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Re: [Linphone-users] Dropped sound problem with Asterisk??

From: John A. Sullivan III
Subject: Re: [Linphone-users] Dropped sound problem with Asterisk??
Date: Thu, 16 Sep 2004 21:38:33 -0400

On Thu, 2004-09-16 at 08:18, Simon MORLAT wrote:
> >2) How do I fix this suspected synchronization/overrun problem?
> >  
> >
> I 'm nearly 100% sure that linphone is well and precisely synchronised: 
> I mean that it emits and receive RTP packets
> with timestamps correlated to the wallclock ( the hardware clock of the 
> pc as returned by gettimeofday()).
> The problem should be in asterisk.
> <snip>

> >PS - here are some console message from Linphone.  I do not know if they
> >are significant but they look unhealthy:
> >  
> >
> All those messages are normal; but I can see something interesting with 
> the RTP stats at the end:
> packet_sent=10328
> packet_recv=6564
> Linphone has received less packets than he has sent. In a full duplex voip 
> session with same codecs in both direction (like it is usually the case), 
> this is not normal. It means that packet output rate from asterisk is too 
> low. Normally, asterisk should output 8000 samples/second, so 8000 
> bytes/second, for example 50 packets/s with 20ms (160bytes) of audio in each 
> packet.
> This gives you some ideas to dig...
> Simon
> >** Message: Sending dtmf 1
> >MediaStreamer-Message: Sending DTMF.
> >** Message: Sending dtmf 1
> >MediaStreamer-Message: Sending DTMF.
> >** Message: Sending dtmf 1
> >MediaStreamer-Message: Sending DTMF.
> >| INFO1 | <nict_callbacks.c: 30> Transaction 11 killed.
> >
> >| INFO1 | <osipdialog.c: 1918> Dialog is removed. It remains 1 dialog(s)
> >in the ua list.
> >
> >** Message: Sending dtmf 2
> >MediaStreamer-Message: Sending DTMF.
> >** Message: Sending dtmf 1
> >MediaStreamer-Message: Sending DTMF.
> >** Message: Sending dtmf 1
> >MediaStreamer-Message: Sending DTMF.
> >** Message: Sending dtmf 1
> >MediaStreamer-Message: Sending DTMF.
> >** Message: Sending dtmf 2
> >MediaStreamer-Message: Sending DTMF.
> >MediaStreamer-Message: Mediastreamer processing thread is exiting.
> >oRTP-stats-Message:
> >   Global statistics :
> > packet_sent=10328
> > sent=1510937 bytes
> > packet_recv=6564
> > hw_recv=1052173 bytes
> > recv=520005 bytes
> > unavaillable=10235 bytes
> > outoftime=532168
> > bad=0
> > discarded=0
> >
> >  
> >
I traced a number of calls - somewhere around 50,000 packets.  It
appears that there is an asymmetric pattern.  I noticed that in periods
of silence, the PBX stopped transmitting but Linphone always seemed to
transmit.  Thus, the number of transmitted packets would outnumber the
received packets.

However, I continue to have this problem with dropped sound and I only
have it with Linphone.  KPhone works fine in that regard (it breaks in
other areas).  I suspect that this is a Linphone problem even if it is
not a syncrhonization problem.

I ran ethereal from the same station where I was using Linphone and
launched a real time graph of the packet stream from the PBX versus to
the PBX.  While listening to the messages coming from the voice mail
system, I could hear the voice and see the packets to and from the PBX. 
Suddenly the voice dropped but I could still see the packets going to
and coming from the PBX.  The PBX was continuing the transmission and my
IP stack was continuing to receive the packets.  Something seems to have
broken when the packet data was handed off to Linphone.

I'm entirely willing to help troubleshoot this problem but this is far
outside my area of expertise.  I will need someone else's guidance.  How
can we further isolate the source of the problem? Thanks - John
John A. Sullivan III
Open Source Development Corporation
Financially sustainable open source development

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