SIP Trunk SRST

[ty.masse]

At our company we are doing SIP Trunking between the provider and CUCM.  The trunk terminates on the call manager itself.  We don't have any CUBE in our environment.  Most of our remote locations have a regular router with no pvdm, fxo, or uck9 and calls are working fine.  The router is only passing data packets and is not voice aware.

My question is this.  If I had a location with a router with the required voice items.  PVDM, FXO with analog lines and UCK9.  Is SRST possible in such a scenario?

Thanks.

[Chris Deren]

You need local trunking for it to make sense. Since SIP trunk is centralized at head end location then unless you are going to get another trunk or POTS lines connected to the FXO line it would not sense.  If you get alternate trunk i.e. POTS that will allow you to make outbound calls, inbound are still going to arrive via the SIP trunk unless you perform some forwarding.

[Francisco Granados]

+5 Chris,

In summary, SRST will help you stay alive for outbound calls like 911, through local dial peers routing out of local POTS lines connected to a VIC-FXO card.

If you have a main number at the remote office, you could choose to use "forward on unregister" and route inbound calls to a POTS line connected to the remote SRST gateway.

Thanks,

FG

[ty.masse]

My confusion is that I know how Fallback works with MGCP, H323 and CUBE they're all essentially local gateways and all the phones will register locally upon detection that they're unavailable.  In this case it's not running any of these how does it know to failover to SRST when it's not running any local voice protocol?

[Chris Deren · Accepted solution]

You need to separate phones from trunk when it comes to understanding SRST. Phones know when to go into SRST based on lost keep-alives to their CUCM servers, and if configured for SRST via DP will reach out to the SRST designation and attempt to register.

Trunks such as H323 and SIP do not register with CUCM, they are peer to peer connections and routing is based on dial-peers. If the destination set on dial-peer to CUCM is no longer available as there is no reply to SIP invite (if SIP) and there are locally registered devices to the SRST router the calls will be extended to them (assuming proper translations, etc when needed).  

If your SIP trunks are centralized and not terminated on this voice gateway then the phones will not have any external access when they failover into SRST and will only be able to reach each other. If there is a POTS line for example with POTS dial-peer defined for external access the phones will be able to use it for external calling.

[ty.masse]

Thanks to everyone that replied.  I appreciate it.