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Asterisk SIP NAT solutions
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Asterisk, SIP and NAT
Asterisk can both act as a SIP client and a SIP server. Asterisk as a SIP client is configured with type=peer (or type=friend) in one or more client sections of sip.conf and, optionally, one or more register=> lines in the [general] section of sip.conf. Asterisk as a SIP server connects clients (SIP Phones) configured by specifying their own username, secret, etc. (and either type=peer or type=friend) in client sections of sip.conf.
Asterisk SIP channels in a NATed network can be generalized like this:
- Asterisk as a SIP client behind nat, connecting to outside SIP Proxies
- Asterisk as a SIP client behind nat, connecting to inside SIP proxies
- Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk
- Asterisk as a SIP server behind nat, clients on the outside behind a second NAT connecting to Asterisk
- Asterisk as a SIP server behind nat, clients on the inside connecting to Asterisk
- Asterisk as a SIP client outside nat, connecting to outside SIP proxies
- Asterisk as a SIP client outside nat, connecting to inside SIP proxies
- Asterisk as a SIP server outside nat, clients on the outside connecting to Asterisk
- Asterisk as a SIP server outside nat, clients on the inside connecting to Asterisk
Every setup works somewhere, but it depends on the client, the NAT, the server and many other factors. In most cases, 1 and 3 is broken. SIP is a peer-to-peer protocol and a NAT can be generalized and simplified as a solution that allows clients on the inside to connect to servers on the outside and _not_ allow clients on the outside to connect to any server on the inside.
- #1 works with a NAT-supporting proxy as SIP Express router as the outside proxy. (Get an account at IPtel.org and try!). Fails with Free World Dialup.
- #2 Works- no NAT in between
- #3 Works with port forwarding and some header mangling magic tips
- #4 Works with port forwarding, STUN on the remote and some fine tuning of RTP port allocation
- #5 Works - no NAT in between
- #6 is no problem. No NAT in the middle
- #7 is a problem if no port forwarding is done, similar to 3 above.
- #8 is no problem. No NAT in the middle
- #9 is solved with nat=yes and qualify=xxx in sip.conf for the client in most cases. Some clients (X-lite) assist themselves by using STUN and sending UDP keep-alive packets. Qualify sends keep-alive packets from Asterisk to the client on the inside.
Then we have even worse cases...
9. Asterisk inside a NAT, client inside ANOTHER NAT <---Isn't this the same as previous option 4???
In this case, we need a middle man to even find each other, an outbound SIP proxy that handles the SIP transaction and is reachable by all parties. To get media streams from point to point we need another middle man, a media server. Asterisk could be that media server, that could add media codec conversion. RTPProxy or AG Projects MediaProxy works together with SIP Express router as a media server in this situation.
I would like to add that this option contains the largest amount of cow poop around it. I've been searching for some time to fix this problem, and I'm sure it's very fixable. Also, this does seem to be a very common thing to do, so, does anyone have a GOOD solution?
I'm sure we can find case #10-xx as well.
For sample config files to setup Asterisk behind a NAT router to make and receive Free World Dialup calls see: Asterisk FWD NAT Config Example
Olles comments 2003-11-04:
I'm afraid if I configure externalIP= in sip.conf, 1 works, like with FWD, but 2 is broken.
I don't know what happens with 4 if I at the same time use externalip= and have clients
configured as 3.
As I see it, externalip= is an ugly hack that causes problems. There are better solutions
in the bug tracking system, being discussed and refined.
STUN support, and the netmask/ip-network configuration helps asterisk to find out itself
if there's a NAT in the middle and if something should be done.
- SER module nathelper: SIP Express Router module NAThelper
IP Tunneling SolutionIf you have an asterisk server, you are obvoiusly running linux (or something that can handle IPIP and GRE tunnels). In most cases when people have just 1 phone, I use an IAXy, and it is no setup required behind nat. In cases where multiple phones are required, IE setting up an client office w/ a virtual PBX, the only real hardphones out are SIP.
Farfon.... been waiting a while...
What I have done to fix this SIP NAT issue is install a small linux router at the client site, which has a public IP, and is the gateway for a private network. Using netfilter and NAT, you can do anything you want, and all you need to do is setup an IPIP tunnel back to your asterisk server to allow SIP phones to have a direct IP connection.
- iptunnel add iptun0 mode ipip remote $CLIENTROUTERIP (tunnel)
- ifconfig iptun0 18.104.22.168 pointopoint 22.214.171.124 (tunnel endpoint)
- route add -net 126.96.36.199/24 dev iptun0 (tunnel network)
- route add -net 10.0.0.0/24 dev iptun0 (client network)
- route add -net 10.0.0.0/24 gw 188.8.131.52 (client gateway)
- iptunnel add iptun0 mode ipip remote $ASTERISKIP (tunnel)
- ifconfig iptun0 184.108.40.206 pointopoint 220.127.116.11 (tunnel endpoint)
From anywhere inside the client 10.0.0.0/24 network, you can access 18.104.22.168 (asterisk) and vice versa. This totally eliminates your SIP NAT issues.
Yes this is insecure compared to IPSEC or the like, however, if you are wanting to run SIP over the net without IPSEC, this is the exact same thing, and a lot easier to setup then IPSEC.
You may want to use this script for multiple ipip tunnels
\#iptunnel add iptun0 mode ipip remote 22.214.171.124
\#ifconfig iptun0 126.96.36.199 pointopoint 188.8.131.52
\#EACH TUNNEL ON NEWLINE
\#FORMAT IS CLIENTIP:CLIENT NETWORK
for tun in $TUNNELS
d1=`echo $tun | cut -d ":" -f 1`
d2=`echo $tun | cut -d ":" -f 2`
iptunnel add iptun$c mode ipip remote $d1
ifconfig iptun$c 200.$c.0.1 pointopoint 200.$c.1.1
route add -net 200.$c.1.0/24 dev iptun$c
route add -net $d2 dev iptun$c
route add -net $d2 gw 200.$c.1.1
c += 1
for tun in $TUNNELS
iptunnel del iptun$c
c += 1
case "$1" in
- Asterisk sip qualify: The Qualify option in sip.conf pings a client behind a nat
- Nat and VOIP
- VOIP + NAT Problem description + possible solutions
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