implement re-assignment to match codecs
This is the last missing piece that allows osmo-msc to make good TFO
codecs choices.
Since the codec_filter, osmo-msc properly gathers codec options and
limitations. But the MO call leg still assigns a voice channel before
getting a response from the MT call leg, and is then stuck with that.
Add the capability to adjust the MO call leg's codec in case the MT side
needs a different codec for TFO.
This is only relevant for 2G; on 3G we always have AMR/IuUP.
For inter-MSC handover, keep the behavior unchanged: offer only the
currently assigned codec to the remote side. Codec-changing HO should be
equally trivial to implement, but that is for another day.
msc_vlr_test_call's codec tests are adjusted to test the new feature in
Ib933554f826c1b4347dfa3f6c4f6fe086be8b133. For now, avoid change in
these tests by validating the first codec in SDP lists only.
Related: OS#6258
Related: osmo-ttcn3-hacks I402ed0523a2a87b83f29c5577b2c828102005d53
Change-Id: I8760feaa8598047369ef8c3ab2673013bac8ac8a
diff --git a/src/libmsc/codec_filter.c b/src/libmsc/codec_filter.c
index a9d93a7..7511f90 100644
--- a/src/libmsc/codec_filter.c
+++ b/src/libmsc/codec_filter.c
@@ -98,46 +98,16 @@
if (remote->audio_codecs.count)
sdp_audio_codecs_intersection(r, &remote->audio_codecs, true);
-#if 0
- /* Future: If osmo-msc were able to trigger a re-assignment after the remote side has picked a codec mismatching
- * the initial Assignment, then this code here would make sense: keep the other codecs as available to choose
- * from, but put the currently assigned codec in the first position. So far we only offer the single assigned
- * codec, because we have no way to deal with the remote side picking a different codec.
- * Another approach would be to postpone assignment until we know the codecs from the remote side. */
if (sdp_audio_codec_is_set(a)) {
/* Assignment has completed, the chosen codec should be the first of the resulting SDP.
- * Make sure this is actually listed in the result SDP and move to first place. */
+ * If present, make sure this is listed in first place.
+ * If 'select' is NULL, the assigned codec is not present in the intersection of possible choices for
+ * TFO. Just omit the assigned codec from the filter result, and it is the CC code's responsibility to
+ * detect this and assign a working codec instead. */
struct sdp_audio_codec *select = sdp_audio_codecs_by_descr(r, a);
-
- if (!select) {
- /* Not present. Add. */
- if (sdp_audio_codec_by_payload_type(r, a->payload_type, false)) {
- /* Oh crunch, that payload type number is already in use.
- * Find an unused one. */
- for (a->payload_type = 96; a->payload_type <= 127; a->payload_type++) {
- if (!sdp_audio_codec_by_payload_type(r, a->payload_type, false))
- break;
- }
-
- if (a->payload_type > 127)
- return -ENOSPC;
- }
- select = sdp_audio_codecs_add_copy(r, a);
- }
-
- sdp_audio_codecs_select(r, select);
+ if (select)
+ 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;
}
diff --git a/src/libmsc/gsm_04_08_cc.c b/src/libmsc/gsm_04_08_cc.c
index fe34127..f6ec81b 100644
--- a/src/libmsc/gsm_04_08_cc.c
+++ b/src/libmsc/gsm_04_08_cc.c
@@ -270,7 +270,16 @@
break;
}
- if (sdp && sdp[0] && (sdp_msg_from_sdp_str(&sdp_msg, sdp) == 0)) {
+ if (sdp && sdp[0]) {
+ int rc = sdp_msg_from_sdp_str(&sdp_msg, sdp);
+ if (rc != 0) {
+ LOG_TRANS_CAT_SRC(trans, DMNCC, LOGL_ERROR, file, line, "%s %s: invalid SDP message (trying anyway)\n",
+ rx_tx,
+ get_mncc_name(mncc->msg_type));
+ LOG_TRANS_CAT_SRC(trans, DMNCC, LOGL_DEBUG, file, line, "erratic SDP: %s\n",
+ osmo_quote_cstr_c(OTC_SELECT, sdp, -1));
+ return;
+ }
LOG_TRANS_CAT_SRC(trans, DMNCC, LOGL_DEBUG, file, line, "%s %s (RTP=%s)\n",
rx_tx,
get_mncc_name(mncc->msg_type),
@@ -748,6 +757,7 @@
static void rx_mncc_sdp(struct gsm_trans *trans, uint32_t mncc_msg_type, const char *sdp,
const struct gsm_mncc_bearer_cap *bcap)
{
+ struct codec_filter *codecs = &trans->cc.codecs;
struct call_leg *cl = trans->msc_a ? trans->msc_a->cc.call_leg : NULL;
struct rtp_stream *rtp_cn = cl ? cl->rtp[RTP_TO_CN] : NULL;
@@ -775,6 +785,30 @@
rtp_stream_set_remote_addr_and_codecs(rtp_cn, &trans->cc.remote);
rtp_stream_commit(rtp_cn);
}
+
+ /* See if we need to switch codecs to maintain TFO: has the remote side changed the codecs information? If we
+ * have already assigned a specific codec here, but the remote call leg has now chosen a different codec, we
+ * need to re-assign this call leg to match the remote leg. */
+ if (!sdp_audio_codec_is_set(&codecs->assignment)) {
+ /* Voice channel assignment has not completed. Do not interfere. */
+ return;
+ }
+ if (!trans->cc.remote.audio_codecs.count) {
+ /* Don't know remote codecs, nothing to do. */
+ return;
+ }
+ if (sdp_audio_codecs_by_descr(&trans->cc.remote.audio_codecs, &codecs->assignment)) {
+ /* The assigned codec is part of the remote codec set. All is well. */
+ /* TODO: maybe this should require exactly the *first* remote codec to match, because we cannot flexibly
+ * transcode, and assume the actual payload we will receive is listed in the first place? */
+ return;
+ }
+
+ /* We've already completed Assignment of a voice channel (some time ago), and now the remote side has changed
+ * to a mismatching codec (list). Try to re-assign this side to a matching codec. */
+ LOG_TRANS(trans, LOGL_INFO, "Remote call leg mismatches assigned codec: %s\n",
+ codec_filter_to_str(&trans->cc.codecs, &trans->cc.local, &trans->cc.remote));
+ msc_a_tx_assignment_cmd(trans->msc_a);
}
static int gsm48_cc_tx_setup(struct gsm_trans *trans, void *arg)
@@ -2049,17 +2083,23 @@
switch (trans->cc.state) {
case GSM_CSTATE_INITIATED:
case GSM_CSTATE_MO_CALL_PROC:
- /* MO call */
+ /* MO call, send ACK in form of an MNCC_RTP_CREATE (below) */
break;
case GSM_CSTATE_CALL_RECEIVED:
case GSM_CSTATE_MO_TERM_CALL_CONF:
- /* MT call */
+ /* MT call, send ACK in form of an MNCC_RTP_CREATE (below) */
break;
case GSM_CSTATE_ACTIVE:
- /* already active. MNCC finished before Abis completed the Assignment. */
- break;
+ /* already active. We decided to re-assign later on during the call - at time of writing this never
+ * happens. */
+ case GSM_CSTATE_CALL_DELIVERED:
+ case GSM_CSTATE_CONNECT_IND:
+ /* MNCC has progressed past the initial assignment. Usually it means that this happened: after
+ * MNCC_ALERT_REQ, MO has triggered a re-assignment, to adjust MO's codec to MT's codec. */
+ LOG_TRANS(trans, LOGL_DEBUG, "Re-Assignment complete\n");
+ return 0;
default:
LOG_TRANS(trans, LOGL_ERROR, "Assignment done in unexpected CC state: %d\n", trans->cc.state);
diff --git a/src/libmsc/msc_a.c b/src/libmsc/msc_a.c
index e64b54d..a933bd2 100644
--- a/src/libmsc/msc_a.c
+++ b/src/libmsc/msc_a.c
@@ -636,7 +636,7 @@
}
/* The MGW has given us a local IP address for the RAN side. Ready to start the Assignment of a voice channel. */
-static void msc_a_call_leg_ran_local_addr_available(struct msc_a *msc_a)
+void msc_a_tx_assignment_cmd(struct msc_a *msc_a)
{
struct ran_msg msg;
struct gsm_trans *cc_trans = msc_a->cc.active_trans;
@@ -804,7 +804,7 @@
rtps->use_osmux ? "yes" : "no", rtps->local_osmux_cid);
switch (rtps->dir) {
case RTP_TO_RAN:
- msc_a_call_leg_ran_local_addr_available(msc_a);
+ msc_a_tx_assignment_cmd(msc_a);
return;
case RTP_TO_CN:
cc_on_cn_local_rtp_port_known(rtps->for_trans);
diff --git a/src/libmsc/msc_ho.c b/src/libmsc/msc_ho.c
index f826975..47f000b 100644
--- a/src/libmsc/msc_ho.c
+++ b/src/libmsc/msc_ho.c
@@ -380,7 +380,7 @@
struct vlr_subscr *vsub = msc_a_vsub(msc_a);
struct gsm_network *net = msc_a_net(msc_a);
struct gsm0808_channel_type channel_type;
- struct gsm0808_speech_codec_list scl;
+ struct gsm0808_speech_codec_list scl = {};
struct gsm_trans *cc_trans = msc_a->cc.active_trans;
struct ran_msg ran_enc_msg = {
.msg_type = RAN_MSG_HANDOVER_REQUEST,
@@ -442,7 +442,13 @@
ran_enc_msg.handover_request.call_id_present = true;
ran_enc_msg.handover_request.call_id = cc_trans->call_id;
- sdp_audio_codecs_to_speech_codec_list(&scl, &cc_trans->cc.local.audio_codecs);
+ /* Call assignment is now capable of re-assigning to overcome a codec mismatch with the remote call leg.
+ * But for inter-MSC handover, that is not supported yet. So keep here the old limitation of only
+ * offering the assigned codec. */
+ if (sdp_audio_codec_is_set(&cc_trans->cc.codecs.assignment))
+ sdp_audio_codec_to_speech_codec_list(&scl, &cc_trans->cc.codecs.assignment);
+ else
+ sdp_audio_codecs_to_speech_codec_list(&scl, &cc_trans->cc.local.audio_codecs);
if (!scl.len) {
msc_ho_failed(msc_a, GSM0808_CAUSE_EQUIPMENT_FAILURE, "Failed to compose"
" Codec List (MSC Preferred) for Handover Request message\n");