Basic Cisco Unified SRST Usage best ccnp course training in gurgaon india

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Cisco Unified SRST provides CUCM with fallback support for Cisco Unified IP Phones
that are attached to a Cisco router on a local network.
Cisco Unified SRST enables routers to provide basic call-handling support for Cisco
Unified IP Phones when they lose connection to remote primary, secondary, and tertiary
CUCM servers or when the WAN connection is down.
Basic SRST supports up to 720 SCCP IP Phones on the highest supported router (3845). In
addition, the VG248 Analog Phone Gateway offers basic SRST support and secure voice
fallback. However, it does not support hunt groups or message waiting indication (MWI)
for voice mail in fallback mode.
Cisco Unified SIP SRST Usage
Cisco Unified SIP SRST provides backup to an external SIP proxy server by providing
basic registrar and redirect server services earlier than Cisco Unified SIP SRST version 3.4
or B2BUA for Cisco Unified SIP SRST version 3.4 and higher services.
A SIP phone uses these services when it is unable to communicate with its primary SIP
proxy of CUCM in the event of a WAN connection outage.
Cisco Unified SIP SRST can support SIP phones with standard RFC 3261 feature support
locally and across SIP WAN networks. With Cisco Unified SIP SRST, SIP phones can place
calls across SIP networks in the same way that SCCP phones do.
Cisco Unified SIP SRST supports the following call combinations: SIP phone to SIP phone,
SIP phone to public switched telephone network (PSTN) or router voice port, SIP phone to
SCCP phone, and SIP phone to WAN VoIP using SIP.
SIP proxy, registrar, and B2BUA servers are key components of a SIP VoIP network. These
servers usually are located in the core of a VoIP network. If SIP phones located at remote
sites at the edge of the VoIP network lose connectivity to the network core (because of
a WAN outage), they may be unable to make or receive calls. Cisco Unified SIP SRST
functionality on a SIP PSTN gateway provides service reliability for SIP-based IP Phones
in the event of a WAN outage. Cisco Unified SIP SRST enables the SIP IP Phones to
continue making and receiving calls to and from the PSTN. They also can continue making
and receiving calls to and from other SIP IP Phones by using the dial peers configured on
the router.
102 Chapter 5: Examining Remote-Site Redundancy Options
When the IP WAN is up, the SIP phone registers with the SIP proxy server and establishes
a connection to the B2BUA SIP registrar (B2BUA router). But any calls from the SIP phone
go to the SIP proxy server through the WAN and out to the PSTN.
When the IP WAN fails or the SIP proxy server goes down, the call from the SIP phone
cannot get to the SIP proxy server. Instead, it goes through the B2BUA router out to the
PSTN.
NOTE The B2BUA acts as a user agent to both ends of a SIP call. The B2BUA is
responsible for handling all SIP signaling between both ends of the call, from call
establishment to termination. Each call is tracked from beginning to end, allowing the
operators of the B2BUA to offer value-added features to the call. To SIP clients, the
B2BUA acts as a user agent server on one side and as a user agent client on the other
(back-to-back) side. The basic implementation of a B2BUA is defined in RFC 3261, as
mentioned earlier in this chapter.