[Asterisk-bsd] Re: One Way Audio

[Jay Adelson]

I've checked the codecs, and everything seems to be using an acceptable
one on all sides.  I've been able to watch the call from the polycom,
it shows the properly negotiated codecs and that the call is active.

The phone itself shows packets received, the LINE on the phone shows zero
packets received.

On a completely different note, my NetBSD compile repeats this error
with some frequency:

Oct 16 10:10:12 WARNING[1262485504]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call 771bcfc62949fb87708d6ab43df176dd at 1.2.3.4 for seqno 102 (Non-critical Request)

Anything I shoudl worry about?

-j

On Sat, Oct 16, 2004 at 08:05:50AM -0700, Jeff Rizzo wrote:
> Tom Ivar Helbekkmo wrote:
> 
> >Jay Adelson <jay at adelson.org> writes:
> >
> > 
> >
> >>If the remote side calls my phone, particularly with reinvite on, I
> >>can hear the remote side fine and the call works normally.
> >>   
> >>
> >
> >Well, since the call runs through an Asterisk that has one foot on the
> >outside, with real addresses, and one on the inside, with the phones
> >on, and NAT in between, it's important that canreinvite=no.  Otherwise, 
> >the phone end points will try to speak RTP directly, and while your
> >phone is then able to address packets to the real IP address on the
> >other end, that end can't reach your phone through NAT.
> >
> > 
> >
> 
> (I've been working with Jay on this problem for a few days now - a bit 
> of the blind leading the blind :-)
> 
> One of the interesting things about this setup is that the _only_ 
> situation where audio is actually two-way is where canreinvite=yes is 
> set on both his extension (1000) and the connection to sipphone.com, and 
> he calls _out_.  When I do a tcpdump on his inside interface, the thing 
> that I notice is that in the half-duplex audio situation, packets from 
> the phone to asterisk are destined for one of the interfaces on the 
> asterisk box, and packets from asterisk to the phone are destined for 
> the other interface, which makes me wonder if the phone itself (a 
> polycom 500) is ignoring them.
> Audio _does_ work both ways when he uses a softphone (SJphone), even 
> though the same situation of packets from the phone going to one 
> interface, and packets to the phone come from the other. 
> 
> >You may also have a codec selection problem.  You should probably
> >clean out the passwords from a copy of your sip.conf, and post it
> >here.  At the same time, check which codecs your phones support, and
> >whether they can be set up with preferences for codec choice.
> >
> > 
> >
> We did play with codecs a little, but as I said we're still kind of 
> stumbling around in the dark here.   :)
> 
> +j
> 
> >-tih
> > 
> >
> 
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