[asterisk-users] Early media support for Asterisk behind NAT

[Mayur]

Hi,

   I have asterisk 1.4.16 behind a NAT-FW which is using a hosted SIP trunk
for PSTN calling. Asterisk is configured to support nat with nat=yes in
sip.conf. Now the hosted PSTN Gateway supports symmetric RTP and early media
using 183 Session Progress. So If I call a PSTN number which has IVR message
played before the call is connected (via 183), those media RTP packets do
not reach the asterisk inside till asterisk sends out media packet to the
PSTN gateway. I have used rtpkeepalive option and set it to 1 sec. But it
seems that I drop rtp voice packets in the initial instructions played by
the IVR.

 

How do I make sure that asterisk sends RTP packets (null rtp) to the PSTN
gateway just after receiving the media details in 183 SDP to open the
firewall port from inside?

 

Regards,

Mayur   

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