mgw: clean up codec negotiation (sdp)
The codec negotiation via SDP is currently in a neglected state. Also
osmo-mgw does some kind of codec decision wile the SDP is parsed, the
result is information for one codec, even when there are multiple codecs
negotiated. This is problematic because we loose all information about
alternate codecs while we parse. This should be untangled and the
information should be presevered. Also we are not really capable
picking a default. Wehen we do not supply any codec information (not
even LCO), then we should pick a sane default codec.
- separate the codec decision from the sdp parser and concentrate
codec related code in a separate c file
- add support for multiple codecs in one SDP negotiation
- do not initalize "magic" codec defaults during conn allocation
- do not allow invalid payload types, especially not 255. When
someone tries to select an invalid payload type, do not fail
hard, just pick a sane default.
- handle the codec decision in protocol.c, pick a sane default
codec when no (valid) codec has been negotiated (no LCO, no SDP)
Change-Id: If730d022ba6bdb217ad4e20b3fbbd1114dbb4b8f
Closes: OS#2658
Related: OS#3114
Related: OS#2728
diff --git a/src/libosmo-mgcp/mgcp_conn.c b/src/libosmo-mgcp/mgcp_conn.c
index 4926768..e49559c 100644
--- a/src/libosmo-mgcp/mgcp_conn.c
+++ b/src/libosmo-mgcp/mgcp_conn.c
@@ -25,6 +25,8 @@
#include <osmocom/mgcp/mgcp_internal.h>
#include <osmocom/mgcp/mgcp_common.h>
#include <osmocom/mgcp/mgcp_endp.h>
+#include <osmocom/mgcp/mgcp_sdp.h>
+#include <osmocom/mgcp/mgcp_codec.h>
#include <osmocom/gsm/gsm_utils.h>
#include <osmocom/core/rate_ctr.h>
#include <ctype.h>
@@ -88,22 +90,6 @@
return -1;
}
-/* Reset codec state and free memory */
-static void mgcp_rtp_codec_init(struct mgcp_rtp_codec *codec)
-{
- /* see also mgcp_sdp.c, mgcp_set_audio_info() */
- talloc_free(codec->subtype_name);
- talloc_free(codec->audio_name);
-
- codec->payload_type = -1;
- codec->subtype_name = NULL;
- codec->audio_name = NULL;
- codec->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
- codec->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
- codec->rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE;
- codec->channels = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS;
-}
-
/* Initialize rtp connection struct with default values */
static void mgcp_rtp_conn_init(struct mgcp_conn_rtp *conn_rtp, struct mgcp_conn *conn)
{
@@ -130,13 +116,15 @@
end->frames_per_packet = 0; /* unknown */
end->packet_duration_ms = DEFAULT_RTP_AUDIO_PACKET_DURATION_MS;
end->output_enabled = 0;
-
- mgcp_rtp_codec_init(&end->codec);
+ end->maximum_packet_time = -1;
conn_rtp->rate_ctr_group = rate_ctr_group_alloc(conn, &rate_ctr_group_desc, rate_ctr_index);
conn_rtp->state.in_stream.err_ts_ctr = &conn_rtp->rate_ctr_group->ctr[IN_STREAM_ERR_TSTMP_CTR];
conn_rtp->state.out_stream.err_ts_ctr = &conn_rtp->rate_ctr_group->ctr[OUT_STREAM_ERR_TSTMP_CTR];
rate_ctr_index++;
+
+ /* Make sure codec table is reset */
+ mgcp_codec_reset_all(conn_rtp);
}
/* Cleanup rtp connection struct */