I've recently moved to a Cisco 881 router for my WAN link. I was previously using a Cisco Linksys WAG320N as my modem/router/WiFi AP/NAT firewall. The WAG320N is now running in bridged mode, so it's simply acting as a modem with one of it's LAN ports connected to FE4 WAN on my Cisco 881.
The Cisco 881 get's a DHCP provided IP from my ISP. My LAN is part of default Vlan 1 (192.168.1.0/24). General internet connectivity is working great, I've managed to setup static NAT rules for my HTTP/HTTPS/SMTP/etc. services which are running on my LAN. I don't know whether it's worth mentioning that I've opted to use NVI NAT (ip nat enable as opposed to the traditional ip nat outside/ip nat inside) setup. My reason for this is that NVI allows NAT loopback from my LAN to the WAN IP and back in to the necessary server on the LAN.
I run an Asterisk 1.8 PBX on my LAN, which connects to a SIP provider on the internet. Both inbound and outbound calls through the old setup (WAG320N providing routing/NAT) worked fine. However, since moving to the Cisco 881, inbound calls drop after around
10 seconds, whereas outbound calls work fine.
The following message is logged on my Asterisk PBX:
[Dec 9 15:27:45] WARNING[27734]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission
[email protected] for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6528ms with no response
[Dec 9 15:27:45] WARNING[27734]: chan_sip.c:3670 retrans_pkt: Hanging up call
[email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
(I know that this is quite a common issue - I've spend the best part of 2 days solid on this, trawling Google.)
I've done as I am told and checked https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions. Referring to the section "Other SIP requests" in the page linked above, I believe that the hangup to be caused by the ACK from my SIP provider not being passed back through NAT to Asterisk on my PBX.
I tried to ascertain this by dumping the packets on my WAN interface on the 881. I managed to obtain a PCAP dump of packets in/out of my WAN interface. Here's an example of an ACK being reveived by the router from my provider:
689 21.219999 193.
x.x.
x 188.
x.x.
x SIP 502 Request: ACK sip:441925xxxxxx@188.
x.x.
x |
However a SIP trace on the Asterisk server show's that there are no ACK's received in response to the 200 OK from my PBX:
http://pastebin.com/wwHpLPPz
In the past, I have been strongly advised to disable any sort of SIP ALGs on routers and/or firewalls and the many posts regarding this issue on the internet seem to support this. However, I believe on Cisco IOS, the config command to disable SIP ALG is no ip nat service sip udp port 5060 however, this doesn't appear to help the situation. To confirm that config setting is set:
Router1#show running-config | include sip
no ip nat service sip udp port 5060
Another interesting twist: for a short period of time, I tried another provider. Luckily, my trial account with them is still available, so I reverted my Asterisk config back to the revision before I integrated with my current provider. I then dialled in to the DDI associated with the trial trunk and the call didn't get hung up and I didn't get the error above! To me, this points at the provider, however I know, like all providers do, will say "There's no issues with our SIP proxies - it's your firewall." I'm tempted to agree with this, as this issue was not apparent with the old WAG320N router when it was doing the NAT'ing.
I'm sure you'll want to see my running-config too:
!
! Last configuration change at 15:55:07 UTC Sun Dec 9 2012 by xxx
version 15.2
no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
no service password-encryption
service sequence-numbers
!
hostname Router1
!
boot-start-marker
boot-end-marker
!
!
security authentication failure rate
10 log
security passwords min-length 6
logging buffered 4096
logging console critical
enable secret 4 xxx
!
aaa new-model
!
!
aaa authentication login local_auth local
!
!
!
!
!
aaa session-id common
!
memory-size iomem
10
!
crypto pki trustpoint TP-self-signed-xxx
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-xxx
revocation-check none
rsakeypair TP-self-signed-xxx
!
!
crypto pki certificate chain TP-self-signed-xxx
certificate self-signed 01
quit
no ip source-route
no ip gratuitous-arps
ip auth-proxy max-login-attempts 5
ip admission max-login-attempts 5
!
!
!
!
!
no ip bootp server
ip domain name dmz.merlin.local
ip domain list dmz.merlin.local
ip domain list merlin.local
ip name-server
x.x.
x.x
ip inspect audit-trail
ip inspect udp idle-time 1800
ip inspect dns-timeout 7
ip inspect tcp idle-time 14400
ip inspect name autosec_inspect ftp timeout 3600
ip inspect name autosec_inspect http timeout 3600
ip inspect name autosec_inspect rcmd timeout 3600
ip inspect name autosec_inspect realaudio timeout 3600
ip inspect name autosec_inspect smtp timeout 3600
ip inspect name autosec_inspect tftp timeout 30
ip inspect name autosec_inspect udp timeout 15
ip inspect name autosec_inspect tcp timeout 3600
ip cef
login block-for 3 attempts 3 within 3
no ipv6 cef
!
!
multilink bundle-name authenticated
license udi pid CISCO881-SEC-K9 sn
!
!
username xxx privilege 15 secret 4 xxx
username xxx secret 4 xxx
!
!
!
!
!
ip ssh time-out 60
!
!
!
!
!
!
!
!
!
interface FastEthernet0
no ip address
!
interface FastEthernet1
no ip address
!
interface FastEthernet2
no ip address
!
interface FastEthernet3
switchport access vlan 2
no ip address
!
interface FastEthernet4
ip address dhcp
no ip redirects
no ip unreachables
no ip proxy-arp
ip nat enable
duplex auto
speed auto
!
interface Vlan1
ip address 192.168.1.1 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp
ip nat enable
!
interface Vlan2
ip address 192.168.0.2 255.255.255.0
!
ip forward-protocol nd
ip http server
ip http access-class 1
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
!
no ip nat service sip udp port 5060
ip nat source list 1 interface FastEthernet4 overload
ip nat source static tcp
x.x.
x.x 80 interface FastEthernet4 80
ip nat source static tcp
x.x.
x.x 443 interface FastEthernet4 443
ip nat source static tcp
x.x.
x.x 25 interface FastEthernet4 25
ip nat source static tcp
x.x.
x.x 587 interface FastEthernet4 587
ip nat source static tcp
x.x.
x.x 143 interface FastEthernet4 143
ip nat source static tcp
x.x.
x.x 993 interface FastEthernet4 993
ip nat source static tcp
x.x.
x.x 1723 interface FastEthernet4 1723
!
!
logging trap debugging
logging facility local2
access-list 1 permit 192.168.1.0 0.0.0.255
access-list 1 permit 192.168.0.0 0.0.0.255
no cdp run
!
!
!
!
control-plane
!
!
banner motd
Authorized Access only
!
line con 0
login authentication local_auth
length 0
transport output all
line aux 0
exec-timeout 15 0
login authentication local_auth
transport output all
line vty 0 1
access-class 1 in
logging synchronous
login authentication local_auth
length 0
transport preferred none
transport input telnet
transport output all
line vty 2 4
access-class 1 in
login authentication local_auth
length 0
transport input ssh
transport output all
!
!
end
...and, if it's of any use, here's my Asterisk SIP config:
[general]
context=default ; Default context for calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
tcpenable=no ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
; Specifying a port in a SIP peer definition or
; when dialing outbound calls will supress SRV
; lookups for that peer or call.
directmedia=no ; Don't allow direct RTP media between extensions (doesn't work through NAT)
externhost=<MY DYNDNS HOSTNAME> ; Our external hostname to resolve to IP and be used in NAT'ed packets
localnet=192.168.1.0/24 ; Define our local network so we know which packets need NAT'ing
qualify=yes ; Qualify peers by default
dtmfmode=rfc2833 ; Set the default DTMF mode
disallow=all ; Disallow all codecs by default
allow=ulaw ; Allow G.711 u-law
allow=alaw ; Allow G.711 a-law
; ----------------------
; SIP Trunk Registration
; ----------------------
; Orbtalk
register => <MY SIP PROVIDER USER NAME>:
[email protected]/<MY DDI> ; Main Orbtalk number
; ----------
; Trunks
; ----------
[orbtalk] ; Main Orbtalk trunk
type=peer
insecure=invite
host=sipgw3.orbtalk.co.uk
nat=yes
username=<MY SIP PROVIDER USER NAME>
defaultuser=<MY SIP PROVIDER USER NAME>
fromuser=<MY SIP PROVIDER USER NAME>
secret=xxx
context=inbound
I really don't know where to go with this. If anyone can help me find out why these calls are being dropped off, I'd be grateful if you could chime in! Please let me know if any further info is required.
