Wow, what an experience it was, trying to pass Cisco CIPT2 300-075 exam during the last 5 weeks. So much was on the line as if I did not pass this exam by 17th of June, 2016, all my CCNP R&S and CCNP Voice was expiring, so I would be facing 7 exams to re-certify as CCNP in both technologies. Unfortunately, I had to tackle it 3 times to pass this exam, and got lucky on 3rd attempt. My first attempt was a lame attempt as I failed by 7 questions, the second attempt was a little bit more decent, failing by a SINGLE question. The passing mark for CIPT2 300-075 exam is 860/1000, which makes one question valued somewhere between 13 pts to 17 pts depending on the weight of the question Cisco is throwing at you. But today, I passed the exam and what a Roller Coaster ride this exam was, I’ve passed the exam with the exact passing score! Finally, the lady luck is on my side. After two failures, trying to tame the beast, I’ve studied so many hours trying to understand the VCS components and finally got a full grasp of the concept and basic configuration. Over 3 days of long weekend, I cranked out 30 hours of study time for VCS C and E studies (no pain, no gain! I am thankful that I’ve failed the second time by one question. I was forced to try my best). Oh, what a feeling!
I want to share some of my study notes with you so, you don’t have to do it the hard way like me, but I urge you to spend some time reading Cisco documentations, watch videos from CiscoLive and read the official study books front-to-end before jumping into the full study drive mode. I hope my notes will help someone on their way to becoming a CCNP-Collaboration. My notes are based on Cisco documents but also comes from the live environment and my experience, so it might not be 100%, but if you disagree with me on those questions, then show me your proof that you are in the right and I am in the wrong with a live Cisco documentations referencing the page and line number. As always, if you cannot avoid it, try to face it with a dignity or try to enjoy it!!! I would choose the latter…. 🙂
1. Regional configuration of Cisco VoIP environment
Note: Cisco Best practice, (G.729/24K) to compress BW for regions. Hardware MTP only supports G.711 a-law and G.711 u-law. Also regions will need transcoders if multiple codecs are deployed, NOT hardware MTP.
2. While using Query wizard to configure the trace and log central feature to collect install logs.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/7_1_2/rtmt/RTMT/rttlc.html
“The time zone of the client machine provides the default setting for the Select Reference Server Time Zone field. All the standard time zones, along with a separate set of entries for all time zones that have Daylight Saving settings, display in the Select Time Zone drop-down list box.”
“Trace and Log Central downloads the file with a time range that is based on your Selected Reference Server Time Zone field. If you have servers in a cluster in a different time zone, TLC will adjust for the time change and get files for the same period of time. For example, if you specify files from 9:00 AM to 10:00 AM and you have a second server (server x) that is in a time zone that is one hour ahead, TLC will download files from 10:00 AM to 11:00 AM from server x.”
3. Standardization of caller addresses between H.323 and SIP endpoints.
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration_guide_chapter09186a00 800eadee.htmlhttp://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X8-2/Cisco-VCS-Basic-Configuration-Control-with-Expressway-Deployment-Guide-X8-2.pdf (Page 17)
“The pre-search transform configuration described in this document is used to standardize destination aliases originating from both H.323 and SIP devices. ”
“The following transform modifies the destination alias of all call attempts made to destination aliases which do not contain an ‘@’. The old destination alias has @example.com appended to it. This has the effect of standardizing all called destination aliases into a SIP URI format.”
From VCS and CUCM Deployment guide:
“Thus, a transform is needed to ensure that the dialed number is transformed into a consistent form, in this case to add the domain (vcs.domain) if required.”
4. CUCM Extension Mobility characteristics
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a0080153e60.html#wp1092734
“Able to adopt a user profile even when no user is logged in”
“Almost same attributes as a physical device”
5. A globalized dial plan, 3 ways enabling ingress gateways to process calls.
Configure the called-party transformation settings for incoming calls on H.323 gateways.
Configure translation patterns in the partitions used by the gateway calling search space
Configure the gateway with prefix digits to add necessary country and region codes.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/dialplan.html#16685
“Localized Call Ingress on Gateways
The called and calling numbers delivered into the Unified Communications system by external networks (for example, the PSTN) are typically localized. The form of the numbers may vary, depending on the service provider’s configuration of the trunk. As a gateway is connected to a PSTN trunk, the system administrator must work with the PSTN service provider to determine the applicable signaling rules to be used for this specific trunk. As calls are delivered into the system from the trunk, some of the information about the calling and called numbers will be provided explicitly and some of it will be implied. Using this information, the system must derive the calls’ globalized calling and called party numbers.
The globalization of the called party number can be implemented through one of the following methods:
In the gateway configuration, configure Call Routing Information > Inbound Calls, where the quantity of significant digits to be retained from the original called number and the prefix digits to be added to the resulting string are used to globalize the called number. The prefix digits should be used to add the applicable + sign and country, region, and city codes.
Place translation patterns in partitions referenced by the gateway’s calling search space. The translation patterns should be configured to match the called party number form used by the trunks connected to the gateway, and should translate it into the global form. The prefix digits should be used to add the applicable + sign and country, region, and city codes.
Use the incoming call’s called party transformation settings available on the gateway and on the gateway’s device pool. There you can define strip and prefix digit instructions or alternatively configure a called party transformation calling search space per numbering type.
The globalization of the calling party number should be implemented by using the Incoming Calling Party Settings configured either on the gateway directly or in the device pool controlling the gateway.”
6. 2 types of devices are affected when an engineer changes the DSCP for Video Calls service parameter
Read “Set DSCP Values”.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/10_6/CJAB_BK_C56DE1AB_00_cisco-jabber-106-deployment-and-installation-guide/CJAB_BK_C56DE1AB_00_cisco-jabber-106-deployment-and-installation-guide_appendix_01111.html
7. Cisco VCS uses 3 Presence status of endpoints for monitoring
http://www.cisco.com/c/en/us/td/docs/telepresence/infrastructure/articles/vcs_monitors_presence_ status_endpoints_kb_186.html
in-all
call-ended
registration
8. 3 steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones
1. configure an SRST reference
2. Configure the SIP registrar
3. Configure voice register pool
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server expires max 600 min 60
!
voice register pool 10
id network 172.26.10.0 mask 255.255.255.0
dtmf-relay rtp-nte cisco-rtp sip-notify
codec g711ulaw
no vad
!
voice register pool 11
id network 172.26.11.0 mask 255.255.255.0
dtmf-relay rtp-nte cisco-rtp sip-notify
codec g711ulaw
no vad
!
sip-ua
registrar ipv4:172.26.10.240 expires 600
(172.26.10.240 is the SRST gateway IP address).
Don’t forget on UCM SRST reference configuration for gateway:
On UCM SRST reference configuration for gateway
SIP Network/IP Address 172.26.10.240
SIP Port 5060
9. Device Mobility – overlapping parameters for roaming
Location
Network Locale
MRGL
Reference: https://supportforums.cisco.com/document/77096/device-mobility
“The overlapping parameters for roaming-sensitive settings are Media Resource Group List, Location, and Network Locale. The overlapping parameters for the Device Mobility-related settings are Calling Search Space (called Device Mobility Calling Search Space at the device pool), AAR Group, and AAR Calling Search Space. Overlapping parameters configured at the phone have higher priority than settings at the home device pool and lower priority than settings at the roaming device pool.”
10. VCS Control routing configuration, user dial brchoi and call gets routed to brchoi@cisco.com
search rule
http://www.manualslib.com/manual/841592/Cisco-Telepresence.html?page=168
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration_guide_chapter09186a00 800eadee.htmlhttp://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X8-2/Cisco-VCS-Basic-Configuration-Control-with-Expressway-Deployment-Guide-X8-2.pdf (Page 17)
“The pre-search transform configuration described in this document is used to standardize destination aliases originating from both H.323 and SIP devices. The following transform modifies the destination alias of all call attempts made to destination aliases which do not contain an ‘@’. The old destination alias has @example.com appended to it. This has the effect of standardizing all called destination aliases into a SIP URI format.”
From VCS and CUCM Deployment guide:
“Thus, a transform is needed to ensure that the dialed number is transformed into a consistent form, in this case to add the domain (vcs.domain) if required.”
11. Configure VG310/VG350 and enable call pickup feature
SCCP gateway
You must check this on a running CUCM. Check CUCM configuration and VG350 gets configurred as SCCP only and then the endpoints can be configured to do a call pick-up.
Check CUCM configuration and VG350 gets configurred as SCCP only and then the endpoints can be configured to do a call pick-up.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmsys/CUCM_BK_SE5FCFB6_00_cucm-system-guide-100/CUCM_BK_SE5FCFB6_00_cucm-system-guide-100_chapter_0100110.html#CUCM_RF_V83D221A_00
12. Intracluster URI dialing configuration
URI Dialing within the same cluster, follow these steps:
Step 1: Configure the URIs to the users
Step 2: Associate the directory URIs to directory numbers
Step 3: Assign the default directory URI (Configure the directory URI partition and calling search space)
Step 4: Configure the SIP profile in your network. (Configure a setting for the Dial String Interpretation drop-down list box and apply the setting for all the SIP profiles in your network. Check the Use Fully Qualified Domain Name in SIP Requests check box for all the SIP profiles in your network.)
13. Enabling video desktop sharing between CUCM video endpoint and Cisco VCS video endpoint.
Use BFCP
https://www.ciscolive.com/online/connect/sessionDetail.ww?SESSION_ID=88574&backBtn=true
BRKCOL-2540 – Video call control and management migration to CUCM (2015 Cancun) – 90 Mins
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/bccm-861-cm/b06siprf.html
Table 76-1 SIP Profile Configuration Settings
Allow Presentation Sharing using BFCP
If the box is checked, Cisco Unified Communications Manager is configured to allow supported SIP endpoints to use the Binary Floor Control Protocol to enable presentation sharing.
The use of BFCP creates an additional media stream in addition to the existing audio and video streams. This additional stream is used to stream a presentation, such as a PowerPoint presentation from someone’s laptop, into a SIP videophone.
If the box is unchecked, Cisco Unified Communications Manager rejects BFCP offers from devices associated with the SIP profile by setting the BFCP application line and associated media line ports to 0 in the answering SDP message. This is the default behavior.
Note BFCP is only supported on SIP networks. BFCP must be enabled on all SIP trunks, lines, and endpoints for presentation sharing to work. BFCP is not supported if the SIP line or SIP trunk uses MTP, RSVP, TRP or Transcoder.
For more information on BFCP, refer to the Cisco Unified Communications Manager System Guide.
14. “Src-port=”25723″ Detail=”Incorrect authentication credential for user”” error
The Expressway-C Traversal Client username/password do not match the Expressway-E Traversal Server username/password.
https://www.ciscolive.com/online/connect/sessionDetail.ww?SESSION_ID=89093&backBtn=true
The Expressway-C is a Client and the Expressway-E is the server. They have client to server relationship. Expressway-C is a Traversal Client.
Traversal Zone
When the peer address is configured as an IP address or the peer address does not match the Common Name (CN), you see this in the logs:
Event=”Outbound TLS Negotiation Error” Service=”SIP” Src-ip=”10.48.80.161″
Src-port=”25697″ Dst-ip=”10.48.36.171″ Dst-port=”7001″ Detail=”Peer’s TLS
certificate identity was unacceptable” Protocol=”TLS” Common-name=”10.48.36.171″
When the password is incorrect, you see this in the Expressway-E logs:
Module=”network.ldap” Level=”INFO”: Detail=”Authentication credential found in
directory for identity: traversal”
Module=”developer.nomodule” Level=”WARN” CodeLocation=”ppcmains/sip/sipproxy/
SipProxyAuthentication.cpp(686)” Method=”SipProxyAuthentication::
checkDigestSAResponse” Thread=”0x7f2485cb0700″: calculated response does not
match supplied response, calculatedResponse=769c8f488f71eebdf28b61ab1dc9f5e9,
response=319a0bb365decf98c1bb7b3ce350f6ec
Event=”Authentication Failed” Service=”SIP” Src-ip=”10.48.80.161″
Src-port=”25723″ Detail=”Incorrect authentication credential for user”
Protocol=”TLS” Method=”OPTIONS” Level=”1″
15. An effective backup method to access TEHO destinations in case the call limit triggers
LRG
https://books.google.com.au/books?id=80iuCwAAQBAJ&pg=PT151&lpg=PT151&dq=device+pool+used+to+globalize+dial+plan&source=bl&ots=uyI8e5Jt4L&sig=hZ-hEJ-XCgJT7pZHCrGR89Q5TR4&hl=en&sa=X&ved=0ahUKEwi0n-33iqLNAhXFIqYKHWdaAVEQ6AEIQjAF#v=onepage&q=device%20pool%20used%20to%20globalize%20dial%20plan&f=false
“If TEHO is configured, the appropriate TEHO Gateway is used for the PSTN call. The TEHO route list can include the Default Local Route Group setting as a backup path. In this cas, if the primary (TEHO) path is not available, the gateway taht is referenced by the local route group of the applicable device pool will be used for the backup path. If the device pool selection is not static, but Cisco Unified device mobility is used, the gateway of the roaming site will be used as a backup for the TEHO path. …”
16. Functionalities of subzones in a Cisco VCS deployment
Apply registration, authentication, and media encryption policies
Manage bandwidth to restrict standard definition endpoints from using more than 2 Mb of bandwidth.
Bandwidth management
The Local Zone’s subzones are used for bandwidth management. After you have set up your subzones you can apply
bandwidth limits to:
– individual calls between two endpoints within the subzone
– individual calls between an endpoint within the subzone and another endpoint outside of the subzone
– the total of calls to or from endpoints within the subzone
For full details of how to create and configure subzones, and apply bandwidth limitations to subzones including the
Default Subzone and Traversal Subzone, see the Bandwidth control section.
Registration, authentication and media encryption policies
In addition to bandwidth management, subzones are also used to control the VCS’s registration, authentication and
media encryption policies.
17. Enabling SAF Call Control Discovery
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_0_2/ccmfeat/fsgd-802- cm/fscallcontroldiscovery.pdf
1. the SIP or H.323 trunk
2. hosted DN patterns
3. Hosted DN groups
18. Cisco VCS Expressway traversal call licenses
http://www.cisco.com/c/en/us/support/docs/unified-communications/telepresence-video-communication-server-vcs/118872-technote-vcs-00.html
According to the document VCS, Gatekeepers and Border Controllers. SIP Trunk is treated as a device by Cisco, but it is not a real device, so not used for licensing.
19. Devices or applications support call preservation
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/CUCM_BK_CD2F83FA _00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system- guide_chapter_01011.html#CUCM_RF_C98194B0_00
The following devices and applications support call preservation. If both parties connect through one of the following devices, Cisco Unified Communications Manager maintains call preservation:
Cisco Unified IP Phones
SIP trunks
Software conference bridge
Software MTP
Hardware conference bridge (Cisco Catalyst 6000 8 Port Voice E1/T1 and Services Module, Cisco Catalyst 4000 Access Gateway Module)
Transcoder (Cisco Catalyst 6000 8 Port Voice E1/T1 and Services Module, Cisco Catalyst 4000 Access Gateway Module)
Non-IOS MGCP gateways (Catalyst 6000 24 Port FXS Analog Interface Module, Cisco DT24+, Cisco DE30+, Cisco VG200)
Cisco IOS H.323 gateways (such as Cisco 2800 series, Cisco 3800 series)
Cisco IOS MGCP Gateways (Cisco VG200, Catalyst 4000 Access Gateway Module, Cisco 2620, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 3810)
Cisco VG248 Analog Phone Gateway
The following devices and applications do not support call preservation:
Annunciator
H.323 endpoints such as NetMeeting or third-party H.323 endpoints
CTI applications
TAPI applications
JTAPI applications
Call Preservation Scenarios
20. Global Dial Plan Replication prevent the local cluster from routing VIP number 6666666666 to the remote cluster.
Create a block learned pattern.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100_chapter_011101.html#CUCM_RF_L56BD6F6_00
Learned pattern can be configured but there is no mentioning of transformation pattern configuration. “Create a block learned pattern” can be used to prevent Global Dial Plan Replication within local cluster.
21. URI calling within the same cluster configuration
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmsys/CUCM_BK_CD2F83FA_00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system-guide_chapter_0101111.html#CUCM_TK_SBE2D597_00
URI Dialing within the same cluster, follow these steps:
Step 1: Configure the URIs to the users
Step 2: Associate the directory URIs to directory numbers
Step 3: Assign the default directory URI (Configure the directory URI partition and calling search space)
Step 4: Configure the SIP profile in your network. (Configure a setting for the Dial String Interpretation drop-down list box and apply the setting for all the SIP profiles in your network. Check the Use Fully Qualified Domain Name in SIP Requests check box for all the SIP profiles in your network.)
22. 2 steps must you take when implementing TEHO in your environment
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/7x/uc7_0/dialplan.html
Implement local failover
Implement centralized failover
23. Globalization dialing functions enhancement since CUCM 7.X and later
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/9x/uc9x/dialplan.html (benefits of new design approach)
AAR
CER
TEHO
24. 2 commands verify Cisco IP Phone registration
show ephone registered
show sip-ua status registrar
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/troubleshooting/guide/ts_phreg.html (see the steps)
Step 3 show sip-ua status registrar
Use this command to display all the SIP endpoints currently registered with the contact address.
Router# show sip-ua status registrar
Line destination expires(sec) contact
============ =============== ============ ===============
91021 192.168.0.3 227 192.168.0.3
91011 192.168.0.2 176 192.168.0.2
95021 10.2.161.50 419 10.2.161.50
95012 10.2.161.50 419 10.2.161.50
95011 10.2.161.50 420 10.2.161.50
95500 10.2.161.50 420 10.2.161.50
94011 10.2.161.40 128 10.2.161.40
94500 10.2.161.40 129 10.2.161.40
I have cross checked with my Voice Gateway and found that ‘show ephone registered’ and ‘show sip-ua status registrar’.
Router#show ephone registered
ephone-1[0] Mac:6C30.4D57.8CD5 TCP socket:[1] activeLine:0 whisperLine:0 REGISTERED
mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0
IP:10.30.11.34 * 7962 keepalive 4929 music 0 1:101 CM Fallback
sp1:01800008584 sp2:01800654112 sp3:00362456600
ephone-2[1] Mac:555D.0608.45B6 TCP socket:[-1] activeLine:0 whisperLine:0 UNREGISTERED
mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0
IP:10.30.11.15 * 6921 keepalive 7 music 0
ephone-3[2] Mac:448D.0407.6BE9 TCP socket:[4] activeLine:0 whisperLine:0 REGISTERED
mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0
IP:10.30.11.124 * 6921 keepalive 4938 music 0 1:103 CM Fallback
ephone-4[3] Mac:544D.0907.532C TCP socket:[12] activeLine:0 whisperLine:0 REGISTERED
mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0
IP:10.30.11.25 * 6921 keepalive 4931 music 0 1:2 CM Fallback
25. Enalbe presence and extension mobility to branch office phones during a WAN failure. Cisco Unified Communications Manager Express in SRST mode
http://www.ciscopress.com/articles/article.asp?p=1744068&seqNum=4
Cisco Unified SRST does not support enhanced features, such as Presence or Cisco Extension Mobility. Message Waiting Indicator (MWI) is also not supported in fallback mode.
26. Configured a Cisco EX60 to register with a Cisco VCS Control, but phone is not registering with VCS C. What’s missing in the configuration.
Click to access Cisco-VCS-Basic-Configuration-Single-VCS-Control-Deployment-Guide-X8-1.pdf
EX60 (uses H.323 and SIP protocol)
H.323 ID user.two.mxp@example.com
H.323 E.164 7654321
Gatekeeper IP Address vcsc.internal-domain.net
SIP URI user.two.mxp@example.com
SIP Proxy1 vcsc.internal-domain.net
EX90 (uses SIP protocol)
SIP URI user.one.ex90@example.com
SIP Proxy1 vcsc.internal-domain.net
27. Your company’s internal number is 4 digit dialing, how to present this as 10-digit number to external clients?
Use “calling party transformation pattern”
https://supportforums.cisco.com/discussion/9848251/external-phone-mask-vs-calling-party-transformation-mask
“An advantage of using Calling Party Transformation Mask is that it allows you to change the Calling party number for a bunch of phones easily. Lets say you have a 100 phones that you need to change the 10 digit number. Rather than going to each phone and change the setting individually, you can do it at the Calling Party tranformation mask.”
“Another advantage is that if you want to change Calling Party number that gets displayed to external users, you can modify that easily with the transformation masks. It also gives you the flexibility of sending different calling party numbers to differnt destinations. For example, for local calls you can dislay the 7 digit number; for long distance you can display 10 digits and for international you can display country code +10 digits.”
28. Default region configurable items on CUCM?
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_0_1/ccmcfg/bccm-801- cm/b02regio.html#wp1077135
Audio Codec
Video Call Bandwidth
Link Loss Type
29. During Intercluster URI dialing, an error message “Local cluster cannot connect to the ILS network” comes up, what could be possible issues?
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmfeat/CUCM_BK_CEF0C471_00_cucm-features-services-guide-90/CUCM_BK_CEF0C471_00_cucm-features-and-services-guide_chapter_011111.pdf (Page. 8)
The Tomcat certificates do not match.
The ILS authentication password does not match.
One cluster is using TLS certificate, and the other is using Password.
30. 2 technologies not utilising MTP.
http://www.cisco.com/c/en/us/td/docs/ios/voice/cube/configuration/guide/vb_8241.html
DTMF inband RTP-NTE (rfc2833)
SIP Delay Offer
Note 1: H.323 fast start:
https://learningnetwork.cisco.com/thread/65786
DTMF inband RTP-NTE (rfc2833) requires MTP only in CM 4.0, 5 and in later versions of CUCM, and lMTP requirement was removed when supporting RFC 2833 DTMF)
Note 2: If both endpoints support NTE, then no MTP is required. http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/8x/uc8x/media.html#wp1046314
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