What happens when WAN link goes down?

[voip7372]

Assuming you have SRST on a router at a remote gateway - if the WAN link to the CallManager server were lost, how long does it generally take before the router goes into SRST mode and takes over running the phones? Do the phones lose service for a few minutes until the router determines it can't reach the CallManager and then takes over? Is there an initial loss of service when the connection is lost...and then another brief loss of service while the router takes over and the phones register with the router??? Just trying to understand the basics of how that process works. Thanks!

[Brandon Buffin]

Take a look at the following link regarding the time it takes to fail over to SRST. The initial loss of service will depend on the endpoints in use. The Call Preservation link below lists devices that do and do not support call preservation. If the device supports call preservation, the call will be preserved and the device will fail over to SRST at the end of the call. If call preservation is not supported, the call will be dropped. When the Callmanager is available again, the phone will wait until a call is completed to reregister.

SRST Description

http://www.cisco.com/en/US/products/sw/iosswrel/ps5012/products_configuration_guide_chapter09186a0080181320.html#wp1388747

Call Preservation

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803edaba.html#wp1020706

Hope this helps. If so, please rate the post.

Brandon

[voip7372]

Thanks. that helps a lot. Another question came up when reading the call preservation link though. It says: The following devices and applications support call preservation. If both parties are connected through one of the following devices, Cisco CallManager maintains call preservation:

Cisco IP phones is one of the things listed. But what I thought about when reading that is, what happens if you're at the SRST site and using an IP phone, but you were connected to an outside part on a phone company line (PSTN)? In that case, would the call be lost? Will only true IP phone to IP phone calls be preserved? Like if I am at that site and I'm on the phone with someone in another site I still have connectivity to and that person also has an IP phone.

Am I thinking right?

[Brandon Buffin]

If the call was made from an IP phone through an MGCP gateway (a survivable endpoint), the call would be preserved. If the call was made from an IP phone through an H.323 gateway (a nonsurvivable endpoint), the call would be dropped. The phone on the other end of the conversation would have no bearing.

Hope this helps. If so, please rate the post.

Brandon

[voip7372]

Yep, that explains it. Thanks

[Brandon Buffin]

You're welcome. If you found the information helpful, would you mind to rate the post(s) that helped by clicking the "Rate this post" dropdown?

Thanks,

Brandon

[arnieg]

That is not altogether correct. Once the gateway goes into SRST mode all ongoing conversations on an H.323 gateway survive for a set period of time before dropping the call to reboot the phone. I believe the default is two minutes, but I think this timer can be adjusted.

[nathan.a.reeves]

One Q with regards to a 2811 using MGCP. We're seeing call preservation fail when there is a loss of connectivity to the Call Managers. Using PVDM's on the router.

Is the list of survivable devices only what is in the docs. Eg Are 2811's classified as a Cisco IOS MGCP Gateway when running MGCP?

[Brandon Buffin]

Yes, a 2811 can be an MGCP gateway if it has MGCP controlled ports. What type of port is in use when you see the failure (FXO, PRI)? Any transcoding resources in use?

Brandon

[mparekh]

One point to be aware of regarding call preservation and MGCP - there is no call preservation when with PRI, only with T1-CAS and Analog lines.

Also H323 can be made to preserve a call during SRST by disabling the h225 timeouts. Although this is not supported by Cisco, a new solution that is supported by cisco was recently introduced. With this I believe you'll need a minimum 12.4.4 and CM 4.1(3) CR2. Best bet is to search through CCO to get the exact configuration needed.

[voip7372]

Thanks everyone for the info. Very helpful. MPARKEKH - Is that information about PRI calls not being preserved documented somewhere?...like in one of the manuals I can find online?

Thanks