-Owner: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: laforge <laforge(a)osmocom.org> Gerrit-Reviewer: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: pespin <pespin(a)sysmocom.de> Gerrit-MessageType: merged

1 0

[M] Change in osmo-msc[master]: MNCC: use codec_mapping, drop mgcp_codec_to_mncc_payload_msg_type()
by neels 14 Mar '23

14 Mar '23

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)) -- To view, visit https://gerrit.osmocom.org/c/osmo-msc/+/30115 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmo-msc Gerrit-Branch: master Gerrit-Change-Id: I8995ef43b9f79bc1db5672362c6433e4d96dd9e0 Gerrit-Change-Number: 30115 Gerrit-PatchSet: 8 Gerrit-Owner: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: laforge <laforge(a)osmocom.org> Gerrit-Reviewer: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: pespin <pespin(a)sysmocom.de> Gerrit-MessageType: merged

1 0

[L] Change in osmo-msc[master]: add codec_mapping.h,c
by neels 14 Mar '23

14 Mar '23

neels has submitted this change. ( https://gerrit.osmocom.org/c/osmo-msc/+/30114 ) Change subject: add codec_mapping.h,c ...................................................................... add codec_mapping.h,c Converting between different codec representations is confusing. This codec mapping provides a consolidated overview of all our codec representations, and how they match up. In particular, it adds the SDP codec representation repertoire, preparing the use of full SDP on the MNCC interface. Related: SYS#5066 Change-Id: Iaa307be6a8487aa8d4ba7cd59d5c5ef04818a744 --- M include/osmocom/msc/Makefile.am A include/osmocom/msc/codec_mapping.h M src/libmsc/Makefile.am A src/libmsc/codec_mapping.c 4 files changed, 604 insertions(+), 0 deletions(-) Approvals: laforge: Looks good to me, but someone else must approve pespin: Looks good to me, approved Jenkins Builder: Verified diff --git a/include/osmocom/msc/Makefile.am b/include/osmocom/msc/Makefile.am index adda44c..80f8972 100644 --- a/include/osmocom/msc/Makefile.am +++ b/include/osmocom/msc/Makefile.am @@ -1,6 +1,7 @@ noinst_HEADERS = \ call_leg.h \ cell_id_list.h \ + codec_mapping.h \ db.h \ debug.h \ e_link.h \ diff --git a/include/osmocom/msc/codec_mapping.h b/include/osmocom/msc/codec_mapping.h new file mode 100644 index 0000000..1232d2d --- /dev/null +++ b/include/osmocom/msc/codec_mapping.h @@ -0,0 +1,64 @@ +/* Routines for translation between codec representations: SDP, CC/BSSMAP variants, MGCP, MNCC */ +#pragma once + +#include <osmocom/gsm/protocol/gsm_04_08.h> +#include <osmocom/gsm/protocol/gsm_08_08.h> +#include <osmocom/mgcp_client/mgcp_client.h> +#include <osmocom/msc/sdp_msg.h> +#include <osmocom/gsm/mncc.h> + +#define NO_MGCP_CODEC 0xffffffff + +extern const struct gsm_mncc_bearer_cap bearer_cap_empty; + +enum codec_frhr { + CODEC_FRHR_NONE = 0, + CODEC_FRHR_FR, + CODEC_FRHR_HR, +}; + +struct codec_mapping { + /* The sdp.payload_type number in a mapping is not necessarily imperative, but may just reflect the usual + * payload type number for a given codec. */ + struct sdp_audio_codec sdp; + /* The id that mgcp_client.h uses for this codec. Must be set in each mapping, because 0 means PCMU. */ + enum mgcp_codecs mgcp; + /* Nr of used entries in speech_ver[] below. */ + unsigned int speech_ver_count; + /* Entries to add to Speech Version lists when this codec is present, if any. */ + enum gsm48_bcap_speech_ver speech_ver[8]; + /* If applicable, one of GSM_TCHF_FRAME, GSM_TCHF_FRAME_EFR, GSM_TCHH_FRAME, GSM_TCH_FRAME_AMR; or zero. */ + uint32_t mncc_payload_msg_type; + /* Set to true if gsm0808_speech_codec below reflects a meaningful value. */ + bool has_gsm0808_speech_codec; + struct gsm0808_speech_codec gsm0808_speech_codec; + /* If applicable, entries to add to Permitted Speech lists when this codec is present; or zero. */ + enum gsm0808_permitted_speech perm_speech; + /* If applicable, indicator whether this codec can work on a GERAN half-rate lchan, or whether full-rate is + * required. Leave zero when this codec does not apply to GERAN. */ + enum codec_frhr frhr; +}; + +const struct codec_mapping *codec_mapping_by_speech_ver(enum gsm48_bcap_speech_ver speech_ver); +const struct codec_mapping *codec_mapping_by_gsm0808_speech_codec_type(enum gsm0808_speech_codec_type sct); +const struct codec_mapping *codec_mapping_by_gsm0808_speech_codec(const struct gsm0808_speech_codec *sc); +const struct codec_mapping *codec_mapping_by_perm_speech(enum gsm0808_permitted_speech perm_speech); +const struct codec_mapping *codec_mapping_by_subtype_name(const char *subtype_name); +const struct codec_mapping *codec_mapping_by_mgcp_codec(enum mgcp_codecs mgcp); + +int bearer_cap_add_speech_ver(struct gsm_mncc_bearer_cap *bearer_cap, enum gsm48_bcap_speech_ver speech_ver); +int sdp_audio_codec_add_to_bearer_cap(struct gsm_mncc_bearer_cap *bearer_cap, const struct sdp_audio_codec *codec); +int sdp_audio_codecs_to_bearer_cap(struct gsm_mncc_bearer_cap *bearer_cap, const struct sdp_audio_codecs *ac); +int bearer_cap_set_radio(struct gsm_mncc_bearer_cap *bearer_cap); + +struct sdp_audio_codec *sdp_audio_codecs_add_speech_ver(struct sdp_audio_codecs *ac, + enum gsm48_bcap_speech_ver speech_ver); +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); + +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); + +int sdp_audio_codecs_to_gsm0808_channel_type(struct gsm0808_channel_type *ct, const struct sdp_audio_codecs *ac); + +enum mgcp_codecs sdp_audio_codec_to_mgcp_codec(const struct sdp_audio_codec *codec); diff --git a/src/libmsc/Makefile.am b/src/libmsc/Makefile.am index de02a17..c7fe4bb 100644 --- a/src/libmsc/Makefile.am +++ b/src/libmsc/Makefile.am @@ -27,6 +27,7 @@ libmsc_a_SOURCES = \ call_leg.c \ cell_id_list.c \ + codec_mapping.c \ sccp_ran.c \ msc_vty.c \ db.c \ diff --git a/src/libmsc/codec_mapping.c b/src/libmsc/codec_mapping.c new file mode 100644 index 0000000..69b8ed1 --- /dev/null +++ b/src/libmsc/codec_mapping.c @@ -0,0 +1,521 @@ +/* Routines for translation between codec representations: SDP, CC/BSSMAP variants, MGCP, MNCC */ +/* + * (C) 2019-2023 by sysmocom - s.m.f.c. GmbH <info(a)sysmocom.de> + * All Rights Reserved + * + * Author: Neels Hofmeyr + * + * SPDX-License-Identifier: AGPL-3.0+ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU Affero General Public License as published by + * the Free Software Foundation; either version 3 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU Affero General Public License for more details. + * + * You should have received a copy of the GNU Affero General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ +#include <string.h> + +#include <osmocom/gsm/mncc.h> + +#include <osmocom/msc/sdp_msg.h> +#include <osmocom/msc/codec_mapping.h> +#include <osmocom/msc/mncc.h> + +const struct codec_mapping codec_map[] = { + /* FIXME: sdp.fmtp handling is not done properly yet, proper mode-set and octet-align handling will follow in + * separate patches. */ + { + .sdp = { + .payload_type = 0, + .subtype_name = "PCMU", + .rate = 8000, + }, + .mgcp = CODEC_PCMU_8000_1, + }, + { + .sdp = { + .payload_type = 3, + .subtype_name = "GSM", + .rate = 8000, + }, + .mgcp = CODEC_GSM_8000_1, + .speech_ver_count = 1, + .speech_ver = { GSM48_BCAP_SV_FR }, + .mncc_payload_msg_type = GSM_TCHF_FRAME, + .has_gsm0808_speech_codec = true, + .gsm0808_speech_codec = { + .fi = true, + .type = GSM0808_SCT_FR1, + }, + .perm_speech = GSM0808_PERM_FR1, + .frhr = CODEC_FRHR_FR, + }, + { + .sdp = { + .payload_type = 8, + .subtype_name = "PCMA", + .rate = 8000, + }, + .mgcp = CODEC_PCMA_8000_1, + }, + { + .sdp = { + .payload_type = 18, + .subtype_name = "G729", + .rate = 8000, + }, + .mgcp = CODEC_G729_8000_1, + }, + { + .sdp = { + .payload_type = 110, + .subtype_name = "GSM-EFR", + .rate = 8000, + }, + .mgcp = CODEC_GSMEFR_8000_1, + .speech_ver_count = 1, + .speech_ver = { GSM48_BCAP_SV_EFR }, + .mncc_payload_msg_type = GSM_TCHF_FRAME_EFR, + .has_gsm0808_speech_codec = true, + .gsm0808_speech_codec = { + .fi = true, + .type = GSM0808_SCT_FR2, + }, + .perm_speech = GSM0808_PERM_FR2, + .frhr = CODEC_FRHR_FR, + }, + { + .sdp = { + .payload_type = 111, + .subtype_name = "GSM-HR-08", + .rate = 8000, + }, + .mgcp = CODEC_GSMHR_8000_1, + .speech_ver_count = 1, + .speech_ver = { GSM48_BCAP_SV_HR }, + .mncc_payload_msg_type = GSM_TCHH_FRAME, + .has_gsm0808_speech_codec = true, + .gsm0808_speech_codec = { + .fi = true, + .type = GSM0808_SCT_HR1, + .cfg = GSM0808_SC_CFG_DEFAULT_HR_AMR, + }, + .perm_speech = GSM0808_PERM_HR1, + .frhr = CODEC_FRHR_HR, + }, + { + .sdp = { + .payload_type = 112, + .subtype_name = "AMR", + .rate = 8000, + /* AMR is always octet-aligned in 2G and 3G RAN, so this fmtp is signalled to remote call legs. + * So far, fmtp is ignored in incoming SIP SDP, so an incoming SDP without 'octet-align=1' will + * match with this entry; we will still reply with 'octet-align=1', which often works out. */ + .fmtp = "octet-align=1", + }, + .mgcp = CODEC_AMR_8000_1, + .speech_ver_count = 1, + .speech_ver = { GSM48_BCAP_SV_AMR_F }, + .mncc_payload_msg_type = GSM_TCH_FRAME_AMR, + .has_gsm0808_speech_codec = true, + .gsm0808_speech_codec = { + .fi = true, + .type = GSM0808_SCT_FR3, + .cfg = GSM0808_SC_CFG_DEFAULT_FR_AMR, + }, + .perm_speech = GSM0808_PERM_FR3, + .frhr = CODEC_FRHR_FR, + }, + { + /* Another entry like the above, to map HR3 to AMR, too. */ + .sdp = { + .payload_type = 112, + .subtype_name = "AMR", + .rate = 8000, + .fmtp = "octet-align=1", + }, + .mgcp = CODEC_AMR_8000_1, + .speech_ver_count = 2, + .speech_ver = { GSM48_BCAP_SV_AMR_H, GSM48_BCAP_SV_AMR_OH }, + .mncc_payload_msg_type = GSM_TCH_FRAME_AMR, + .has_gsm0808_speech_codec = true, + .gsm0808_speech_codec = { + .fi = true, + .type = GSM0808_SCT_HR3, + .cfg = GSM0808_SC_CFG_DEFAULT_HR_AMR, + }, + .perm_speech = GSM0808_PERM_HR3, + .frhr = CODEC_FRHR_HR, + }, + { + .sdp = { + .payload_type = 113, + .subtype_name = "AMR-WB", + .rate = 16000, + .fmtp = "octet-align=1", + }, + .mgcp = CODEC_AMRWB_16000_1, + .speech_ver_count = 2, + .speech_ver = { GSM48_BCAP_SV_AMR_OFW, GSM48_BCAP_SV_AMR_FW }, + .mncc_payload_msg_type = GSM_TCH_FRAME_AMR, + .has_gsm0808_speech_codec = true, + .gsm0808_speech_codec = { + .fi = true, + .type = GSM0808_SCT_FR5, + .cfg = GSM0808_SC_CFG_DEFAULT_FR_AMR_WB, + }, + .perm_speech = GSM0808_PERM_FR5, + .frhr = CODEC_FRHR_FR, + }, + { + /* Another entry like the above, to map HR4 to AMR-WB, too. */ + .sdp = { + .payload_type = 113, + .subtype_name = "AMR-WB", + .rate = 16000, + .fmtp = "octet-align=1", + }, + .mgcp = CODEC_AMRWB_16000_1, + .speech_ver_count = 1, + .speech_ver = { GSM48_BCAP_SV_AMR_OHW }, + .mncc_payload_msg_type = GSM_TCH_FRAME_AMR, + .has_gsm0808_speech_codec = true, + .gsm0808_speech_codec = { + .fi = true, + .type = GSM0808_SCT_HR4, + .cfg = GSM0808_SC_CFG_DEFAULT_OHR_AMR_WB, + }, + .perm_speech = GSM0808_PERM_HR4, + .frhr = CODEC_FRHR_HR, + }, + { + .sdp = { + .payload_type = 96, + .subtype_name = "VND.3GPP.IUFP", + .rate = 16000, + }, + .mgcp = CODEC_IUFP, + }, +}; + +#define foreach_codec_mapping(CODEC_MAPPING) \ + for ((CODEC_MAPPING) = codec_map; (CODEC_MAPPING) < codec_map + ARRAY_SIZE(codec_map); (CODEC_MAPPING)++) + +const struct gsm_mncc_bearer_cap bearer_cap_empty = { + .speech_ver = { -1 }, + }; + +const struct codec_mapping *codec_mapping_by_speech_ver(enum gsm48_bcap_speech_ver speech_ver) +{ + const struct codec_mapping *m; + foreach_codec_mapping(m) { + int i; + for (i = 0; i < m->speech_ver_count; i++) + if (m->speech_ver[i] == speech_ver) + return m; + } + return NULL; +} + +const struct codec_mapping *codec_mapping_by_gsm0808_speech_codec_type(enum gsm0808_speech_codec_type sct) +{ + const struct codec_mapping *m; + foreach_codec_mapping(m) { + if (!m->has_gsm0808_speech_codec) + continue; + if (m->gsm0808_speech_codec.type == sct) + return m; + } + return NULL; +} + +const struct codec_mapping *codec_mapping_by_gsm0808_speech_codec(const struct gsm0808_speech_codec *sc) +{ + const struct codec_mapping *m; + foreach_codec_mapping(m) { + if (!m->has_gsm0808_speech_codec) + continue; + if (m->gsm0808_speech_codec.type != sc->type) + continue; + /* Return only those where sc->cfg is a subset of m->gsm0808_speech_codec.cfg. */ + if ((m->gsm0808_speech_codec.cfg & sc->cfg) != sc->cfg) + continue; + return m; + } + return NULL; +} + +const struct codec_mapping *codec_mapping_by_perm_speech(enum gsm0808_permitted_speech perm_speech) +{ + const struct codec_mapping *m; + foreach_codec_mapping(m) { + if (m->perm_speech == perm_speech) + return m; + } + return NULL; +} + +const struct codec_mapping *codec_mapping_by_subtype_name(const char *subtype_name) +{ + const struct codec_mapping *m; + foreach_codec_mapping(m) { + if (!strcmp(m->sdp.subtype_name, subtype_name)) + return m; + } + return NULL; +} + +const struct codec_mapping *codec_mapping_by_mgcp_codec(enum mgcp_codecs mgcp) +{ + const struct codec_mapping *m; + foreach_codec_mapping(m) { + if (m->mgcp == mgcp) + return m; + } + return NULL; +} + +/* Append given Speech Version to the end of the Bearer Capabilities Speech Version array. Return 1 if added, zero + * otherwise (as in, return the number of items added). */ +int bearer_cap_add_speech_ver(struct gsm_mncc_bearer_cap *bearer_cap, enum gsm48_bcap_speech_ver speech_ver) +{ + int i; + for (i = 0; i < ARRAY_SIZE(bearer_cap->speech_ver) - 1; i++) { + if (bearer_cap->speech_ver[i] == speech_ver) + return 0; + if (bearer_cap->speech_ver[i] == -1) { + bearer_cap->speech_ver[i] = speech_ver; + bearer_cap->speech_ver[i+1] = -1; + return 1; + } + } + return 0; +} + +/* From the current speech_ver list present in the bearer_cap, set the bearer_cap.radio. + * If a HR speech_ver is present, set to GSM48_BCAP_RRQ_DUAL_FR, otherwise set to GSM48_BCAP_RRQ_FR_ONLY. */ +int bearer_cap_set_radio(struct gsm_mncc_bearer_cap *bearer_cap) +{ + bool hr_present = false; + int i; + for (i = 0; i < ARRAY_SIZE(bearer_cap->speech_ver) - 1; i++) { + const struct codec_mapping *m; + + if (bearer_cap->speech_ver[i] == -1) + break; + + m = codec_mapping_by_speech_ver(bearer_cap->speech_ver[i]); + + if (!m) + continue; + + if (m->frhr == CODEC_FRHR_HR) + hr_present = true; + } + + if (hr_present) + bearer_cap->radio = GSM48_BCAP_RRQ_DUAL_FR; + else + bearer_cap->radio = GSM48_BCAP_RRQ_FR_ONLY; + + return 0; +} + +/* Try to convert the SDP audio codec name to Speech Versions to append to Bearer Capabilities. + * Return the number of Speech Version entries added (some may add more than one, others may be unknown/unapplicable and + * return 0). */ +int sdp_audio_codec_add_to_bearer_cap(struct gsm_mncc_bearer_cap *bearer_cap, const struct sdp_audio_codec *codec) +{ + const struct codec_mapping *m; + int added = 0; + foreach_codec_mapping(m) { + int i; + if (strcmp(m->sdp.subtype_name, codec->subtype_name)) + continue; + /* TODO also match rate and fmtp? */ + for (i = 0; i < m->speech_ver_count; i++) + added += bearer_cap_add_speech_ver(bearer_cap, m->speech_ver[i]); + } + return added; +} + +/* Append all audio codecs found in given sdp_msg to Bearer Capability, by traversing all codec entries with + * sdp_audio_codec_add_to_bearer_cap(). Return the number of Speech Version entries added. + * Note that Speech Version entries are only appended, no previous entries are removed. + * Note that only the Speech Version entries are modified; to make a valid Bearer Capabiliy, at least bearer_cap->radio + * must also be set (before or after this function); see also bearer_cap_set_radio(). */ +int sdp_audio_codecs_to_bearer_cap(struct gsm_mncc_bearer_cap *bearer_cap, const struct sdp_audio_codecs *ac) +{ + const struct sdp_audio_codec *codec; + int added = 0; + + foreach_sdp_audio_codec(codec, ac) { + added += sdp_audio_codec_add_to_bearer_cap(bearer_cap, codec); + } + + return added; +} + +/* Convert Speech Version to SDP audio codec and append to SDP message struct. */ +struct sdp_audio_codec *sdp_audio_codecs_add_speech_ver(struct sdp_audio_codecs *ac, + enum gsm48_bcap_speech_ver speech_ver) +{ + const struct codec_mapping *m; + struct sdp_audio_codec *ret = NULL; + foreach_codec_mapping(m) { + int i; + for (i = 0; i < m->speech_ver_count; i++) { + if (m->speech_ver[i] == speech_ver) { + ret = sdp_audio_codecs_add_copy(ac, &m->sdp); + break; + } + } + } + return ret; +} + +struct sdp_audio_codec *sdp_audio_codecs_add_mgcp_codec(struct sdp_audio_codecs *ac, enum mgcp_codecs mgcp_codec) +{ + const struct codec_mapping *m = codec_mapping_by_mgcp_codec(mgcp_codec); + if (!m) + return NULL; + return sdp_audio_codecs_add_copy(ac, &m->sdp); +} + +void sdp_audio_codecs_from_bearer_cap(struct sdp_audio_codecs *ac, const struct gsm_mncc_bearer_cap *bc) +{ + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(bc->speech_ver); i++) { + if (bc->speech_ver[i] == -1) + break; + sdp_audio_codecs_add_speech_ver(ac, bc->speech_ver[i]); + } +} + +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; + + *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)) + break; + scl->codec[scl->len] = m->gsm0808_speech_codec; + /* FIXME: apply AMR configuration according to codec->fmtp */ + scl->len++; + } +} + +void sdp_audio_codecs_from_speech_codec_list(struct sdp_audio_codecs *ac, const struct gsm0808_speech_codec_list *cl) +{ + int i; + for (i = 0; i < cl->len; i++) { + const struct gsm0808_speech_codec *sc = &cl->codec[i]; + const struct codec_mapping *m = codec_mapping_by_gsm0808_speech_codec(sc); + if (!m) + continue; + sdp_audio_codecs_add_copy(ac, &m->sdp); + /* FIXME: for AMR, apply sc->cfg to the added codec's fmtp */ + } +} + +int sdp_audio_codecs_to_gsm0808_channel_type(struct gsm0808_channel_type *ct, const struct sdp_audio_codecs *ac) +{ + const struct sdp_audio_codec *codec; + bool fr_present = false; + int first_fr_idx = -1; + bool hr_present = false; + int first_hr_idx = -1; + int idx = -1; + + *ct = (struct gsm0808_channel_type){ + .ch_indctr = GSM0808_CHAN_SPEECH, + }; + + foreach_sdp_audio_codec(codec, ac) { + const struct codec_mapping *m; + int i; + bool dup; + idx++; + foreach_codec_mapping(m) { + if (strcmp(m->sdp.subtype_name, codec->subtype_name)) + continue; + + switch (m->perm_speech) { + default: + continue; + + case GSM0808_PERM_FR1: + case GSM0808_PERM_FR2: + case GSM0808_PERM_FR3: + case GSM0808_PERM_FR4: + case GSM0808_PERM_FR5: + fr_present = true; + if (first_fr_idx < 0) + first_fr_idx = idx; + break; + + case GSM0808_PERM_HR1: + case GSM0808_PERM_HR2: + case GSM0808_PERM_HR3: + case GSM0808_PERM_HR4: + case GSM0808_PERM_HR6: + hr_present = true; + if (first_hr_idx < 0) + first_hr_idx = idx; + break; + } + + /* Avoid duplicates */ + dup = false; + for (i = 0; i < ct->perm_spch_len; i++) { + if (ct->perm_spch[i] == m->perm_speech) { + dup = true; + break; + } + } + if (dup) + continue; + + ct->perm_spch[ct->perm_spch_len] = m->perm_speech; + ct->perm_spch_len++; + } + } + + if (fr_present && hr_present) { + if (first_fr_idx <= first_hr_idx) + ct->ch_rate_type = GSM0808_SPEECH_FULL_PREF; + else + ct->ch_rate_type = GSM0808_SPEECH_HALF_PREF; + } else if (fr_present && !hr_present) + ct->ch_rate_type = GSM0808_SPEECH_FULL_BM; + else if (!fr_present && hr_present) + ct->ch_rate_type = GSM0808_SPEECH_HALF_LM; + else + return -EINVAL; + return 0; +} + +enum mgcp_codecs sdp_audio_codec_to_mgcp_codec(const struct sdp_audio_codec *codec) +{ + const struct codec_mapping *m; + foreach_codec_mapping(m) { + if (!sdp_audio_codec_cmp(&m->sdp, codec, false, false)) + return m->mgcp; + } + return NO_MGCP_CODEC; +} -- To view, visit https://gerrit.osmocom.org/c/osmo-msc/+/30114 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmo-msc Gerrit-Branch: master Gerrit-Change-Id: Iaa307be6a8487aa8d4ba7cd59d5c5ef04818a744 Gerrit-Change-Number: 30114 Gerrit-PatchSet: 9 Gerrit-Owner: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: laforge <laforge(a)osmocom.org> Gerrit-Reviewer: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: pespin <pespin(a)sysmocom.de> Gerrit-CC: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-MessageType: merged

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[S] Change in osmo-msc[master]: [codecs filter] add trans.cc.codecs
by neels 14 Mar '23

14 Mar '23

neels has submitted this change. ( https://gerrit.osmocom.org/c/osmo-msc/+/30117 ) Change subject: [codecs filter] add trans.cc.codecs ...................................................................... [codecs filter] add trans.cc.codecs Add the central codecs_filter for Call Control. The new member is not used in this patch yet, subsequent patches will start to populate the various stages of this codec filter, one by one. Related: SYS#5066 Change-Id: Ib3fdeff8d1e1ea0760168d63ee6e1b1fb993aa5f --- M include/osmocom/msc/transaction.h 1 file changed, 17 insertions(+), 0 deletions(-) Approvals: laforge: Looks good to me, approved Jenkins Builder: Verified diff --git a/include/osmocom/msc/transaction.h b/include/osmocom/msc/transaction.h index 14b89bb..0966627 100644 --- a/include/osmocom/msc/transaction.h +++ b/include/osmocom/msc/transaction.h @@ -8,6 +8,7 @@ #include <osmocom/msc/mncc.h> #include <osmocom/msc/msc_a.h> #include <osmocom/msc/debug.h> +#include <osmocom/msc/codec_filter.h> #include <osmocom/gsm/gsm0411_smc.h> #include <osmocom/gsm/gsm0411_smr.h> @@ -101,6 +102,8 @@ struct gsm_mncc msg; /* stores setup/disconnect/release message */ bool mncc_initiated; /* Whether an MNCC Release is necessary on failure */ struct osmo_lcls *lcls; + /* Track codec choices from BSS and remote call leg */ + struct codec_filter codecs; } cc; struct { struct gsm411_smc_inst smc_inst; -- To view, visit https://gerrit.osmocom.org/c/osmo-msc/+/30117 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmo-msc Gerrit-Branch: master Gerrit-Change-Id: Ib3fdeff8d1e1ea0760168d63ee6e1b1fb993aa5f Gerrit-Change-Number: 30117 Gerrit-PatchSet: 8 Gerrit-Owner: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: laforge <laforge(a)osmocom.org> Gerrit-Reviewer: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: pespin <pespin(a)sysmocom.de> Gerrit-MessageType: merged

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[L] Change in osmo-msc[master]: rtp_stream: allow multiple codecs / use codec filter from Assignment
by neels 14 Mar '23

14 Mar '23

Attention is currently required from: pespin. neels has posted comments on this change. ( https://gerrit.osmocom.org/c/osmo-msc/+/30125 ) Change subject: rtp_stream: allow multiple codecs / use codec filter from Assignment ...................................................................... Patch Set 7: Code-Review+2 (2 comments) Patchset: PS7: combine votes File src/libmsc/call_leg.c: https://gerrit.osmocom.org/c/osmo-msc/+/30125/comment/591d3a90_ab67c855 PS1, Line 354: struct sdp_audio_codecs *codecs; > This belongs to another patch IIUC according to the commit description (rename codec to codecs). Done -- To view, visit https://gerrit.osmocom.org/c/osmo-msc/+/30125 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmo-msc Gerrit-Branch: master Gerrit-Change-Id: If9c67b298b30f893ec661f84c9fc622ad01b5ee5 Gerrit-Change-Number: 30125 Gerrit-PatchSet: 7 Gerrit-Owner: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: laforge <laforge(a)osmocom.org> Gerrit-Reviewer: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: pespin <pespin(a)sysmocom.de> Gerrit-Attention: pespin <pespin(a)sysmocom.de> Gerrit-Comment-Date: Tue, 14 Mar 2023 22:40:43 +0000 Gerrit-HasComments: Yes Gerrit-Has-Labels: Yes Comment-In-Reply-To: pespin <pespin(a)sysmocom.de> Gerrit-MessageType: comment

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[XL] Change in osmo-upf[master]: add unique_ids_test.c
by neels 14 Mar '23

14 Mar '23

neels has posted comments on this change. ( https://gerrit.osmocom.org/c/osmo-upf/+/31482 ) Change subject: add unique_ids_test.c ...................................................................... Patch Set 4: (1 comment) Patchset: PS4: AFAICT last chance to fix this is https://gerrit.osmocom.org/c/libosmo-pfcp/+/31741 else i'll have to add LIBOSMOGSM_* to the unique_ids_test Makefile.am -- To view, visit https://gerrit.osmocom.org/c/osmo-upf/+/31482 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmo-upf Gerrit-Branch: master Gerrit-Change-Id: I36acff15f22d23ade4d281c2af3eb117dfc10359 Gerrit-Change-Number: 31482 Gerrit-PatchSet: 4 Gerrit-Owner: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-CC: laforge <laforge(a)osmocom.org> Gerrit-Comment-Date: Tue, 14 Mar 2023 22:38:07 +0000 Gerrit-HasComments: Yes Gerrit-Has-Labels: No Gerrit-MessageType: comment

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[M] Change in libosmocore[master]: New osmo-gsmtap-logsend utility
by keith 14 Mar '23

14 Mar '23

Attention is currently required from: neels, laforge. keith has posted comments on this change. ( https://gerrit.osmocom.org/c/libosmocore/+/31916 ) Change subject: New osmo-gsmtap-logsend utility ...................................................................... Patch Set 4: (1 comment) File utils/gsmtap-logsend.c: https://gerrit.osmocom.org/c/libosmocore/+/31916/comment/1f13107e_e5bae635 PS2, Line 18: #include <string.h> : #include <stdlib.h> : #include <string.h> : #include <fcntl.h> > apparently these 4 #includes are not needed. […] tested that, and removed them. -- To view, visit https://gerrit.osmocom.org/c/libosmocore/+/31916 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: libosmocore Gerrit-Branch: master Gerrit-Change-Id: Ida96e87d84e0c349c5069edc67fec1c3cf19d1ab Gerrit-Change-Number: 31916 Gerrit-PatchSet: 4 Gerrit-Owner: laforge <laforge(a)osmocom.org> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: keith <keith(a)rhizomatica.org> Gerrit-Reviewer: neels <nhofmeyr(a)sysmocom.de> Gerrit-Attention: neels <nhofmeyr(a)sysmocom.de> Gerrit-Attention: laforge <laforge(a)osmocom.org> Gerrit-Comment-Date: Tue, 14 Mar 2023 22:32:10 +0000 Gerrit-HasComments: Yes Gerrit-Has-Labels: No Comment-In-Reply-To: neels <nhofmeyr(a)sysmocom.de> Gerrit-MessageType: comment

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[S] Change in libosmo-pfcp[master]: build: add gen__pfcp_ies_auto_LDFLAGS = -no-install
by fixeria 14 Mar '23

14 Mar '23

Attention is currently required from: neels. fixeria has posted comments on this change. ( https://gerrit.osmocom.org/c/libosmo-pfcp/+/31918 ) Change subject: build: add gen__pfcp_ies_auto_LDFLAGS = -no-install ...................................................................... Patch Set 1: Code-Review+2 -- To view, visit https://gerrit.osmocom.org/c/libosmo-pfcp/+/31918 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: libosmo-pfcp Gerrit-Branch: master Gerrit-Change-Id: I4c1e1eed90acc3466262750e235f03e8e00105b6 Gerrit-Change-Number: 31918 Gerrit-PatchSet: 1 Gerrit-Owner: neels <nhofmeyr(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-Attention: neels <nhofmeyr(a)sysmocom.de> Gerrit-Comment-Date: Tue, 14 Mar 2023 22:31:29 +0000 Gerrit-HasComments: No Gerrit-Has-Labels: Yes Gerrit-MessageType: comment

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