[asterisk-users] Firewall & audio : need a wide range to work !

[jonas kellens]

Hello list !

I have the following problem at a customer :

Their is a firewall in between the internal network (with IP-phones) and
the public Asterisk-server.

I see the following message when "sip debug" enabled :

[Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] --- (11
headers 11 lines) ---
[Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found RTP
audio format 8
[Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found RTP
audio format 101
[Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Peer audio
RTP is at port 192.168.0.24:11772
[Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found audio
description format PCMA for ID 8
[Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found audio
description format telephone-event for ID 101 alaw)
d - 0x1 (telephone-event)
[Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Peer audio
RTP is at port 192.168.0.24:11772
[Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] list_route:
hop: <sip:ict00 at 192.168.0.24:5062>
[Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36]
set_destination: Parsing <sip:ict00 at 192.168.0.24:5062> for address/port
to send to
[Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36]
set_destination: set destination to 192.168.0.24, port 5062


But when opening a range of ports on the firewall 11700 --> 11800, the
audio is not coming through !!

When opening the ports 11000 --> 11800, then the audio is coming through
fine !


Can someone explain me why range 1 is not enough fot the RTP-traffic ?!


Jonas.
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