Today's project phase implementation was to configure VoIP over GRE Tunnel and enable voice through the tunnel from one end to the other (HQ to Remote Site)
After successfully configuring Voice in the internal network last Friday, 12th Sep, one was able to identify how this can also be implemented on remote site so I tried configuring the same layout from HQ into remote site. Somehow, the phone at first could not register and it only resets itself before configuring the IP details.
Before troubleshooting, i made sure that all cables were connected properly and correctly from the routers and switches at both ends to the end point interface that connects to the phone. Once all that was confirmed i used "spot-the-difference" troubleshooting method to compare the successfully working network (HQ site) configurations to the remote site configurations which wasn't able to register the IP phone when connected. I checked the DHCP configurations and made sure it was dissipating the right pool with the correct default-gateway address, checked ip-helper address and also each switchports to see if there was any difference between the 2. The default gateway on the Layer 3 switch was corrected since it was pointing back to itself but made no difference in the phone registration.
All seems properly configured. The only other reason why phones cannot receive ip addresses from the pool was because of the "option 150" command that enables each phone to find it's configuration from the assigned TFTP server in this case using the router.
Checking the dhcp pool configuration of Remote site router, it was obvious that this command was missing. I entered the command into the config file and tried connecting the phone to verify registration which then turns out to be successful.
Next in line was to try and call from HQ site to Remote site. Doing a bit of reading, i found out a basic configuration that can enable one to call from one phone to another (considering the use of ip phones) that are in different networks. Given the configuration below, i tried running them in my configuration file to see if it goes through.
!VoIP configuration from HQ to Remote - applied vice-versa
dial-peer voice 7 voip
destination-pattern 200.
session target ipv4:172.16.3.2
dial-peer voice 7 voip: (enables one to connect to another network and make calls - 7 represents the voice tag will be used by both sites for traffic identification
destination-pattern 200.: gives you the dial plan of the other remote site, or HQ site which will be used as line ID. There is a dot (.) at the end of the last digit which basically means that all digits that comes after is acceptable and calls will be passed through as long as the first 3 digits stays the same.
session target ipv4:"ip address": the destination next hop of which is used to reach the destination network. At first i tried using the global address assigned to the serial interfaces of each end sites given 209.165.100.2 for HQ and 209.165.200.2 for remote site. I tried making calls and the calls were successfully passed through the network. Just to make sure it is going through the tunnel, i changed the target address to the end tunnel destination address of each end which was using 172.16.3.0/30 network. This also enables calls to go through the network successfully which defines traffic path going through the GRE tunnel.
Next Phase: Features implementation and network performance measurements.
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