[Asterisk-Dev] SIP failover using Asterisk and openais

[Kevin P. Fleming]

John Todd wrote:

> I think perhaps someone from Digium could comment on this, or someone 
> from IBM, but I have no idea who is responsible for the work/product 
> from either company.

The work that has been done to date has been by staff at IBM, in 
cooperation with Solid, who makes a database platform used as part of 
the solution. There is another Asterisk partner, Verso Technologies, 
that also demonstrated high-availability Asterisk using Solid's 
database, but using a different solution that IBM demonstrated.

Digium has not been involved in the code development of either of these 
solutions, although we have obviously been in contact with both 
companies a great deal. As I understand it, IBM's solution is already 
built on SAF technologies, although they chose the Fujitsu 
implementation rather than OpenAIS (for reasons that I am not aware of).

In any case, properly implementing HA Asterisk is a huge project; 
keeping SIP (and RTP) state is a small, but important, part of it. To 
answer Steven's question in an earlier message: yes, there is a great 
deal of state that must be preserved beyond just the SIP/RTP state. For 
example, the call may not even be SIP<->SIP, it might be using another 
technology on one side. There may also be transcoding of the media 
stream in place, in which case the codecs will have state. The Asterisk 
PBX engine will also have 'dialplan' state, channel variables and 
various other bits it uses to keep track of the state of the call beyond 
just the SIP sessions.

For a true HA solution, all of this state must be preserved.