2019-10-07 19:14:02 +00:00
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# Generated by msc_log_to_ladder.py
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msc {
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hscale="3";
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moms[label="MS,BSS (MO)\nUE,hNodeB (MO)"],momgw[label="MGW for MSC (MO)"],momsc[label="MSC (MO)"],sip[label="MNCC to PBX via\nosmo-sip-connector"],mtmsc[label="MSC (MT)"],mtmgw[label="MGW for MSC (MT)"],mtms[label="BSS,MS (MT)\nhNodeB,UE (MT)"];
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moms =>> momsc [label="MM CM_SERV_REQ"];
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moms <<= momsc [label="MM AUTH_REQ"];
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moms =>> momsc [label="MM AUTH_RESP"];
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moms <<= momsc [label="(BSSMAP) CIPHER_MODE_COMMAND"];
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moms =>> momsc [label="(BSSMAP) Ciphering Mode Complete"];
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moms =>> momsc [label="RR CIPH_M_COMPL"];
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moms =>> momsc [label="CC SETUP"];
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momsc note momsc [label="CC starts guard timer (180s)"];
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momsc abox momsc [label="CC state:\nINITIATED"];
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add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
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momgw <<= momsc [label="for CN: CRCX\nrtpbridge/*@msc"];
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momgw =>> momsc [label="for CN: CRCX OK\nEP-1 CI-1"];
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momgw <<= momsc [label="for RAN: CRCX\nEP-1"];
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2019-10-07 19:14:02 +00:00
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momsc =>> sip [label="MNCC_SETUP_IND"];
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add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
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momgw =>> momsc [label="for RAN: CRCX OK\nEP-1 CI-2"];
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2019-10-07 19:14:02 +00:00
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moms <<= momsc [label="(BSSMAP) ASSIGNMENT_COMMAND"];
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add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
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momsc <<= sip [label="MNCC_RTP_CREATE"];
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2019-10-07 19:14:02 +00:00
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moms =>> momsc [label="(BSSMAP) Assignment Complete"];
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add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
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momgw <<= momsc [label="for RAN: MDCX\nEP-1 CI-2"];
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2019-10-07 19:14:02 +00:00
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momsc =>> sip [label="MNCC_RTP_CREATE\nIP:port-1"];
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momsc <<= sip [label="MNCC_CALL_PROC_REQ"];
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momsc note momsc [label="CC stops guard timer"];
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momsc note momsc [label="CC starts guard timer (180s)"];
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momsc abox momsc [label="CC state:\nMO_CALL_PROC"];
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moms <<= momsc [label="CC CALL_PROC"];
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add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
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momgw =>> momsc [label="for RAN: MDCX OK\nEP-1 CI-2"];
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2019-10-07 19:14:02 +00:00
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mtmsc <<= sip [label="MNCC_SETUP_REQ"];
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mtms <<= mtmsc [label="Paging"];
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mtms =>> mtmsc [label="RR PAG_RESP"];
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mtms <<= mtmsc [label="MM AUTH_REQ"];
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mtms =>> mtmsc [label="MM NULL"];
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mtms =>> mtmsc [label="MM AUTH_RESP"];
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mtms <<= mtmsc [label="(RANAP) SecurityModeCommand"];
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mtms =>> mtmsc [label="(RANAP) SecurityModeControl successfulOutcome"];
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mtms <<= mtmsc [label="(RANAP) CommonId"];
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mtmsc abox mtmsc [label="CC state:\nCALL_PRESENT"];
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add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
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mtmsc note mtmsc [label="CC starts timer T303 (30s)"];
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2019-10-07 19:14:02 +00:00
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mtms <<= mtmsc [label="CC SETUP"];
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mtms =>> mtmsc [label="CC CALL_CONF"];
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mtmsc note mtmsc [label="CC stops timer T303"];
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mtmsc note mtmsc [label="CC starts timer T310 (30s)"];
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add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
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mtmgw <<= mtmsc [label="for CN: CRCX\nrtpbridge/*@msc"];
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2019-10-07 19:14:02 +00:00
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mtmsc abox mtmsc [label="CC state:\nMO_TERM_CALL_CONF"];
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mtmsc =>> sip [label="MNCC_CALL_CONF_IND"];
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mtmsc <<= sip [label="MNCC_RTP_CREATE"];
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add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
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mtmgw =>> mtmsc [label="for CN: CRCX OK\nEP-2 CI-3"];
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mtmgw <<= mtmsc [label="for RAN: CRCX\nEP-2"];
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mtmsc =>> sip [label="MNCC_RTP_CREATE\nIP:port-2"];
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mtmgw =>> mtmsc [label="for RAN: CRCX OK\nEP-2 CI-4"];
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2019-10-07 19:14:02 +00:00
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mtms <<= mtmsc [label="(RANAP) RAB AssignmentRequest"];
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mtms =>> mtmsc [label="(RANAP) RAB Assignment Response"];
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add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
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mtmgw <<= mtmsc [label="for RAN: MDCX\nEP-2 CI-4"];
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2019-10-07 19:14:02 +00:00
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mtmsc =>> sip [label="MNCC_RTP_CREATE\nIP:port-2"];
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add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
|
|
|
mtmgw =>> mtmsc [label="for RAN: MDCX OK\nEP-2 CI-4"];
|
2019-10-07 19:14:02 +00:00
|
|
|
mtms =>> mtmsc [label="CC ALERTING"];
|
|
|
|
mtmsc note mtmsc [label="CC stops timer T310"];
|
|
|
|
mtmsc note mtmsc [label="CC starts timer T301 (180s)"];
|
|
|
|
mtmsc abox mtmsc [label="CC state:\nCALL_RECEIVED"];
|
|
|
|
mtmsc =>> sip [label="MNCC_ALERT_IND"];
|
|
|
|
momsc <<= sip [label="MNCC_ALERT_REQ"];
|
|
|
|
momsc note momsc [label="CC stops guard timer"];
|
|
|
|
momsc note momsc [label="CC starts guard timer (180s)"];
|
|
|
|
momsc abox momsc [label="CC state:\nCALL_DELIVERED"];
|
|
|
|
moms <<= momsc [label="CC ALERTING"];
|
|
|
|
mtms =>> mtmsc [label="CC CONNECT"];
|
|
|
|
mtmsc note mtmsc [label="CC stops timer T301"];
|
|
|
|
mtmsc abox mtmsc [label="CC state:\nCONNECT_REQUEST"];
|
|
|
|
mtmsc =>> sip [label="MNCC_SETUP_CNF"];
|
add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
|
|
|
mtmsc <<= sip [label="MNCC_RTP_CONNECT"];
|
|
|
|
mtmgw <<= mtmsc [label="for CN: MDCX\nEP-2 CI-3"];
|
2019-10-07 19:14:02 +00:00
|
|
|
mtmsc <<= sip [label="MNCC_SETUP_COMPL_REQ"];
|
|
|
|
mtmsc note mtmsc [label="CC starts guard timer (180s)"];
|
|
|
|
mtmsc abox mtmsc [label="CC state:\nACTIVE"];
|
|
|
|
mtmsc note mtmsc [label="CC stops guard timer"];
|
|
|
|
mtms <<= mtmsc [label="CC CONNECT_ACK"];
|
add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
|
|
|
mtmgw =>> mtmsc [label="for CN: MDCX OK\nEP-2 CI-3"];
|
|
|
|
momsc <<= sip [label="MNCC_RTP_CONNECT"];
|
|
|
|
momgw <<= momsc [label="for CN: MDCX\nEP-1 CI-1"];
|
2019-10-07 19:14:02 +00:00
|
|
|
momsc <<= sip [label="MNCC_SETUP_RSP"];
|
|
|
|
momsc note momsc [label="CC stops guard timer"];
|
|
|
|
momsc note momsc [label="CC starts guard timer (180s)"];
|
|
|
|
momsc note momsc [label="CC starts timer T313 (30s)"];
|
|
|
|
momsc abox momsc [label="CC state:\nCONNECT_IND"];
|
|
|
|
moms <<= momsc [label="CC CONNECT"];
|
add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
|
|
|
momgw =>> momsc [label="for CN: MDCX OK\nEP-1 CI-1"];
|
2019-10-07 19:14:02 +00:00
|
|
|
moms =>> momsc [label="CC CONNECT_ACK"];
|
|
|
|
momsc note momsc [label="CC stops timer T313"];
|
|
|
|
momsc abox momsc [label="CC state:\nACTIVE"];
|
|
|
|
momsc note momsc [label="CC stops guard timer"];
|
|
|
|
momsc =>> sip [label="MNCC_SETUP_COMPL_IND"];
|
add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
|
|
|
mtms =>> mtmsc [label="CC DISCONNECT"];
|
2019-10-07 19:14:02 +00:00
|
|
|
mtmsc abox mtmsc [label="CC state:\nDISCONNECT_IND"];
|
add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
|
|
|
mtmsc =>> sip [label="MNCC_DISC_IND"];
|
|
|
|
mtmsc <<= sip [label="MNCC_REL_REQ"];
|
|
|
|
mtmsc note mtmsc [label="CC starts guard timer (180s)"];
|
|
|
|
mtmsc note mtmsc [label="CC starts timer T308 (10s)"];
|
|
|
|
mtmsc abox mtmsc [label="CC state:\nRELEASE_REQ"];
|
|
|
|
mtms <<= mtmsc [label="CC RELEASE"];
|
|
|
|
momsc <<= sip [label="MNCC_DISC_REQ"];
|
|
|
|
momsc note momsc [label="CC starts guard timer (180s)"];
|
|
|
|
momsc note momsc [label="CC starts timer T306 (30s)"];
|
|
|
|
momsc abox momsc [label="CC state:\nDISCONNECT_IND"];
|
|
|
|
moms <<= momsc [label="CC DISCONNECT"];
|
|
|
|
moms =>> momsc [label="CC RELEASE"];
|
|
|
|
momsc note momsc [label="CC stops timer T306"];
|
|
|
|
moms <<= momsc [label="CC RELEASE_COMPL"];
|
2019-10-07 19:14:02 +00:00
|
|
|
momsc abox momsc [label="CC state:\nNULL"];
|
|
|
|
momsc note momsc [label="CC stops guard timer"];
|
|
|
|
moms <<= momsc [label="(BSSMAP) CLEAR_COMMAND"];
|
add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
|
|
|
momsc =>> sip [label="MNCC_REL_IND"];
|
2019-10-07 19:14:02 +00:00
|
|
|
moms =>> momsc [label="(BSSMAP) Clear Complete"];
|
add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
|
|
|
mtms =>> mtmsc [label="CC RELEASE_COMPL"];
|
|
|
|
mtmsc note mtmsc [label="CC stops timer T308"];
|
2019-10-07 19:14:02 +00:00
|
|
|
mtmsc abox mtmsc [label="CC state:\nNULL"];
|
|
|
|
mtmsc note mtmsc [label="CC stops guard timer"];
|
|
|
|
mtms <<= mtmsc [label="(RANAP) Iu Release"];
|
add full SDP codec information to the MNCC socket
This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.
To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.
The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.
Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.
Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.
Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)
Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f
2019-10-21 01:24:11 +00:00
|
|
|
mtmsc =>> sip [label="MNCC_REL_CNF"];
|
2019-10-07 19:14:02 +00:00
|
|
|
mtms =>> mtmsc [label="(RANAP) Iu Release successfulOutcome"];
|
|
|
|
moms =>> momsc [label="IMSI Detach"];
|
|
|
|
mtms =>> mtmsc [label="IMSI Detach"];
|
|
|
|
}
|