Wednesday, August 24, 2011

Major 15.1(3)T Voice Features

Major 15.1(3)T Voice Features

RSVP Ingress CAC (Connection Admission Control)
         With load balancing, inbound traffic for the Resource Reservation Protocol (RSVP) agent is not guaranteed to use same WAN link as outbound traffic. This traffic can overrun the priority queue because reservations are currently made only on outgoing CE (Customer Edge) interfaces, in the direction of the Real-Time Protocol (RTP) stream. The Ingress CAC mechanism improves the solution by performing CAC on the ingress interface on the next hop.
Benefits
         RSVP can now work with asymmetric media paths (asymmetric routing cases).
         RSVP can now meet the needs of headquarters-to-branch-office deployment scenarios in which bandwidth profiles are not identical.
         RSVP is now aligned with service provider deployments.
Mediatrace 1.0
         Mediatrace is a powerful tracing and dynamic configuration tool. It provides automatic path discovery of traffic streams and can discover both Layer 2 and 3 devices along the path. Mediatrace while following the path of particular flow can gather crucial hop by hop information like packet loss and network jitter that the flow might be experiencing. It can also be used to collect system information such as CPU and memory utilization data of the intermediate hops. Mediatrace can be run on demand or scheduled for periodic monitoring and data collection.
Benefits
          Mediatrace provides powerful troubleshooting and fault isolation from a single point in the network.
          Dynamic configuration provides an efficient monitoring mechanism to help reduce system resource consumption for network devices.
          Support for both Layer 2 and 3 devices provides visibility on the complete path.

Cisco Unified Communications Manager Express (UCME) and Selected Survivable Remote Site Telephony (SRST) 8.5 Enhancements

         The main Cisco UCME and SRST features delivered in Release 8.5 release of the products include support for the following:

         Overlap sending on ISDN and Primary Rate Interface (PRI)
         Forced authorization code (FAC)
         Signal-to-noise ratio (SNR) enhancements on Cisco Unified Communications Manager Express
         SSL VPN client on SCCP UCME Customization of services and directories page
         Immediate divert (iDivert) feature on Session Initiation Protocol (SIP) phones
         Busy lamp field on phone applications

   Benefits

         Release 8.5 of Cisco Unified Communications Manager Express and Cisco Unified SRST reduces the cost of Cisco Unified Communications Manager Express teleworkers through improved deployment flexibility of SSL VPN clients on SCCP Cisco Unified IP Phones.
         This release also improves the Cisco IP Phone user interface and the billing and accounting options, and it makes Cisco Unified Communications Manager Express and Cisco Unified SRST easier to deploy and maintain.

Cisco Advanced Foreign Exchange Station (FXS) Analog Gateway Features
         This release adds advanced FXS Analog Gateway features and SCCP over Transport Layer Security (TLS) with Cisco Unified Communications Manager Express.
         The security endpoint uses the existing PKI, which is used to connect to Cisco Unified Communications Manager Express. The PKI obtains the certificate from a third party certificate authority (CA) and uses it to establish TLS sessions with Cisco Unified Communications Manager. Secure RTP (SRTP) is used to encrypt the media.
         Advanced supplementary features on FXS analog ports with Cisco Unified Communications Manager include:
o    Repetitive call-waiting tone
o    Callback on no answer
o    Media renegotiation
o    cBarge
o    Configurable exclusive audio message waiting indicator (AMWI) and visual message waiting indicator (VMWI)
o    Single-number reach (SNR) (supported for both Cisco Unified Communications Manager and Cisco Unified Communications Manager Express)
         Enhanced serviceability for analog FXS voice port line measurement on Cisco VG224 Analog Voice Gateways and Cisco IAD2430 Integrated Access Devices and certain voice interface cards will be added. New interfaces will be created to measure line voltage, current, impedance, and capacitance of the analog FXS voice port.
  Benefits
         The capability to provide granular troubleshooting capabilities during initial network provisioning improves deployment when disparate analog endpoints are included.
         Automated signaling and media continuity checks make network operations and health monitoring easier in large analog deployments.
Cisco ISR SIP Gateway and Cisco Unified Border Element 8.6 Enhancements: Reporting End-of-Call Statistics in SIP BYE Message
         Cisco Unified Border Element and the Cisco ISR SIP voice gateway report end-of-call statistics through the SIP BYE message.
Benefits
         This gateway can provide demarcation functions on the public switched telephone network (PSTN) gateway as well as the Cisco Unified Border Element. These statistics can be used to update call data records on Cisco Unified Communications Manager or Cisco Unified Communications Manager Express
Cisco ISR SIP Gateway and Cisco Unified Border Element 8.6 Enhancements: Conditional SIP Header Manipulation
         Cisco Unified Border Element 8.6 adds support for manipulation of one SIP header based on the contents of another SIP header. This feature extends the SIP profile function that is already available on the Cisco Unified Border Element. It includes the capability to copy the contents of one header to another, or modify the contents of one header based on the contents of another. SIP headers on the out-leg can be modified based on the contents of any SIP header on any SIP message received on the in-leg.
Benefits
         This enhancement increases flexibility in network design.
Cisco ISR Unified Border Element 8.6 Enhancements: Media Flow-Around High-Density Transcoding with SIP Signaling
         Cisco Unified Border Element provides support for dynamic media flow-around with no additional media control for SIP calls from branch offices to PSTNs that are controlled by a central Cisco Unified Communications Manager. Cisco Unified Border Element performs the delayed-offer-to-early-offer translation. Cisco Unified Border Element controls SIP signaling for the entire call duration, with media flowing around the Cisco Unified Border Element directly from the branch office to PSTN.
Benefits
         This enhancement increases flexibility in network design.
Cisco ISR and Cisco Unified Border Element 8.6 Enhancements: RFC 3311 SIP UPDATE Message
         This release adds RFC 311 SIP UPDATE message support on the Cisco Unified Border Element, which allows modification of the parameters of a SIP session. The UPDATE message allows modification of session parameters just as the reINVITE SIP message does, except that the UPDATE message can be received either during early dialog (during the initial Invite transaction) or during established dialog. This feature enables Cisco Unified Border Element to interwork with service providers that use the UPDATE header instead of ReINVITE.
Benefits
         This enhancement increases flexibility in network design.
Cisco ISR and Unified Border Element 8.6 Enhancements: SIP-Based RSVP to Non-RSVP Cisco Unified Communications Manager Interworking
         In this release, Cisco Unified Border Element provides interworking between the Cisco Unified Communications Manager controlled RSVP leg and the non-RSVP call leg for SIP calls. This support includes early offer-to-early-offer, delayed-offer-to-delayed-offer, and delayed-offer-to-early-offer combinations. Interworking between a non-RSVP H.323 call leg and a Cisco Unified Communications Manager controlled RSVP SIP call leg for fast-start-to-early-offer and slow-start-to-delayed-offer calls is also supported.
Benefits
         This enhancement increases flexibility in network design.
Cisco ISR and Cisco Unified Border Element 8.6 Enhancements: SIP Named Signaling Events Capability Negotiations Using Session Description Protocol
         The addition of SIP named signaling events (NSE) capability negotiations using the Session Description Protocol (SDP) provides fax and modem interworking.
Benefits
         This enhancement increases flexibility in network design.

Cisco ISR and Cisco Unified Border Element 8.6 Enhancements: SIP Registration Proxy
         In this release, Cisco Unified Border Element adds a SIP registration proxy. This feature adds the capability to send outbound registrations based on incoming registrations. It enables direct registration of SIP endpoints with the SIP registrar in hosted unified communications deployments that use Cisco Unified Border Element. This feature allows header manipulation of the REGISTER message through a SIP profile. Cisco Unified Border Element provides the flexibility to use authentication credentials from either the incoming registration requests or the CLI configuration. Registration requests are authenticated, or the authentication occurs end-to-end, in which case Cisco Unified Border Element will pass the credentials.
Benefits
         This enhancement increases flexibility in network design.

Cisco ISR and Cisco Unified Border Element 8.6 Enhancements: SIP Request Processing Limit
         This release adds support for limiting SIP request processing on the Cisco Unified Border Element to Incoming SIP call processing. This feature augments the security functions offered by the Cisco Unified Border Element. This feature also enables the monitoring (through the CLI) and display of the current call rate being processed by the Cisco Unified Border Element. This SIP trunk enhancement prevents DoS attacks and provides a mechanism for troubleshooting managed services deployments.
Benefits
         This enhancement increases flexibility in network design and management.

Cisco ISR and Cisco Unified Border Element 8.6 Enhancements: Media Flow Release While Maintaining SIP Signaling Control
         With this release, Cisco Unified Border Element can dynamically release the media flow while retaining SIP signaling control. This feature is used for media trombone designs and media hairpinning. Cisco Unified Border Element will hairpin SIP calls that end up as PSTN-to-PSTN calls, which may occur as a result of an external call-transfer or call-forward operation. After the media flow is released, Cisco Unified Border Element exerts no further control over media.
Benefits
         This enhancement increases flexibility in network design and management.



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