static void create_transcoder(struct mgcp_endpoint *endp)
{
int port;
int in_endp = ENDPOINT_NUMBER(endp);
int out_endp = endp_back_channel(in_endp);
if (!endp->is_transcoded)
return;
send_msg(endp, in_endp, endp->trans_bts.local_port, "CRCX", "sendrecv");
send_msg(endp, in_endp, endp->trans_bts.local_port, "MDCX", "sendrecv");
send_msg(endp, out_endp, endp->trans_net.local_port, "CRCX", "sendrecv");
send_msg(endp, out_endp, endp->trans_net.local_port, "MDCX", "sendrecv");
port = rtp_calculate_port(in_endp, endp->cfg->transcoder_remote_base);
endp->trans_bts.rtp_port = htons(port);
endp->trans_bts.rtcp_port = htons(port + 1);
port = rtp_calculate_port(out_endp, endp->cfg->transcoder_remote_base);
endp->trans_net.rtp_port = htons(port);
endp->trans_net.rtcp_port = htons(port + 1);
}
/*
* Handle the assignment request message.
*
* See §3.2.1.1 for the message type
*/
static int bssmap_handle_assignm_req(struct osmo_bsc_sccp_con *conn,
struct msgb *msg, unsigned int length)
{
struct msgb *resp;
struct osmo_msc_data *msc;
struct tlv_parsed tp;
uint8_t *data;
uint8_t timeslot;
uint8_t multiplex;
enum gsm48_chan_mode chan_mode = GSM48_CMODE_SIGN;
int i, supported, port, full_rate = -1;
if (!conn->conn) {
LOGP(DMSC, LOGL_ERROR, "No lchan/msc_data in cipher mode command.\n");
return -1;
}
tlv_parse(&tp, gsm0808_att_tlvdef(), msg->l4h + 1, length - 1, 0, 0);
if (!TLVP_PRESENT(&tp, GSM0808_IE_CHANNEL_TYPE)) {
LOGP(DMSC, LOGL_ERROR, "Mandatory channel type not present.\n");
goto reject;
}
if (!TLVP_PRESENT(&tp, GSM0808_IE_CIRCUIT_IDENTITY_CODE)) {
LOGP(DMSC, LOGL_ERROR, "Identity code missing. Audio routing will not work.\n");
goto reject;
}
conn->cic = ntohs(read_data16(TLVP_VAL(&tp, GSM0808_IE_CIRCUIT_IDENTITY_CODE)));
timeslot = conn->cic & 0x1f;
multiplex = (conn->cic & ~0x1f) >> 5;
/*
* Currently we only support a limited subset of all
* possible channel types. The limitation ends by not using
* multi-slot, limiting the channel coding, speech...
*/
if (TLVP_LEN(&tp, GSM0808_IE_CHANNEL_TYPE) < 3) {
LOGP(DMSC, LOGL_ERROR, "ChannelType len !=3 not supported: %d\n",
TLVP_LEN(&tp, GSM0808_IE_CHANNEL_TYPE));
goto reject;
}
/*
* Try to figure out if we support the proposed speech codecs. For
* now we will always pick the full rate codecs.
*/
data = (uint8_t *) TLVP_VAL(&tp, GSM0808_IE_CHANNEL_TYPE);
if ((data[0] & 0xf) != 0x1) {
LOGP(DMSC, LOGL_ERROR, "ChannelType != speech: %d\n", data[0]);
goto reject;
}
/*
* go through the list of preferred codecs of our gsm network
* and try to find it among the permitted codecs. If we found
* it we will send chan_mode to the right mode and break the
* inner loop. The outer loop will exit due chan_mode having
* the correct value.
*/
full_rate = 0;
msc = conn->msc;
for (supported = 0;
chan_mode == GSM48_CMODE_SIGN && supported < msc->audio_length;
++supported) {
int perm_val = audio_support_to_gsm88(msc->audio_support[supported]);
for (i = 2; i < TLVP_LEN(&tp, GSM0808_IE_CHANNEL_TYPE); ++i) {
if ((data[i] & 0x7f) == perm_val) {
chan_mode = gsm88_to_chan_mode(perm_val);
full_rate = (data[i] & 0x4) == 0;
break;
} else if ((data[i] & 0x80) == 0x00) {
break;
}
}
}
if (chan_mode == GSM48_CMODE_SIGN) {
LOGP(DMSC, LOGL_ERROR, "No supported audio type found.\n");
goto reject;
}
/* map it to a MGCP Endpoint and a RTP port */
port = mgcp_timeslot_to_endpoint(multiplex, timeslot);
conn->rtp_port = rtp_calculate_port(port, msc->rtp_base);
return gsm0808_assign_req(conn->conn, chan_mode, full_rate);
reject:
resp = gsm0808_create_assignment_failure(GSM0808_CAUSE_NO_RADIO_RESOURCE_AVAILABLE, NULL);
if (!resp) {
LOGP(DMSC, LOGL_ERROR, "Channel allocation failure.\n");
return -1;
}
bsc_queue_for_msc(conn, resp);
return -1;
}