Background:
WAN1 - Fixed IP low latency, low jitter
WAN2 - Fixed IP medium latency, higher jitter than I like for good VoIP
Firewall/Router not SIP aware
NATed LAN
Asterisk on server located on LAN.
Most, but not all ATA/IP phones on LAN
In the past I was running a v1.2 Asterisk which acted as a B2BUA (all
RTP streams relayed through Asterisk server) thus presenting only one
SIP device to the FW. Used rules in the FW to allow SIP and RTP to/
from Asterisk and WAN1. Used manual FW and Asterisk (external IP)
reconfigure for failover to less desirable WAN2 if WAN1 failed.
Upgraded to Asterisk 1.4 and now internal ATA/IP phones are now
attempting to send RTP streams directly to the Internet. Amazingly,
this seems to work for my primary ITSP (I wonder what magic they are
using to map RTP datagrams from a different IP/port than the SIP setup
negotiated?). But it does not work for ENUM destinations.
I have tried various sip.conf changes (nat=yes/no, canreinvite=yes/no/
nonat and directrtpsetup=yes/no) values trying to get all RTP traffic
to go through the Asterisk box instead of direct but have been unable
to do so. Any suggestions?
I know, the best way would be to get a SIP aware FW but replacing the
current one is not in the budget nor is there an old computer sitting
around that is suitable to press into service as a FW.
--Tod
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