[asterisk-dev] chan_jingle

[Theo Belder]

Hi,

I have a question about chan_jingle in combination with sip.
In the same network I have a googletalk client and a sip-client. These are
both behind NAT.
My asterisk server with jabber/jingle and sip is somewhere on the internet.
When I connect asterisk to the XMPP server, I can see the user is coming up
in my googletalk client.
When I call this user from googletalk, my sip client is ringing and i can
answer the call. But i only have a audiostream from googletalk to the sip
client and no stream from the sip client to googletalk.
In sip.conf I have specified 'nat=yes'.
When iIplace the asterisk server in my local lan, then everything work fine.
So i think this is a NAT issue in combination with Asterisk.

Is there something what I might doing wrong?

--
Theo
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