neels has submitted this change. ( 
https://gerrit.osmocom.org/c/osmo-msc/+/15962 )

Change subject: cc trans: remove unused tch_rtp_create
......................................................................

cc trans: remove unused tch_rtp_create

Use of this flag was dropped when adding inter-BSC and inter-MSC Handover
support, I forgot to remove it.

Change-Id: I5ec78e30eb36fbe78a3f7c46bfa44af5a4eb7bf2
---
M include/osmocom/msc/transaction.h
M src/libmsc/gsm_04_08_cc.c
2 files changed, 0 insertions(+), 18 deletions(-)

Approvals:
  Jenkins Builder: Verified
  pespin: Looks good to me, approved



diff --git a/include/osmocom/msc/transaction.h 
b/include/osmocom/msc/transaction.h
index 61d8c1a..928b137 100644
--- a/include/osmocom/msc/transaction.h
+++ b/include/osmocom/msc/transaction.h
@@ -87,10 +87,6 @@
        /* bearer capabilities (rate and codec) */
        struct gsm_mncc_bearer_cap bearer_cap;

-       /* if true, TCH_RTP_CREATE is sent after the
-        * assignment is done */
-       bool tch_rtp_create;
-
        union {
                struct {

diff --git a/src/libmsc/gsm_04_08_cc.c b/src/libmsc/gsm_04_08_cc.c
index 792ab61..7e2d70b 100644
--- a/src/libmsc/gsm_04_08_cc.c
+++ b/src/libmsc/gsm_04_08_cc.c
@@ -1655,20 +1655,6 @@
        }
        LOG_TRANS_CAT(trans, DMNCC, LOGL_DEBUG, "rx %s\n", 
get_mncc_name(MNCC_RTP_CREATE));

-       /* When we call msc_mgcp_call_assignment() we will trigger, depending
-        * on the RAN type the call assignment on the A or Iu interface.
-        * msc_mgcp_call_assignment() also takes care about sending the CRCX
-        * command to the MGCP-GW. The CRCX will return the port number,
-        * where the PBX (e.g. Asterisk) will send its RTP stream to. We
-        * have to return this port number back to the MNCC by sending
-        * it back with the TCH_RTP_CREATE message. To make sure that
-        * this message is sent AFTER the response to CRCX from the
-        * MGCP-GW has arrived, we need will instruct msc_mgcp_call_assignment()
-        * to take care of this by setting trans->tch_rtp_create to true.
-        * This will make sure that gsm48_tch_rtp_create() (below) is
-        * called as soon as the local port number has become known. */
-       trans->tch_rtp_create = true;
-
        /* Assign call (if not done yet) */
        return msc_a_try_call_assignment(trans);
 }

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Gerrit-Project: osmo-msc
Gerrit-Branch: master
Gerrit-Change-Id: I5ec78e30eb36fbe78a3f7c46bfa44af5a4eb7bf2
Gerrit-Change-Number: 15962
Gerrit-PatchSet: 1
Gerrit-Owner: neels <nhofm...@sysmocom.de>
Gerrit-Reviewer: Jenkins Builder
Gerrit-Reviewer: neels <nhofm...@sysmocom.de>
Gerrit-Reviewer: pespin <pes...@sysmocom.de>
Gerrit-MessageType: merged

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