I'm running into a problem with SER & rtpproxy trying to get a NAT scenario to work correctly. I'm still learning about SER, so bear with me... :) Here is my network layout in a nutshell: SIP ATA - 10.0.201.196 (private only, no NAT to internet) SER - eth0: 208.208.53.5/24 - eth1: 10.0.201.5/24 Asterisk - eth0: 65.234.34.5/24 *Everything* is configured to proxy through SER to Asterisk currently. My configuration file is here: http://webdev.digitalpath.net/~rayvd/voice/ser.cfg Here's the problem I'm running into: 1. SIP ATA places call (INVITE sent) 2. SER routes INVITE to Asterisk (and modifies c= in the SDP portion to point back to 208.208.53.5 as I understand it). 3. Ring, ring, other side answers. 4. Asterisk sends back a 200 OK message with c= 65.234.34.5 in its SDP message body. 5. My configuration changes this to 208.208.53.5 6. Voice packets now are outbound on SER eth1 with src ip of 208.208.53.5 and dst ip of 10.0.201.196. Obviously this will not work. regardless of the direction traffic is going. If I bind rtpproxy to 10.0.201.5 on SER, then the IP in step 6 above will be correct, but the Asterisk server will be trying to send RTP traffic to 10.0.201.5 instead of 208.208.53.5. What am I doing wrong here? Can I run two instances of rtpproxy depending on the direction of the audio? I've tried different parameters to force_rtp_proxy() to no avail. Is there a function to manually change the c= line in the SDP body before it gets sent out? Thanks for any help! -- Ray Van Dolson Linux/Unix Systems Administrator Digital Path, Inc.