Sip2sim Source IPs
With SIP2SIM, we register the SIM to the customer's SIP PBX just like a normal VoIP phone would do.
However, unlike a normal VoIP phone we need to register and route calls to the customer PBX in a resilient way - that means we have multiple devices at our side that are used for SIP2SIM registrations and call routing. This follows the SIP specification, but some PBXs have problems.
"Voiceless" is the name of our SIP platform. At the moment AAISP have 2 'voiceless' SIP servers.
We have reserved 10 IPv4 addresses for all of our VoIP control traffic. For our current platform "Voiceless" these are also used for the media (RTP) traffic. This makes firewalling simpler, etc. See the VoIP Firewall page for the details.
We have a help page for Asterisk and SIP2SIM: Asterisk sip2sim
Customers using asterisk will know that the config for these can be somewhat complex, listing 10 hostnames each for IPv4 and IPv6 as the way asterisk works is to look up an IP for a hostname, pick the first, and check that against the request IP address. Asterisk really needs fixing.
The "voiceless" call servers use two addresses per server for each of IPv4 and IPv6. These shall be:
18.104.22.168 22.214.171.124 2001:8b0:0:30:5060::1 2001:8b0:0:30:5060::3 (for the A server) 126.96.36.199 188.8.131.52 2001:8b0:0:30:5060::2 2001:8b0:0:30:5060::4 (for the B server)
You do not need to know if A or B. The additional two addresses will be used as "source" addresses for any request from these servers to you that need authentication. This allows asterisk to be configured separately for authenticated and unauthenticated requests.
Requests may also come from other addresses within the published block when test servers are used, etc.
We may adjust which IPs are which server at a future date as we also have to consider how we expand beyond the two servers in use currently. These IPs will be accessible via DNS as:
a.voiceless.aa.net.uk a.auth.voiceless.aa.net.uk b.voiceless.aa.net.uk b.auth.voiceless.aa.net.uk ...and so on.
If your existing asterisk config is working as per the recommendations, no changes will be needed. The wiki will be updated to explain how you can use these changes.