[asterisk-users] Help! How to make Asterisk support ICE in public network

[曹贵林]

Hi friends,
I am just starting use asterisk for our VoIP server. It works fine in LAN. But when it is deployed in public network(with a public IP), the SIP clients in different NAT fails to communicate with each other. I have set 'icesupport' to 'yes' in sip.conf and set STURN and TURN server in rtp.conf. It still fails! 


Hope someone to help me out! Thanks in advance:) 


This is the output of CLI:
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  == Using SIP RTP CoS mark 5
    -- Called SIP/6003
    -- SIP/6003-00000001 is ringing
    -- SIP/6003-00000001 is ringing
    -- SIP/6003-00000001 is ringing
    -- SIP/6003-00000001 is ringing
    -- SIP/6003-00000001 is ringing
    -- SIP/6003-00000001 answered SIP/6004-00000000
    -- Channel SIP/6004-00000000 joined 'simple_bridge' basic-bridge <2a01fb30-96e2-48b7-baaa-c2f172127c07>
    -- Channel SIP/6003-00000001 joined 'simple_bridge' basic-bridge <2a01fb30-96e2-48b7-baaa-c2f172127c07>
       > Bridge 2a01fb30-96e2-48b7-baaa-c2f172127c07: switching from simple_bridge technology to native_rtp
       > Remotely bridged 'SIP/6003-00000001' and 'SIP/6004-00000000' - media will flow directly between them
       > Remotely bridged 'SIP/6003-00000001' and 'SIP/6004-00000000' - media will flow directly between them
       > 0x7f5968006760 -- Probation passed - setting RTP source address to 114.81.254.172:4145
       > 0x1fefbb0 -- Probation passed - setting RTP source address to 114.92.58.65:7076
st-srv-cs2*CLI> 
st-srv-cs2*CLI> 
st-srv-cs2*CLI> 
    -- Channel SIP/6004-00000000 left 'native_rtp' basic-bridge <2a01fb30-96e2-48b7-baaa-c2f172127c07>
  == Spawn extension (my-phone, 6003, 1) exited non-zero on 'SIP/6004-00000000'
    -- Channel SIP/6003-00000001 left 'native_rtp' basic-bridge <2a01fb30-96e2-48b7-baaa-c2f172127c07>
[Mar 18 12:04:22] NOTICE[30157]: chan_sip.c:23890 handle_response_peerpoke: Peer '6003' is now Lagged. (3285ms / 2000ms)
[Mar 18 12:04:33] NOTICE[30157]: chan_sip.c:23890 handle_response_peerpoke: Peer '6003' is now Reachable. (1244ms / 2000ms)
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5




------------------
Dennis Cao (曹贵林 )
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