Fix RTP jitter buffer that never stop to increase.
Duration passed to osmo_rtp_send_frame_ext function is based
on the last frame and the current one. Duration must then be
added to the timestamp before being transmitted.
Change-Id: I0593d6530c097cca34125a0ae2dd1b019b4dd80d
diff --git a/src/trau/osmo_ortp.c b/src/trau/osmo_ortp.c
index 141065e..4e9df56 100644
--- a/src/trau/osmo_ortp.c
+++ b/src/trau/osmo_ortp.c
@@ -467,9 +467,9 @@
return -ENOMEM;
rtp_set_markbit(mblk, marker);
+ rs->tx_timestamp += duration;
rc = rtp_session_sendm_with_ts(rs->sess, mblk,
rs->tx_timestamp);
- rs->tx_timestamp += duration;
if (rc < 0) {
/* no need to free() the mblk, as rtp_session_rtp_send()
* unconditionally free()s the mblk even in case of