guilin.cao at teclub.cn
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 Posted: Sat Mar 28, 2015 7:07 pm Post subject: [asterisk-users] Help! How to make Asterisk support ICE in p |
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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>
| Quote: | 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
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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
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Dennis Cao (曹贵林 ) |
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