[OpenSIPS-Users] openSIPS and RTP media stream fault-tolerance

[Lars Ramchter]

Hello,

I would like to use OpenSIPs for a voip demo.
I've seen it's possible to do NAT traversal with nathelper and RTPproxy.
It seems possible to use multiple RTPproxy instances. The RTPproxy homepage
says "The nathelper module included into the SIP Express Router (SER), OpenSIPS
or Kamailio as well Sippy B2BUA allow using multiple RTPproxy instances
running on remote machines for fault-tolerance and load-balancing purposes."

I've been wondering what fault-tolerance means ?
Does it mean that if a RTPproxy instance stops, no next call will go through
it and all further calls will use an other one ?
Or does it mean that in case of a failure of the RTPproxy during a call (I'm
just concern about the RTP media stream), it would be redirected on
the fly through
another RTPproxy instance with minimum loss ?

Is that solution even possible ?
Or would the call be ended, and then I would have to make another call
through another RTPproxy instance ?

I don't know if I should post this message on the RTPproxy mailing list too.

Best regards,
Lars Ramchter
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