mgcp/rtp: Refactored packet_duration computation
Since the packet duration is given in ms with the 'ptime' RTP media
attribute and also with the 'p' MGCP local connection option, the
computation is changed to use this value (if present). The
computation assumes, that there are N complete frames in a packet and
takes into account, that the ptime value possibly had been rounded
towards the next ms value (which is never the case with a frame length
of exact 20ms).
Sponsored-by: On-Waves ehf
diff --git a/openbsc/src/libmgcp/mgcp_network.c b/openbsc/src/libmgcp/mgcp_network.c
index 6463ae0..bf205d1 100644
--- a/openbsc/src/libmgcp/mgcp_network.c
+++ b/openbsc/src/libmgcp/mgcp_network.c
@@ -237,10 +237,7 @@
state->initialized = 1;
state->jitter = 0;
state->transit = arrival_time - timestamp;
- state->packet_duration =
- rtp_end->rate * rtp_end->frames_per_packet *
- rtp_end->frame_duration_num /
- rtp_end->frame_duration_den;
+ state->packet_duration = mgcp_rtp_packet_duration(endp, rtp_end);
state->out_stream = state->in_stream;
state->out_stream.last_timestamp = timestamp;
state->out_stream.ssrc = ssrc - 1; /* force output SSRC change */
diff --git a/openbsc/src/libmgcp/mgcp_protocol.c b/openbsc/src/libmgcp/mgcp_protocol.c
index b8e1ecd..ab94164 100644
--- a/openbsc/src/libmgcp/mgcp_protocol.c
+++ b/openbsc/src/libmgcp/mgcp_protocol.c
@@ -75,7 +75,7 @@
/* Assume audio frame length of 20ms */
#define DEFAULT_RTP_AUDIO_FRAME_DUR_NUM 20
#define DEFAULT_RTP_AUDIO_FRAME_DUR_DEN 1000
-#define DEFAULT_RTP_AUDIO_FRAMES_PER_PACKET 1
+#define DEFAULT_RTP_AUDIO_PACKET_DURATION_MS 20
#define DEFAULT_RTP_AUDIO_DEFAULT_RATE 8000
static void mgcp_rtp_end_reset(struct mgcp_rtp_end *end);
@@ -631,6 +631,23 @@
rtp->force_constant_ssrc ? ", force constant ssrc" : "");
}
+uint32_t mgcp_rtp_packet_duration(struct mgcp_endpoint *endp,
+ struct mgcp_rtp_end *rtp)
+{
+ int f = 0;
+
+ /* Get the number of frames per channel and packet */
+ if (rtp->frames_per_packet)
+ f = rtp->frames_per_packet;
+ else if (rtp->packet_duration_ms && rtp->frame_duration_num) {
+ int den = 1000 * rtp->frame_duration_num;
+ f = (rtp->packet_duration_ms * rtp->frame_duration_den + den/2)
+ / den;
+ }
+
+ return rtp->rate * f * rtp->frame_duration_num / rtp->frame_duration_den;
+}
+
static struct msgb *handle_create_con(struct mgcp_parse_data *p)
{
struct mgcp_trunk_config *tcfg;
@@ -1086,8 +1103,9 @@
/* Set default values */
end->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
end->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
- end->frames_per_packet = DEFAULT_RTP_AUDIO_FRAMES_PER_PACKET;
- end->rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE;
+ end->frames_per_packet = 0; /* unknown */
+ end->packet_duration_ms = DEFAULT_RTP_AUDIO_PACKET_DURATION_MS;
+ end->rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE;
}
static void mgcp_rtp_end_init(struct mgcp_rtp_end *end)