[Linphone-developers] Possible issue with the RTP recv timestamps treatment in the 0.12.2 code tree...

[Dave Harley]

Hi there, 
Sorry to post out of the blue but I'm having an issue with received
audio data in the rtp stream.  Any input greatly welcome...

The symptom is that I lose audio in the speaker; it goes quiet.  The
"Mic" side is fine and the called party (remote) telephone receives its
audio perfectly.  The called party also seems to transmit fine and the
packets are seen on the wire and with good headers.  The setup is
linphone 0.12.2 on a 1 Mhz Via machine over a cross-over cable (I use a
hub when taking line traces) to a standalone sip phone.  100 base T
ethernet full duplex (at least on the crossover).

Investigation so far:  Traced the issue up the stack from the ALSA API
through the mlaw decoder and down to the RTP receiver.  A merry dance.
It appears that my rtp recv queue is going empty and that when in this
state it remains empty until a RESYNC occurs.  At that point audio
returns for a few seconds, then the queue returns to the same state
<empty> as before and audio is gone again.  Packets are available from
the socket all through this sequence.
If I do not call:
  rtp_session_unset_flag (session, RTP_SESSION_RECV_SYNC);
  in rtp_session(...) 
then all works fine.  I know that this is not a fix but its interesting
that ignoring the time_stamps seems to get around the issue.  Clear
bidirectional audio.

So underlying all this seems to be the user_ts (time-stamp) getting out
of sync with the rpt's time-stamp.  In this state the rtp_parse function
throws the packet away as being "too old" and so never they never get to
the rtp_recv queue.  I had a look at the 1.0.1 code and there seems to
have been a lot of work in this area.  Is this a known issue in the
0.12.2 build?

Anyway thanks in advance for any input.  Hope my description is in some
way clear.

Best regards,
Dave