mgcp/rtp: Fix timestamp offset when patching RTP packets
The current implementation increments the seqno but does not increment
the RTP timestamp, leading to two identical timestamps following one
after the other.
This patch fixes this by adding the computed tsdelta when the offset
is calulated. In the unlikely case, that a tsdelta hasn't been
computed yet when the SSRC changes, a tsdelta is computed based on
the RTP rate and a RTP packet duration of 20ms (one speech frame per
channel and packet). If the RTP rate is not known, a rate of 8000 is
assumed.
Note that this approach presumes, that the per RTP packet duration
(in samples) is the same for the last two packets of the stream being
replaced (the first one).
Sponsored-by: On-Waves ehf
diff --git a/openbsc/src/libmgcp/mgcp_protocol.c b/openbsc/src/libmgcp/mgcp_protocol.c
index aec2cb0..7096495 100644
--- a/openbsc/src/libmgcp/mgcp_protocol.c
+++ b/openbsc/src/libmgcp/mgcp_protocol.c
@@ -40,6 +40,12 @@
for (line = strtok_r(NULL, "\r\n", &save); line;\
line = strtok_r(NULL, "\r\n", &save))
+/* 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_DEFAULT_RATE 8000
+
static void mgcp_rtp_end_reset(struct mgcp_rtp_end *end);
struct mgcp_parse_data {
@@ -965,6 +971,12 @@
end->local_alloc = -1;
talloc_free(end->fmtp_extra);
end->fmtp_extra = NULL;
+
+ /* 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;
}
static void mgcp_rtp_end_init(struct mgcp_rtp_end *end)