I've an issue where outbound calls work from skype clients to external phones but inbound dont ; the call connects but no audio is available. The setup is fairly simple. We have a gateway server on the LAN that connects out to the sip provider cloud. The setup is
client/server>switch>NAT router Cisco 2911>Internet>SIP Provider
Information from the Provider is as follows. We only need these ports open between the sip server and phone gateway.
SIP Server: sip.xxxxxxx.com
SIP Port: 5060
RTP Port Range: 30000 to 40000
1st Priority: G711A (Alaw)
Current NAT rules are as follows where for example 10.10.10.1 is the Gateway on the LAN
ip nat inside source list ANY interface VlanXXX overload
ip nat inside source static udp 10.10.10.1 5060 interface Vlan666 5060
The LAN voice gateway is configured to send traffic on udp ports 35000+
The NAT translations on the outbound calls that work are below.
For the example the
LAN Gateway = 10.10.10.1
sip public IP = 184.108.40.206
NAT router Public IP = 220.127.116.11
CISCO 2911# sh ip nat translations
udp 18.104.22.168:5060 10.10.10.1:5060 22.214.171.124:5060 126.96.36.199:5060
udp 188.8.131.52:1028 10.10.10.1:35190 184.108.40.206:30496 220.127.116.11:30496
udp 18.104.22.168:1026 10.10.10.1:35191 22.214.171.124:30497 126.96.36.199:30497
I've tried to do a static udp NAT with port range/route map and also using an 'ip nat port-map command.
Does this type of setup require a static NAT on the RTP ports as well to work ? The NAT device is a Cisco2911
Any advice/documentation on this would be appreciated.
Thanks in advance.
Only config on the router is the NAT and an ACL on the ISP facing interface which is not blocking anything. I'm not sure about the ISP. I will try and get packet capture from client.