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Change subject: osmo_io: Clean up code
......................................................................
osmo_io: Clean up code
- Remove osmo_io_init() from header, since it has no function definition
- Add osmo_iofd_init() to header
Change-Id: I77f7ae2b211507f420d87c484ec75ee054fceb63
---
M include/osmocom/core/osmo_io.h
1 file changed, 13 insertions(+), 1 deletion(-)
git pull ssh://gerrit.osmocom.org:29418/libosmocore refs/changes/24/34524/2
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Change subject: mgw: TC_conn_timeout Fix timeout left set upon test exit
......................................................................
mgw: TC_conn_timeout Fix timeout left set upon test exit
The test mentioned in a comment that the value is reset upon each test
start, but that's false.
Change-Id: I9447fa6c6b3103e05fa3ee756ee572ddecb69006
---
M mgw/MGCP_Test.ttcn
1 file changed, 16 insertions(+), 1 deletion(-)
Approvals:
osmith: Looks good to me, but someone else must approve
fixeria: Looks good to me, approved
Jenkins Builder: Verified
diff --git a/mgw/MGCP_Test.ttcn b/mgw/MGCP_Test.ttcn
index 73591b2..0cedb0e 100644
--- a/mgw/MGCP_Test.ttcn
+++ b/mgw/MGCP_Test.ttcn
@@ -2562,7 +2562,7 @@
f_init(ep);
log("Setting conn-timeout to 1s");
- f_vty_config(MGWVTY, "mgcp", "conn-timeout 1"); /* reset in f_init_vty() */
+ f_vty_config(MGWVTY, "mgcp", "conn-timeout 1");
log("Sending RTP data for 1.5s");
flow := valueof(t_RtpFlow(mp_local_ipv4, mp_remote_ipv4, 111, "GSM-HR-08/8000/1"));
@@ -2587,6 +2587,9 @@
f_sleep(0.2);
f_rtpem_conn_refuse_verify(RTPEM[0]);
+ log("Setting conn-timeout back to 0 (disabled)");
+ f_vty_config(MGWVTY, "mgcp", "conn-timeout 0");
+
setverdict(pass);
}
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Change subject: stream_srv: Set io_fd pointer to NULL after freeing
......................................................................
Patch Set 1: Code-Review+1
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Change subject: ASCI: Add group receive and transmit mode support to MM layer
......................................................................
Patch Set 13:
(1 comment)
File src/host/layer23/src/mobile/gsm48_mm.c:
https://gerrit.osmocom.org/c/osmocom-bb/+/34492/comment/9783c735_a3e48771
PS10, Line 4854: // There are event handlers for group call in this implementation.
> Oops, I missed some words. I rephrased it.
v13 still hasn't this rephreased afaict.
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Change subject: Fix reception of MM messages at gsm48_mm_data_ind()
......................................................................
Patch Set 1: Code-Review+2
(1 comment)
File src/host/layer23/src/mobile/gsm48_mm.c:
https://gerrit.osmocom.org/c/osmocom-bb/+/34570/comment/6400f0c6_cdc8b1ce
PS1, Line 4091: switch (pdisc) {
Sounds like you probably want to put all the code from here to line 4155 to a subfunction to simplify this one and avoid the goto. Can be done in another patch.
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Hello Jenkins Builder, laforge, pespin,
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Verified+1 by Jenkins Builder
Change subject: ASCI: Add group receive and transmit mode support to MM layer
......................................................................
ASCI: Add group receive and transmit mode support to MM layer
Related: OS#5364
Change-Id: I05957182a57423ad947ab200b52f65fde859e110
---
M src/host/layer23/include/osmocom/bb/mobile/gsm48_mm.h
M src/host/layer23/src/mobile/gsm48_mm.c
2 files changed, 677 insertions(+), 77 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmocom-bb refs/changes/92/34492/13
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Hello Jenkins Builder, fixeria, pespin,
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Change subject: ASCI: Add protocol type to trans_find_by_callref() function
......................................................................
ASCI: Add protocol type to trans_find_by_callref() function
This is required, because different protocols may share the same
callref, but use different protocols. E.g. a voice group call can share
the same callref with a voice broadcast call, but these calls are
different transactions.
Related: OS#5364
Change-Id: Ifea3a81aae3b4ae897851f867b13fa987c8cbe11
---
M src/host/layer23/include/osmocom/bb/mobile/transaction.h
M src/host/layer23/src/mobile/gsm411_sms.c
M src/host/layer23/src/mobile/gsm480_ss.c
M src/host/layer23/src/mobile/gsm48_cc.c
M src/host/layer23/src/mobile/transaction.c
5 files changed, 22 insertions(+), 7 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmocom-bb refs/changes/93/34493/13
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Change subject: ASCI: Add VTY commands to control voice group/broadcast calls
......................................................................
ASCI: Add VTY commands to control voice group/broadcast calls
Related: OS#5364
Change-Id: Id32253b4e10b8df48e819d8a92bbcda332dd11e6
---
M src/host/layer23/include/osmocom/bb/mobile/vty.h
M src/host/layer23/src/mobile/vty_interface.c
2 files changed, 248 insertions(+), 8 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmocom-bb refs/changes/95/34495/14
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Change subject: ASCI: Show NCH position in VTY together with system information
......................................................................
ASCI: Show NCH position in VTY together with system information
Related: OS#5364
Change-Id: I5e0a9d469eb70608502dca881808621fa153b666
---
M src/host/layer23/src/common/sysinfo.c
1 file changed, 22 insertions(+), 0 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmocom-bb refs/changes/96/34496/14
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Change subject: ASCI: Add group receive and transmit mode support to MM layer
......................................................................
Patch Set 12:
(1 comment)
File src/host/layer23/src/mobile/gsm48_mm.c:
https://gerrit.osmocom.org/c/osmocom-bb/+/34492/comment/058305d5_3478639a
PS10, Line 4854: // There are event handlers for group call in this implementation.
> not sure what do you mean with this.
Oops, I missed some words. I rephrased it.
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osmith has submitted this change. ( https://gerrit.osmocom.org/c/osmo-msc/+/34559?usp=email )
Change subject: CC: don't start guard timer on mid-call MNCC messages
......................................................................
CC: don't start guard timer on mid-call MNCC messages
The intent of the guard timer is to clear hung or stuck states
during call setup or teardown. However, there are some MNCC
messages that will be exchanged between OsmoMSC (passing CC
messages to and from the MS) and the external MNCC agent during
the active call state, not related to setup or teardown: DTMF
start and stop, plus call hold and retrieve operations for call
waiting. Unpatched OsmoMSC restarts the guard timer on every
received MNCC message, even those that pass through to CC without
affecting any state, and the result is breakage for users.
Consider the case of an IVR where you have to press some DTMF keys
before you can be transferred to a human operator. You press the
needed keys, get the human operator, and start talking. Then
3 minutes into your conversion (default guard timer duration)
your call unceremoniously disconnects without any warning.
Fix: look at the MNCC message type, and skip the call to start
the guard timer for known-benign MNCC messages.
Change-Id: Ibe2dd53f8e9e06d175b64df67d2a2e3e2d4155aa
---
M src/libmsc/gsm_04_08_cc.c
1 file changed, 49 insertions(+), 1 deletion(-)
Approvals:
Jenkins Builder: Verified
diff --git a/src/libmsc/gsm_04_08_cc.c b/src/libmsc/gsm_04_08_cc.c
index 6148bbd..1e2c5af 100644
--- a/src/libmsc/gsm_04_08_cc.c
+++ b/src/libmsc/gsm_04_08_cc.c
@@ -2383,7 +2383,27 @@
log_mncc_rx_tx(trans, "rx", msg);
- gsm48_start_guard_timer(trans);
+ /*
+ * The step of gsm48_start_guard_timer() needs to be done for
+ * major state-impacting MNCC messages, but not for those
+ * that are a mere pass-through to CC messages to MS.
+ */
+ switch (msg->msg_type) {
+ case MNCC_PROGRESS_REQ:
+ case MNCC_NOTIFY_REQ:
+ case MNCC_FACILITY_REQ:
+ case MNCC_START_DTMF_RSP:
+ case MNCC_START_DTMF_REJ:
+ case MNCC_STOP_DTMF_RSP:
+ case MNCC_HOLD_CNF:
+ case MNCC_HOLD_REJ:
+ case MNCC_RETRIEVE_CNF:
+ case MNCC_RETRIEVE_REJ:
+ case MNCC_USERINFO_REQ:
+ break;
+ default:
+ gsm48_start_guard_timer(trans);
+ }
trans->cc.mncc_initiated = true;
if (trans->msc_a)
--
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osmith has submitted this change. ( https://gerrit.osmocom.org/c/osmo-msc/+/34557?usp=email )
Change subject: fix codecs in internal call bridge
......................................................................
fix codecs in internal call bridge
This is a fixup for the patch
'3G: decapsulate IuUP to AMR at the MGW; allow 3G<-AMR->2G'
I386a6a426c318040b019ab5541689c67e94672a1
After above patch, osmo-msc intelligently decides which codecs to run on
which legs of the RTP streams. In the meantime, it seems the necessary
matching changes to call_leg_local_bridge() had been lost somehow.
Testing 3G to 3G voice now, I noticed that call_leg_local_bridge()
overwrites the intelligent choices made earlier.
The history of an MGW endpoint that should convert from IUFP to plain
AMR, extracted from a pcap, looks like this:
<- CRCX None None
-> CRCX-OK audio 4050 RTP/AVP 112 None
<- MDCX audio 4056 RTP/AVP 112 AMR
-> MDCX-OK audio 4050 RTP/AVP 112 AMR
<- MDCX audio 4056 RTP/AVP 96 VND.3GPP.IUFP
-> MDCX-OK audio 4050 RTP/AVP 96 VND.3GPP.IUFP
So after call_leg_local_bridge(), there is an extra MDCX + MDCX-OK that
switches the codec from 112 AMR back to 96 IUFP.
That is because call_leg_local_bridge() copies the *RAN* side's codec to
both CN sides, which used to be ok when RAN and CN codecs were always
identical.
Instead, adjust only the CN sides of the MGW endpoints, and adjust them
so that both CN sides are identical. osmo-mgw should then be able to
trivially translate the codecs appropriately.
Change-Id: I130bcd77ec57e332370c487a11b0b973b6e1089d
---
M src/libmsc/call_leg.c
1 file changed, 73 insertions(+), 11 deletions(-)
Approvals:
Jenkins Builder: Verified
diff --git a/src/libmsc/call_leg.c b/src/libmsc/call_leg.c
index d0dc642..b797322 100644
--- a/src/libmsc/call_leg.c
+++ b/src/libmsc/call_leg.c
@@ -352,25 +352,46 @@
int call_leg_local_bridge(struct call_leg *cl1, uint32_t call_id1, struct gsm_trans *trans1,
struct call_leg *cl2, uint32_t call_id2, struct gsm_trans *trans2)
{
- struct sdp_audio_codecs *codecs;
+ struct sdp_audio_codecs *cn_codecs = NULL;
cl1->local_bridge = cl2;
cl2->local_bridge = cl1;
- /* We may just copy the codec info we have for the RAN side of the first leg to the CN side of both legs. This
- * also means that if both legs use different codecs the MGW must perform transcoding on the second leg. */
- if (!cl1->rtp[RTP_TO_RAN] || !cl1->rtp[RTP_TO_RAN]->codecs_known) {
- LOG_CALL_LEG(cl1, LOGL_ERROR, "RAN-side RTP stream codec is not known, not ready for bridging\n");
+ /* Marry the two CN sides of the call legs. Call establishment should have made all efforts for these to be
+ * compatible. However, for local bridging, the codecs and payload type numbers must be exactly identical on
+ * both sides. Both sides may so far have different payload type numbers or slightly differing codecs, but it
+ * will only work when the SDP on the RTP_TO_CN sides of the call legs talk the same payload type numbers.
+ * So, simply take the SDP from one RTP_TO_CN side, and overwrite the other RTP_TO_CN side's SDP with it.
+ * If all goes to plan, the codecs will be identical, or possibly the MGW will do a conversion like AMR-BE to
+ * AMR-OA. In the worst case, the other call leg cannot transcode, and the call fails -- because codec
+ * negotiation did not do a good enough job.
+ *
+ * Copy one call leg's CN config to the other:
+ *
+ * call leg 1 call leg 2
+ * ---MGW-ep------- ---MGW-ep-------
+ * RAN CN CN RAN
+ * AMR:112 AMR:112 AMR:96 AMR:96
+ * |
+ * +-------+
+ * |
+ * V
+ * AMR:112 AMR:112 AMR:112 AMR:96
+ * ^MGW-endpoint converts payload type numbers between 112 and 96.
+ */
+ if (cl1->rtp[RTP_TO_CN] && cl1->rtp[RTP_TO_CN]->codecs_known)
+ cn_codecs = &cl1->rtp[RTP_TO_CN]->codecs;
+ else if (cl2->rtp[RTP_TO_CN] && cl2->rtp[RTP_TO_CN]->codecs_known)
+ cn_codecs = &cl2->rtp[RTP_TO_CN]->codecs;
+ if (!cn_codecs) {
+ LOG_CALL_LEG(cl1, LOGL_ERROR, "RAN-side CN stream codec is not known, not ready for bridging\n");
+ LOG_CALL_LEG(cl2, LOGL_ERROR, "RAN-side CN stream codec is not known, not ready for bridging\n");
return -EINVAL;
}
- codecs = &cl1->rtp[RTP_TO_RAN]->codecs;
-
- if (!cl1->rtp[RTP_TO_CN] || !cl2->rtp[RTP_TO_CN])
- return -ENOTCONN;
call_leg_ensure_ci(cl1, RTP_TO_CN, call_id1, trans1,
- codecs, &cl2->rtp[RTP_TO_CN]->local);
+ cn_codecs, &cl2->rtp[RTP_TO_CN]->local);
call_leg_ensure_ci(cl2, RTP_TO_CN, call_id2, trans2,
- codecs, &cl1->rtp[RTP_TO_CN]->local);
+ cn_codecs, &cl1->rtp[RTP_TO_CN]->local);
return 0;
}
--
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Change subject: gsup_client_mux: set destination_name in error reply function
......................................................................
gsup_client_mux: set destination_name in error reply function
If the GSUP request message to which we are replying is an MT SMS
delivery from an SMSC relayed via OsmoHLR, we must set destination_name
in our reply - otherwise our reply won't make it back to the SMSC.
Related: OS#6135
Change-Id: I892fe87a733a78ed9d5761a8ce238caa135dea1e
---
M src/libmsc/gsup_client_mux.c
1 file changed, 16 insertions(+), 0 deletions(-)
Approvals:
Jenkins Builder: Verified
diff --git a/src/libmsc/gsup_client_mux.c b/src/libmsc/gsup_client_mux.c
index 1900377..9a0dc16 100644
--- a/src/libmsc/gsup_client_mux.c
+++ b/src/libmsc/gsup_client_mux.c
@@ -150,6 +150,8 @@
.cause = cause,
.message_type = OSMO_GSUP_TO_MSGT_ERROR(gsup_orig->message_type),
.message_class = gsup_orig->message_class,
+ .destination_name = gsup_orig->source_name,
+ .destination_name_len = gsup_orig->source_name_len,
/* RP-Message-Reference is mandatory for SM Service */
.sm_rp_mr = gsup_orig->sm_rp_mr,
--
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Change subject: mgcp-client: Fix missing include in mgcp_client_pool.h
......................................................................
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Change subject: mgw: Configure IuUP if codec set during MDCX
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Change subject: Bump version: 1.12.0.2-23174 → 1.12.1
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Change subject: ASCI: Remove duplicated CLEAR COMMAND from VGCS/VBS channel handling
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Change subject: SMS over GSUP: set source_name in GSUP reply messages
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Change subject: Bump version: 1.11.0.7-dbe88 → 1.11.1
......................................................................
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Change subject: SMS over GSUP: correctly route GSUP responses to MT SMS
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Change subject: gsup_client_mux: set destination_name in error reply function
......................................................................
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Change subject: CC: don't start guard timer on mid-call MNCC messages
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Change subject: make two functions static
......................................................................
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Change subject: fix codecs in internal call bridge
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Change subject: mgw: Configure IuUP if codec set during MDCX
......................................................................
mgw: Configure IuUP if codec set during MDCX
The mgcp client may first configure the connection to use RTP-AMR, but
after setting up another call leg may find out that both legs are IuUP
and hence want to forward the IuUP between the 2 connections instead.
In that case, an MDCX with codec VND.3GPP.IUFP would be set.
Until now, osmo-mgw didn't take that scenario into account, and it was
only upgrading the rtp conn to iuup internally during CRCX.
As a result, in the mentioned scenario osmo-mgw would continue to
output RTP instead of IuUP after the MDCX with VND.3GPP.IUFP, which is
wrong.
Related: SYS#6578
Change-Id: Ic94bf90f54d8ba3e65a2cd52734867847f3a60c2
(cherry picked from commit 58d5b97831a9307737606809eeb013533dd9afb7)
---
M src/libosmo-mgcp/mgcp_protocol.c
1 file changed, 26 insertions(+), 0 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmo-mgw refs/changes/66/34566/1
diff --git a/src/libosmo-mgcp/mgcp_protocol.c b/src/libosmo-mgcp/mgcp_protocol.c
index 7795734..809622b 100644
--- a/src/libosmo-mgcp/mgcp_protocol.c
+++ b/src/libosmo-mgcp/mgcp_protocol.c
@@ -1293,6 +1293,10 @@
error_code = rc;
goto error3;
}
+ /* Upgrade the conn type RTP_DEFAULT->RTP_IUUP if needed based on requested codec: */
+ /* TODO: "codec" probably needs to be moved from endp to conn */
+ if (conn->type == MGCP_RTP_DEFAULT && strcmp(conn->end.codec->subtype_name, "VND.3GPP.IUFP") == 0)
+ rc = mgcp_conn_iuup_init(conn);
/* check connection mode setting */
if (conn->conn->mode != MGCP_CONN_LOOPBACK
--
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Change subject: mgcp-client: Fix missing include in mgcp_client_pool.h
......................................................................
mgcp-client: Fix missing include in mgcp_client_pool.h
struct vty is used as a param but it is never defined.
Change-Id: Ia27bb20a79966cb90e04720462d24a236a168ada
(cherry picked from commit c415ed511372ba909e95e33845c65804ac50a182)
---
M include/osmocom/mgcp_client/mgcp_client_pool.h
1 file changed, 14 insertions(+), 0 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmo-mgw refs/changes/65/34565/1
diff --git a/include/osmocom/mgcp_client/mgcp_client_pool.h b/include/osmocom/mgcp_client/mgcp_client_pool.h
index ba31fb0..726eea1 100644
--- a/include/osmocom/mgcp_client/mgcp_client_pool.h
+++ b/include/osmocom/mgcp_client/mgcp_client_pool.h
@@ -2,6 +2,8 @@
#include <stdbool.h>
+#include <osmocom/vty/vty.h>
+
struct mgcp_client;
struct mgcp_client_pool;
struct mgcp_client_pool_member;
--
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Change subject: gsup_client_mux: set destination_name in error reply function
......................................................................
gsup_client_mux: set destination_name in error reply function
If the GSUP request message to which we are replying is an MT SMS
delivery from an SMSC relayed via OsmoHLR, we must set destination_name
in our reply - otherwise our reply won't make it back to the SMSC.
Related: OS#6135
Change-Id: I892fe87a733a78ed9d5761a8ce238caa135dea1e
---
M src/libmsc/gsup_client_mux.c
1 file changed, 16 insertions(+), 0 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmo-msc refs/changes/61/34561/1
diff --git a/src/libmsc/gsup_client_mux.c b/src/libmsc/gsup_client_mux.c
index 1900377..9a0dc16 100644
--- a/src/libmsc/gsup_client_mux.c
+++ b/src/libmsc/gsup_client_mux.c
@@ -150,6 +150,8 @@
.cause = cause,
.message_type = OSMO_GSUP_TO_MSGT_ERROR(gsup_orig->message_type),
.message_class = gsup_orig->message_class,
+ .destination_name = gsup_orig->source_name,
+ .destination_name_len = gsup_orig->source_name_len,
/* RP-Message-Reference is mandatory for SM Service */
.sm_rp_mr = gsup_orig->sm_rp_mr,
--
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Change subject: CC: don't start guard timer on mid-call MNCC messages
......................................................................
CC: don't start guard timer on mid-call MNCC messages
The intent of the guard timer is to clear hung or stuck states
during call setup or teardown. However, there are some MNCC
messages that will be exchanged between OsmoMSC (passing CC
messages to and from the MS) and the external MNCC agent during
the active call state, not related to setup or teardown: DTMF
start and stop, plus call hold and retrieve operations for call
waiting. Unpatched OsmoMSC restarts the guard timer on every
received MNCC message, even those that pass through to CC without
affecting any state, and the result is breakage for users.
Consider the case of an IVR where you have to press some DTMF keys
before you can be transferred to a human operator. You press the
needed keys, get the human operator, and start talking. Then
3 minutes into your conversion (default guard timer duration)
your call unceremoniously disconnects without any warning.
Fix: look at the MNCC message type, and skip the call to start
the guard timer for known-benign MNCC messages.
Change-Id: Ibe2dd53f8e9e06d175b64df67d2a2e3e2d4155aa
---
M src/libmsc/gsm_04_08_cc.c
1 file changed, 49 insertions(+), 1 deletion(-)
git pull ssh://gerrit.osmocom.org:29418/osmo-msc refs/changes/59/34559/1
diff --git a/src/libmsc/gsm_04_08_cc.c b/src/libmsc/gsm_04_08_cc.c
index 6148bbd..1e2c5af 100644
--- a/src/libmsc/gsm_04_08_cc.c
+++ b/src/libmsc/gsm_04_08_cc.c
@@ -2383,7 +2383,27 @@
log_mncc_rx_tx(trans, "rx", msg);
- gsm48_start_guard_timer(trans);
+ /*
+ * The step of gsm48_start_guard_timer() needs to be done for
+ * major state-impacting MNCC messages, but not for those
+ * that are a mere pass-through to CC messages to MS.
+ */
+ switch (msg->msg_type) {
+ case MNCC_PROGRESS_REQ:
+ case MNCC_NOTIFY_REQ:
+ case MNCC_FACILITY_REQ:
+ case MNCC_START_DTMF_RSP:
+ case MNCC_START_DTMF_REJ:
+ case MNCC_STOP_DTMF_RSP:
+ case MNCC_HOLD_CNF:
+ case MNCC_HOLD_REJ:
+ case MNCC_RETRIEVE_CNF:
+ case MNCC_RETRIEVE_REJ:
+ case MNCC_USERINFO_REQ:
+ break;
+ default:
+ gsm48_start_guard_timer(trans);
+ }
trans->cc.mncc_initiated = true;
if (trans->msc_a)
--
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