diff options
author | Neels Hofmeyr <neels@hofmeyr.de> | 2019-10-21 03:24:11 +0200 |
---|---|---|
committer | Neels Hofmeyr <neels@hofmeyr.de> | 2020-01-06 18:00:40 +0100 |
commit | f31a1ccd9a3eb474936f5b946287581514b29436 (patch) | |
tree | ec59ae18c6e54b3530bf2f56e09d99bc23390b0d /src/libmsc/msc_t.c | |
parent | 02dd265d68b771bf315cfe6620c9b2371edea828 (diff) |
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
Diffstat (limited to 'src/libmsc/msc_t.c')
-rw-r--r-- | src/libmsc/msc_t.c | 7 |
1 files changed, 5 insertions, 2 deletions
diff --git a/src/libmsc/msc_t.c b/src/libmsc/msc_t.c index af0ddaaef..413ffd10d 100644 --- a/src/libmsc/msc_t.c +++ b/src/libmsc/msc_t.c @@ -454,9 +454,12 @@ static int msc_t_patch_and_send_ho_request_ack(struct msc_t *msc_t, const struct if (r->codec_present) { LOG_MSC_T(msc_t, LOGL_DEBUG, "From Handover Request Ack, got %s\n", osmo_mgcpc_codec_name(r->codec)); - rtp_stream_set_codec(rtp_ran, r->codec); + if (!rtp_stream_set_codecs_from_mgcp_codec(rtp_ran, r->codec)) { + LOG_MSC_T(msc_t, LOGL_ERROR, "Cannot resolve codec in Handover Request Ack: %s\n", + osmo_mgcpc_codec_name(r->codec)); + } if (rtp_cn) - rtp_stream_set_codec(rtp_cn, r->codec); + rtp_stream_set_codecs_from_mgcp_codec(rtp_cn, r->codec); } else { LOG_MSC_T(msc_t, LOGL_DEBUG, "No codec in Handover Request Ack\n"); } |