Ine Vo Dd Wb Vol1 Mod12 Cube Tasks

Voice Deep Dive Workbook Version 3.5 Module 12

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Cisco Unified Border Element: CUBE :: Tasks

Lab Instructions:

Prior to starting this lab, ensure that you have both read through the Voice Rack Rental Guide before beginning this lab, so that you are fully aware of how all equipment works and is accessed, and that you have loaded both the initial router/switch configs from INE’s members’ site, and that you have manually loaded the CUCM initial configs as described below.

Note

- Most Deep Dive lab scenarios require existing configuration before they can begin to be configured. Also, most lab scenarios use different pre-configuration from other Deep Dive labs. Therefore it is necessary to load the "Startup Configs" prior to beginning any of these lab scenarios.

- To load your router and switch configs, first log into your INE.com Members account, then navigate to the "Rack Rental" tab, and click on "Control Panel" >> "Click here to choose a configuration to be loaded on your Voice Rack", then choose the appropriately named selection for this module. This will only load your router and switch configs (pstn, r1, r2, r3, sw1, sw2).

- To load your CUCM server initial configs, please log into the CUCM Admin WebUI, and use the “Bulk Administration” tool to upload & import the appropriately named “DD-ModXX-CUCM-Startup.tar" file that was included in your Members account for this module. Here are two brief demonstration videos on how to successfully accomplish this upload and import into CUCM: Importing or Exporting Configurations in CSV file from CUCM (15 mins), and Fixing Any Issues That May Arise With Importing CSS and CdPTP & CngPTPs (8 mins).

http://ine.com/l/vo_rrag

http://members.ine.com/




http://ieclass.ine.com/p98538031/








12.1 SIP Trunk to AT&T ITSP for All CorpHQ PSTN Call Routing

Provision your CUCM Cluster to provide both inbound and outbound PSTN calling services to all Local, National and International numbers (no emergency services) for phones at the CorpHQ site via a new AT&T ITSP SIP Trunk, as their primary routing option. AT&T will now be routing all inbound calls to CorpHQ phone DID's via this SIP trunk primarily as well.

Provision a demarcation point between your CUCM Cluster and the AT&T SIP Trunk using the CorpHQ router

AT&T has assigned you FQDNs to use for connectivity that provide for redundancy and load balancing of your calls to them, and they are listed below. Also, a few more FQDNs have been pre-provisioned for you that define things such as the CorpHQ router. Ensure you use these FQDNs, that AT&T sees calls coming from your CorpHQ FQDN (not IP address in SIP messages), and that all calls are load balanced to AT&T. The DNS server and FQDNs are as follows:

o DNS Server IP Address: 177.1.10.110

o AT&T IP FlexReach FQDNs: sip1.att.com

o sip2.att.com

o INE CorpHQ-R1 FQDN: corphqr1.ine.com

Ensure that IP's from both the "Inside" and the "Outside" of the CUBE are hidden (CUCM, IP Phones, etc are hidden from AT&T, and vice-versa that AT&T's IP addresses are hidden from the CUCM and IP Phones)

AT&T has told you to send them all calls over this SIP Trunk in the Full +E.164 format (include +)

Ensure that a backup route to the PSTN via ISDN on the CorpHQ router is still fully functional






12.2 Conforming to ITSP Requirements :: Various SIP-Attributes

Ensure that any error messages from CUCM are passed through, intact, to AT&T

Ensure CLID and CNAM are sent to AT&T and seen on the PSTN Phone

Ensure SIP signaling integrity is maintained end-to-end across CUBE throughout the course of the call

Ensure G.729 Annex B is allowed to be negotiated by CUBE and is interoperable with CUCM, otherwise AT&T tells you that calls will fail

AT&T has informed us that they will only support RFC2833 as a DTMF Relay type, and typically use the payload value of 100, although it is possible that a value of 96 could be sent. Ensure that DTMF Relay is negotiated properly, and that even if AT&T happens to send an RTP value of 96, that DTMF works for PSTN users calling into the CUCM and any VM or IVR they might encounter.

AT&T has informed us that they will only respond to a call setup if the SDL is sent in the SIP INVITE header, therefore ensure that this is supported, however ensure CUCM is not forced to allocate a MTP to support this feature request

12.3 Conforming to ITSP Requirements :: SIP Header Conversions

After this trunk has been functioning for a few days, you being to notice that if a call goes on for longer than 30 minutes, that it drops exactly at the 30 minute mark. After you troubleshoot this by setting up a live call, you find that exactly at that 30 minute mark, AT&T is sending you a SIP UPDATE message, to which the CUCM doesn't respond well to, since all mid-call messaging is passed through. To solve this issue, configure CUBE to instruct AT&T that you do not support SIP UPDATE messages

Your company has an engineering department that sometimes has a need to perform tests that involve calls that can sometimes last many hours at a time. Determine what value CUCM is passing to (and ultimately through) CUBE as the maximum call duration, and then configure CUBE to inform AT&T that each call should be allowed to stay active for a maximum of 24 hours. Do not modify any parameters in CUCM to accomplish this task.

12.4 Advance Call Admission Control Mechanisms with CUBE

Configure CUBE to send the "User-Busy" flag with a reason indicating a lack of resources, back to any party sending CUBE a SIP INVITE message, when any of the following conditions are met:

o If the router's CPU rises above 90%, block all calls until it falls below 50%

o If the router's I/O memory rises above 80%, block all calls until it falls below 65%

o If there is a sudden spike of SIP INVITE messages - where the router receives more than 50 INVITES and/or more than 8 calls in less than a 100ms sliding window of time - block all calls until it falls below that value






12.5 Skype SIP Trunk for Branch2 Site

Being quite happy with your existing SIP Trunks, you now wish to incorporate the usage of the low-cost, high performance option of Skype to your Branch2 site in Amsterdam. Provision another CUBE element at the Branch2 site and switch outbound call routing for that site accordingly

Ensure that any error messages from CUCM are passed through, intact, to AT&T

Ensure CLID and CNAM are sent to AT&T and seen on the PSTN Phone

Ensure SIP signaling integrity is maintained end-to-end across CUBE throughout the course of the call

Ensure that calls from phones at the Branch2 site now use the CUBE to Skype trunk as their primary route for all outbound Local, National and International calls. Skype cannot assume ownership of the Branch2 site's existing ISDN DID range of phone numbers, however it has already been provisioned with a new number local to Amsterdam and is already configured to send inbound calls to the new range of numbers listed below:

o +312075745XX

Skype and the Branch2 router already have FQDNs provisioned and they are as follows:

o INE Branch1-R2 FQDN: branch2r3.ine.com

o Skype Business SIP Trunk FQDN: sip.skype.com

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Ine Vo Dd Wb Vol1 Mod12 Cube Tasks