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Sip2sim Source IPs: Difference between revisions
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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.▼
==IP Addresses==
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 [[
==Asterisk Specific==
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.
▲At the moment AAISP have 2 'voiceless' SIP servers.
The "voiceless" call servers use two addresses per server for each of IPv4 and [[IPv6]]. These shall be:
81.187.30.111
81.187.30.113
2001:8b0:0:30
2001:8b0:0:30
(for the A server)
81.187.30.112
81.187.30.114
2001:8b0:0:30
2001:8b0:0:30
(for the B server)
▲[[Category:SIP2SIM]]
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