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authorHarald Welte <laforge@gnumonks.org>2011-09-07 11:55:08 +0200
committerHarald Welte <laforge@gnumonks.org>2011-09-07 11:55:08 +0200
commit5355ead9d99262b756bbe736f8feb3e189264ca8 (patch)
tree47e5780132f55a983fbfc9fe532e796396ecdd0a /src
parent42f80bc7afcbad2b2a6c44ccf224265812fca3ba (diff)
use osmo_ortp in POLLING mode instead of select loop integration
This solves a lot of the problems we've been seeing in the context of large jitter (uRTP) or classic RTP with SID frames
Diffstat (limited to 'src')
-rw-r--r--src/common/rsl.c3
-rw-r--r--src/osmo-bts-sysmo/l1_if.c25
2 files changed, 24 insertions, 4 deletions
diff --git a/src/common/rsl.c b/src/common/rsl.c
index 3dbfda9c..1ccd2080 100644
--- a/src/common/rsl.c
+++ b/src/common/rsl.c
@@ -1264,7 +1264,8 @@ static int rsl_rx_ipac_XXcx(struct msgb *msg)
}
/* FIXME: select default value depending on speech_mode */
//if (!payload_type)
- lchan->abis_ip.rtp_socket = osmo_rtp_socket_create(lchan->ts->trx);
+ lchan->abis_ip.rtp_socket = osmo_rtp_socket_create(lchan->ts->trx,
+ OSMO_RTP_F_POLL);
if (!lchan->abis_ip.rtp_socket) {
LOGP(DRSL, LOGL_ERROR,
"%s IPAC Failed to create RTP/RTCP sockets\n",
diff --git a/src/osmo-bts-sysmo/l1_if.c b/src/osmo-bts-sysmo/l1_if.c
index 01f146c6..fcef1db7 100644
--- a/src/osmo-bts-sysmo/l1_if.c
+++ b/src/osmo-bts-sysmo/l1_if.c
@@ -35,6 +35,8 @@
#include <osmocom/gsm/gsm_utils.h>
#include <osmocom/gsm/lapdm.h>
+#include <osmocom/trau/osmo_ortp.h>
+
#include <osmo-bts/logging.h>
#include <osmo-bts/bts.h>
#include <osmo-bts/oml.h>
@@ -215,7 +217,6 @@ get_lapdm_chan_by_hl2(struct gsm_bts_trx *trx, uint32_t hLayer2)
return &lchan->lapdm_ch;
}
-
static const uint8_t fill_frame[GSM_MACBLOCK_LEN] = {
0x01, 0x2B, 0x2B, 0x2B, 0x2B, 0x2B, 0x2B, 0x2B, 0x2B, 0x2B,
0x2B, 0x2B, 0x2B, 0x2B, 0x2B, 0x2B, 0x2B, 0x2B, 0x2B, 0x2B,
@@ -258,10 +259,28 @@ static int handle_ph_readytosend_ind(struct femtol1_hdl *fl1,
if (!lchan)
break;
+ if (lchan->abis_ip.rtp_socket) {
+ osmo_rtp_socket_poll(lchan->abis_ip.rtp_socket);
+ /* FIXME: we _assume_ that we never miss TDMA
+ * frames and that we always get to this point
+ * for every to-be-transmitted voice frame. A
+ * better solution would be to compute
+ * rx_user_ts based on how many TDMA frames have
+ * elapsed since the last call */
+ lchan->abis_ip.rtp_socket->rx_user_ts += GSM_RTP_DURATION;
+ }
/* get a msgb from the dl_tx_queue */
resp_msg = msgb_dequeue(&lchan->dl_tch_queue);
- if (!resp_msg)
- break;
+ /* if there is none, try to generate empty TCH frame
+ * like AMR SID_BAD */
+ if (!resp_msg) {
+ LOGP(DL1C, LOGL_NOTICE, "%s DL TCH Tx queue underrun\n",
+ gsm_lchan_name(lchan));
+ resp_msg = gen_empty_tch_msg(lchan);
+ /* if there really is none, break here and send empty */
+ if (!resp_msg)
+ break;
+ }
/* fill header */
data_req_from_rts_ind(msgb_l1prim(resp_msg), rts_ind);