Does somebody know if there is a way to send re-invites or update methods when Asterisk detects that its own public IP Address has been changed?
Here is the case of use:
When i send and receive traffic from ISP 1 and suddenly, there is a network problem (Could be packet loss, latency, specific routing errors) i change the routing priority to ISP 2, but, the already established calls and states dialog through ISP 1 instead of ISP 2, so, i was wondering if there is a way to update an already “active” channel using a SIP method.
Having my own IP addresses would surely fix the problem since the fix would be applied out of asterisk ambient, but for now i don’t have access to propietary ip addresses.
Any ideas will be greatly appreciated
Warm Regards
For signalling, they only way that might be achievable with SIP would be using the Transfer application, but ITSPs might not accept that at all, might only accept it from the original address, or might require you to have re-registered from the new address. You cannot change the Contact address mid-dialogue: RFC 3261: SIP: Session Initiation Protocol
It is possible that some ITSPs may use symmetric RTP in a way that automatically recovers from the change in RTP address, but it is also possible that, for security reasons, they will only allow symmetric RTP updates at the start of a sesseion.
If using Transfer, you will probably have to have re-registered, under the new address, before issuing the transfer, as ITSPs may well either ignore the domain name or require it to the AOR one at their end.