jonas kellens
2010-Mar-24 10:33 UTC
[asterisk-users] Firewall & audio : need a wide range to work !
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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100324/9c9f6775/attachment.htm
Alex Balashov
2010-Mar-24 10:39 UTC
[asterisk-users] Firewall & audio : need a wide range to work !
Have a look at rtp.conf. On 03/24/2010 06:33 AM, jonas kellens wrote:> 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 <sip:itczak00 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. >-- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/