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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