[Asterisk-bsd] Re: One Way Audio

[Jeff Rizzo]

Tom Ivar Helbekkmo wrote:

>Jay Adelson <jay at adelson.org> writes:
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>>If the remote side calls my phone, particularly with reinvite on, I
>>can hear the remote side fine and the call works normally.
>>    
>>
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>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.
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(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.
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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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