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Re: [Linphone-developers] Possible issue with the RTP recv timestamps tr

From: Simon Morlat
Subject: Re: [Linphone-developers] Possible issue with the RTP recv timestamps treatment in the 0.12.2 code tree...
Date: Tue, 19 Apr 2005 10:53:16 +0200
User-agent: KMail/1.7.2


The problem of received packet being out of time is usually a problem with the 
sender, not the receiver. The sender may sent too fast, or too slowly, 
resulting in such behaviours. Usually those sender problems are related to 
soundcard problems. Using alsa driver everywhere solves it.
There have been progresses in 1.0.x to allow linphone to compensate slow clock 
slides between sender and receiver. This may help.

Le Jeudi 14 Avril 2005 20:23, Dave Harley a écrit :
> 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
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