Deploying SIP Trunks with Cisco Unified Border Element (CUBE/vCUBE) Enterprise Hussain Ali, CCIE# 38068 (Voice, Collaboration) Technical Marketing Engineer Dilip Singh, CCIE# 16545 (Collaboration) Technical Leader BRKCOL-2125
Deploying SIP Trunks with Cisco Unified Border Element (CUBE/vCUBE) EnterpriseHussain Ali, CCIE# 38068 (Voice, Collaboration)Technical Marketing Engineer
Dilip Singh, CCIE# 16545 (Collaboration)Technical Leader
BRKCOL-2125
• SIP Trunking and CUBE Overview
• CUBE Architecture (Physical & Virtual)
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE (Call Recording, Multi-Tenancy, HA)
• CUBE Serviceability
• Futures & Key Takeaways
Agenda
CUBE Overview
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
CUBE (Enterprise) Product Portfolio
2900 Series ISR-G2 (2901, 2911, 2921, 2951)
ASR 1004/6 RP2
Active Concurrent Voice Calls Capacity
CP
S
<5
8-12
50-150
14-16K<50 500-600 900-1000
3900 Series ISR-G2 (3925, 3945)
17
3900E Series ISR-G2 (3925E, 3945E)
2000-2500
20-35
4
800 ISR
7000-10,000
50-100
12K-14K
ASR 1002-X
4500-6000
ISR 4451-X
ASR 1001-X
4000
ISR 4431
ISR-4K (4321, 4331)
ISR 4351
Introducing CUBE on
CSR
vCUBE [Performance
dependent on vCPU and
memory]
Note: SM-X-PVDM module
supported on XE3.16 or
later for ISR 4K platforms
BRKCOL-2125 4
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
CUBE Session Capacity Summary
Platform CUBE SIP-SIP Sessions (Audio)
NanoCUBE (8XX and SPIAD Platforms) 15 - 120
2901 – 4321 100
2911 – 2921 200 – 400
4331 500
2951 600
3925 – 3945 800 – 950
4351 1000
3925E – 3945E 2100 – 2500
4431 3000
4451 6000
ASR1001-X 12000
ASR1002-X 14000
ASR1004/1006 RP2 16000
For Your
Reference
BRKCOL-2125 5
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Introducing IOS-XE Release 16 • New OS from the platform team with the intent of consolidating OS’ on different
product portfolio
• UX will be the same as IOS-XE, no difference to end user
• IOS-XE Release 16.3.1 support for UC (CUBE, CME, SRST)
Impacts XE based (ASR1K, ISR4K, and vCUBE) platforms
There will be no CUBE 11.5.1 for the XE based platforms [ASR1K, ISR4K, vCUBE]. CUBE 11.5.2 (July 2016 release) will have newer and March 2016 features for the XE based platforms introduced in IOS-XE release 16.3.1
IOS-XE 16 requires a minimum of ASR1001-X, 1002-X, 1004/1006 RP2, ESP20 (Embedded Service Processor, SIP40 (SPA Interface processor)
It will include all features up to and including IOS-XE 3.17 as well
• Due to new hardware requirements, customers will have the following migrations options as IOS-XE 3.17 rebuilds will stop by June 2017
Replace unsupported ASR1K hardware and upgrade to IOS-XE 16.3.1 or later and continue to enjoy new feature set/support for any issues
Drop using new feature set and move back to IOS-XE3.16 long maintenance release for longer support
BRKCOL-2125 6
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
ISR G2 ASR 1K / ISR-4K/vCUBE (CSR)
CUBE Vers. 2900/ 3900 FCS CUBE Vers. IOS XE Release FCS
11.1.0 15.5(3)M July 2015 11.1.0 3.16 15.5(3)S July 2015
11.5.0 15.6(1)T Nov 2015 11.5.0 3.17 15.6(1)S Nov 2015
CUBE Vers.
2900/ 3900 FCSCUBE Vers.
IOS XE Release 16 2 FCS
11.5.14 15.6(2)T14 Mar 2016 N/A 3 16.2.13 Mar 2016
11.5.2 15.6(3)M1 Dec 2016 11.5.23 16.3.2/16.4.13 Nov 2016
EOL EOL EOL 11.6.0 16.5.1 Mar 2017
7
CUBE Software Release Mapping
2 IOS-XE 16 requires a minimum of ASR1001-X, 1002-X, 1004/1006 RP2, ESP20 (Embedded Service Processor, SIP40 (SPA Interface
processor)
3 IOS-XE release 16.2.1 does not support CUBE functionality on the platforms. There is no CUBE version 11.5.1 for the XE based
platforms. All CUBE features from 11.5.0 (IOS-XE 3.17) and earlier versions along with CUBE 11.5.1 (March 2016 release) on ISR G2
are included in CUBE release 11.5.2 for the IOS-XE based platforms, IOS-XE release 16.3.1 [July 2016 release]
4 IOS 15.6(2)T will show CUBE Release version to be 12.0.0 but due to DDTS# CSCuz43735, rebuilds for this release train will align
to CUBE release 11.5.1, that is 15.6(2)T1/T2/T3/T4 and so on will be CUBE version 11.5.1
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
ISR G2 CUBE EntASR Parity
with ISR
ASR 1K Series
CUBE Vers. 2900/ 3900 FCS CUBE Vers. IOS XE Release FCS
8.5 15.1(2)T July 2010 <50% 1.4 3.2 15.1(1)S Nov 2010
8.6 15.1(3)T Nov 2010 <50% 1.4.1 3.3 15.1(2)S March 2011
8.7 15.1(4)M April 2011 ~50% 1.4.2 3.4 15.1(3)S July 2011
8.8 15.2(1)T July 2011 ~70% 1.4.3 3.5 15.2(1)S Nov 2011
8.9 15.2(2)T Nov 2011 >80% 1.4.4 3.6 15.2(2)S Mar 2012
9.015.2(3)T/
15.2(4)MMar 2012 >85% 9.0 3.7 15.2(4)S July 2012
9.0.1 15.3(1)T Oct 2012 >95% 9.0.1 3.8 15.3(1)S Oct 2012
9.0.2 15.3(2)T Mar 2013 >95% 9.0.2 3.9 15.3(2)S Mar 2013
9.5.1 15.3(3)M1 Oct 2013 >95% 9.5.1 3.10.1 15.3(3)S1 Oct 2013
10.0.0 15.4(1)T Nov 2013 >95% 10.0.0 3.11 15.4(1)S Nov 2013
10.0.1 15.4(2)T Mar 2014 >95% 10.0.1 3.12 15.4(2)S Mar 2014
CUBE Software Release Mapping – Earlier Releases
8
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
ISR G2 CUBE EntASR Parity
with ISR
ASR 1K / ISR-4K Series
CUBE Vers.
2900/ 3900 FCSCUBE Vers.
IOS XE Release FCS
10.0.2 15.4(3)M July 2014 >95% 10.0.2 3.13 15.4(3)S July 2014
10.5.0 15.5(1)T Nov 2014 >95% 10.5.0 3.14 15.5(1)S Nov 2014
11.0.0 15.5(2)T Mar 2015 >95% 11.0.0 3.15 15.5(2)S Mar 2015
CUBE Software Release Mapping – Earlier Releases
9
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Proven Interoperability and Interworking with
Service Providers Worldwide
Cisco Interoperability Portal:
www.cisco.com/go/interoperability
• Validated with Service
Providers World-Wide
• Independently Tested
with 3-Party PBXs in
tekVizion Labs
• Standards basedVerified by
CUBE Interoperability
BRKCOL-2125 10
• SIP Trunking and CUBE Overview
• CUBE Architecture (Physical & Virtual)
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE (Call Recording, Multi-Tenancy, HA)
• CUBE Serviceability
• Futures & Key Takeaways
Agenda
CUBE Call Flow
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
CUBE Call Processing
Actively involved in the call treatment, signaling and media streams
SIP B2B User Agent
Signaling is terminated, interpreted and re-originated
Provides full inspection of signaling, and protection against malformed and malicious packets
Media is handled in two different modes: Media Flow-Through
Media Flow-Around
Digital Signal Processors (DSPs) are only required for transcoding (calls with dissimilar codecs)
IP
CUBE
CUBE
IP
Media Flow-Around
Only Signaling is terminated on CUBE
Media bypasses the Cisco Unified Border Element
Media Flow-Through
Signaling and media terminated by the Cisco Unified Border Element
Transcoding and complete IP address hiding require this model
BRKCOL-2125 13
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Cisco Unified Border Element Basic Call Flow
1. Incoming VoIP setup message from originating endpoint
2. This matches inbound VoIP dial peer 1 for characteristics such as codec, VAD, DTMF method, protocol, etc.
3. Match the called number to outbound VoIP dial peer 2
4. Outgoing VoIP setup message
Incoming VoIP Call Outgoing VoIP Call
dial-peer voice 1 voipdestination-pattern 1000session protocol sipv2 session target ipv4:1.1.1.1codec g711ulaw
dial-peer voice 2 voipdestination-pattern 2000session protocol sipv2session target ipv4:2.2.2.2codec g711ulaw
Originating Endpoint -
1000
TerminatingEndpoint –
2000
CUBE
voice service voip
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
BRKCOL-2125 14
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
CUBE
10001.1.1.1
20002.2.2.2
Incoming VoIP Call LegMatches an Incoming Dial-peer
Outgoing VoIP Call LegMatches an Outbound Dial-peer
INVITE /w SDP INVITE /w SDP
100 TRYING100 TRYING
VRF1 – 10.10.10.10 20.20.20.20 – VRF2
c= 1.1.1.1m=audio abc RTP/AVP 0
c= 20.20.20.20m=audio xxx RTP/AVP 0
180 RINGING180 RINGING
200 OK200 OK c= 2.2.2.2
m=audio uvw RTP/AVP 0c= 10.10.10.10m=audio xyz RTP/AVP 0
RTP (Audio)
ACKACK
1.1.1.1 10.10.10.10 20.20.20.20 2.2.2.2
Understanding the Call flow
BYE BYE
200 OK200 OK
BRKCOL-2125 15
CUBE ArchitectureISR G2 vs ASR1K vs ISR 4K vs vCUBE (CUBE on CSR)
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
ASR/ISR-4K & ISR-G2 Architecture Comparison
ISR: Pkt fwd’ing and signaling are handled by the same CPU
ASR: Pkt fwd’ing and signaling are handled by different CPUs
‒ ESP must be programmed or instructed by the control plane to do specific media functions
‒ Performed by Forwarding Plane Interface (FPI)
I/O ESP I/O
Kernel
IOS-XEIOS-XE
Msg I/f
Control Plane
Data (Forwarding) Plane
ASR/ISR-4K (IOS-XE) Architecture
RPIOS
CPU
I/O I/O
Control
Plane
Data Plane
ISR G2 Architecture
Sig
nalin
g
Sig
nalin
g
MediaBRKCOL-2125 17
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Introducing vCUBE (CUBE on CSR 1000v)Architecture
• CSR (Cloud Services Router) 1000v runs on a Hypervisor – IOS XE without the router
Console Mgmt ENET Ethernet NICsFlash / DiskMemoryVirtual CPU
RP (control plane)
Chassis Mgr.
Forwarding Mgr.IOS-XE
Kernel (incl. utilities)
ESP (data plane)
Chassis Mgr.
Forwarding Mgr.
QFP Client / Driver
FFP code
Hypervisor
Hardware
vSwitch NIC
GE GE…X86 Multi-Core CPU Memory Banks
ESXi Container
CUBE signaling CUBE media processing
CSR 1000v (virtual IOS-XE)
BRKCOL-2125 18
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Introducing vCUBE (CUBE on CSR 1000v) –Cont’d
• CSR1000v is a virtual machine, running on x86 server (no specialized hardware) with physical resources are managed by hypervisor and shared among VMs
• Requires APPX (No TLS/SRTP) or AX (All vCUBE features) CSR licensing package to access voice CLI and increase throughput from 100 kbps default. CUBE Licensing follows ASR1K SKUs and still trust based
• No DSP based features (transcoding/inband-RFC2833 DTMF/ASP/NR) available
• Specs based hardware supported but performance benchmarked for Cisco UCS B and C series only
• vCUBE Tested Reference Configurations [UCS base-M2-C460, C220-M3S, ESXi 5.1.0 & 5.5.0]. ESXi 6.0 supported with IOS-XE 16.3.1 or later
• SIP Trunking and CUBE Overview
• CUBE Architecture (Physical & Virtual)
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE (Call Recording, Multi-Tenancy, HA)
• CUBE Serviceability
• Futures & Key Takeaways
Agenda
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
SIP Trunk Pointing to CUBE
Step 1: Configure CUCM to route calls to the edge SBC
Standby
IP PSTNA
TDM PBX
SRST
CME
MPLS
Enterprise Branch Offices
Enterprise
CampusCUBE with High
Availability
Active
CUBE
CUBE
PSTN is now used only for emergency calls over FXO lines
• Configure CUCM to route all PSTN calls (central and branch) to CUBE via a SIP trunk
• Make sure all different patterns of calls – local, long distance, international, emergency, informational etc.. are pointing to CUBE
BRKCOL-2125 21
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Step 2: Get details from SIP Trunk provider
Item SIP Trunk service provider requirementSample
Response
1 SIP Trunk IP Address (Destination IP Address for INVITES) 20.1.1.2 or DNS
2 SIP Trunk Port number (Destination port number for INVITES) 5060
3 SIP Trunk Transport Layer (UDP or TCP) UDP
4 Codecs supported G711, G729
5 Fax protocol support T.38
6 DTMF signaling mechanism RFC2833
7 Does the provider require SDP information in initial INVITE (Early offer required) Yes
8 SBC’s external IP address that is required for the SP to accept/authenticate calls
(Source IP Address for INVITES)
20.1.1.1
9 Does SP require SIP Trunk registration for each DID? If yes, what is the
username & password
No
10 Does SP require Digest Authentication? If yes, what is the username & password No
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Step 3: Enable CUBE Application on Cisco routers
voice service voip
mode border-element license capacity 20 License count entered here not enforced though this CLI is
required to see “show cube” CLI output
allow-connections sip to sip By default IOS/IOS-XE voice devices do not allow an incoming
VoIP leg to go out as VoIP
2. Configure any other global settings to meet SP’s requirements
voice service voipmedia bulk-stats To increment Rx/Tx counters on IOS-XE based platforms. W/O this CLI, it will show 0/0
sipearly-offer forced
header-passing error-passthru
3. Create a trusted list of IP addresses to prevent toll-fraud
voice service voipip address trusted list Applications initiating signaling towards CUBE, e.g. CUCM, CVP,
ipv4 66.77.37.2 ! ITSP SIP Trunk Service Provider’s SBC. IP Addresses from dial-peers with “session target
ipv4 10.10.1.20/28 ! CUCM ip” or Server Group are trusted by default and need not be populated here
sipsilent-discard untrusted Default configuration starting XE 3.10.1 /15.3(3)M1 to mitigate TDoS Attack
1. Enable CUBE Application
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Step 4: Configure Call routing on CUBE
IP PSTNA
TDM PBX
SRST
CME
MPLS
Enterprise Branch Offices
Enterprise Campus
CUBE with High
Availability
Active
Standby
CUBE
CUBE
PSTN is now used only for emergency calls over FXO lines
WAN Dial-PeersLAN Dial-Peers
BRKCOL-2125 24
• Dial-Peer – “static routing” table mapping phone numbers to interfaces or IP addresses
• LAN Dial-Peers – Dial-peers that are facing towards the IP PBX for sending and
receiving calls to & from the PBX
• WAN Dial-Peers – Dial-peers that are facing towards the SIP Trunk provider for
sending & receiving calls to & from the provider
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
WAN Dial-Peer Configuration
dial-peer voice 100 voip
description *** Inbound WAN side dial-peer ***
incoming called-number 408527….$
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
codec g711ulaw
dtmf-relay rtp-nte
no vad
Outbound Dial-Peer for call legs from CUBE to SP
dial-peer voice 200 voip
description *** Outbound WAN side dial-peer ***
translation-profile outgoing Digitstrip
destination-pattern 91[2-9]..[2-9]......$
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
session target ipv4:<SIP_Trunk_IP_Address>
codec g711ulaw
dtmf-relay rtp-nte
no vad
Specific to your DID range
assigned by the SP
Dial-peer for making long distance
calls to SP, based on NANP (North
American Numbering Plan)
Note: Separate outgoing DP to be created for Local, International,
Emergency, Informational calls etc.
Inbound Dial-Peer for call legs from SP to CUBE
Apply bind to all dial-peers when
CUBE has multiple interfaces.
Gig0/1 faces SP.
Translation rule/profile to strip the
access code (9) before delivering
the call to the SP
BRKCOL-2125 25
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
LAN Dial-Peer Configuration
dial-peer voice 300 voip
description *** Inbound LAN side dial-peer ***
incoming called-number 9T
session protocol sipv2
voice-class sip bind control source gig0/0
voice-class sip bind media source gig0/0
codec g711ulaw
dtmf-relay rtp-nte
no vad
Outbound Dial-Peer for call legs from CUBE to CUCM
dial-peer voice 400 voip
description *** Outbound LAN side dial-peer ***
destination-pattern 408527….$
session protocol sipv2
voice-class sip bind control source gig0/0
voice-class sip bind media source gig0/0
session target ipv4:<CUCM_IP_Address>
codec g711ulaw
dtmf-relay rtp-nte
no vad
CUCM sending 9 (access code) + All
digits dialed
SP will be sending 10 digits (NANP)
based on your DID that is being
delivered to CUCM
Inbound Dial-Peer for call legs from CUCM to CUBE
Apply bind to all dial-peers when
CUBE has multiple interfaces. Gig0/0
faces CUCM.
Default codec is G729 if none is
specified
Note: If more than 1 CUCM cluster exists, you will have to create multiple such LAN dial-peers with “preference CLI” for CUCM redundancy/load balancing as the
traditional way to accommodate multiple trunks
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Step 5: SIP Normalization
SIP incompatibilities arise due to:
• A device rejecting an unknown header (value or parameter) instead of ignoring it
• A device expecting an optional header value/parameter or can be implemented in multiple ways
• A device sending a value/parameter that must be changed or suppressed (“normalized”) before it leaves/enters the enterprise to comply with policies
• Variations in the SIP standards of how to achieve certain functions
• With CUBE 10.0.1 SIP Profiles can be applied to inbound SIP messages as well
More information at http://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/118825-technote-sip-00.html
Incoming Outgoing
INVITE
sip:[email protected]:5060
user=phone SIP/2.0
INVITE
sip:[email protected]:5060
SIP/2.0
voice class sip-profiles 100
request INVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
request REINVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
Add user=phone for INVITEs
Modify a “sip:” URI to a “tel:” URI in INVITEs
Incoming Outgoing
INVITE
tel:2222000020
SIP/2.0
INVITE
sip:[email protected]:5060
SIP/2.0
voice class sip-profiles 100
request INVITE sip-header SIP-Req-URI modify "sip:(.*)@[^ ]+" "tel:\1"
request INVITE sip-header From modify "<sip:(.*)@.*>" "<tel:\1>"
request INVITE sip-header To modify "<sip:(.*)@.*>" "<tel:\1>"
CUBE
CUBE
SIP profiles is a mechanism to normalize or customize SIP at the network border to provide interop between incompatible devices
BRKCOL-2125 27
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Normalize Inbound SIP Message (Example 2)
voice class sip-profiles 700
request INVITE sip-header Diversion modify “sip:” sip:1234@
dial-peer voice 4000 voip
description Incoming/outgoing SP
voice-class sip profiles 700 inbound
Received:
INVITE sip:[email protected]:5060 SIP/2.0
………
User-Agent: SP-SBC
………
Diversion: <sip:9.44.44.4>;privacy=off;
reason=unconditional;screen=yes
……...
m=audio 6001 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
……...
Configure Inbound
SIP Profile to add a
dummy user part
Apply to incoming
Dial-peer
Received:
INVITE sip:[email protected]:5060 SIP/2.0
……….
User-Agent: SP-SBC
……….
Diversion: <sip:[email protected]>;
privacy=off;reason=unconditional;screen=yes
……….
m=audio 32278 RTP/AVP 18 8 101
a=rtpmap:0 PCMU/8000
………..
CUBE
RequirementSIP Diversion header must include a user portion
SIP INVITE received by CUBE SIP INVITE CUBE expects
Enable Inbound
SIP Profile feature
voice service voip
sip
sip-profiles inbound
For Your
Reference
SIP Profile Rule Tagging
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
SIP Profile – Feature Overview Existing Implementation
1. Insertion
New rules are always inserted at the end, there was no way to insert a rule at the beginning or in between existing rules.
Only way to achieve this is by removing the complete profile and configuring it again in the desired order.
2. Deletion
While deleting a rule User has to give complete no form of that rule.
If there are duplicate rules, always 1st one is deleted.
3. Modification
There is no direct way to modify an existing rule. User has to delete and reconfigure the profile.
4. Duplication
If the same profile/rules applied more than once, then the rules are be duplicated
BRKCOL-2125 30
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
SIP Profile Tagging EnhancementNew rule tagging mechanism is being introduced
1. Insertion :
New rules can be inserted at any position i.e at the beginning, at the end or in between existing rules by specifying rule tag number.
2. Deletion :
Rules can be deleted by giving no form of the rule with just the tag number.
3. Modification :
Any of the existing rules can be modified by specifying the rule tag number.
4. Duplication :
When a rule with an existing tag number is applied again, the rule will be over-written, without creating any duplicate rules.
BRKCOL-2125 31
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
A mechanism to automatically upgrade the legacy SIP Profile configurations to the new rule format has been provided. The following exec CLI is being provided to upgrade existing implementation
voice sip sip-profiles upgrade
A mechanism to automatically downgrade the SIP Profile configurations with the rule tags to non-rule format has been provided. The following exe CLI has been provided for this purpose
voice sip sip-profiles downgrade
Note: When SIP Profiles are configured in “rule <tag>” format and the IOS version is migrated to a version which does not have this capability, then all the SIP Profile configurations will be lost. Hence, it is advisable to execute voice sip sip-profiles downgrade before IOS version migration.
SIP Profile Tagging Enhancement – Cont’d
BRKCOL-2125 32
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
• For tagging the rules, an additional option of “rule <tag>” has been provided
SIP Profile Tagging – Configuration
CUBE(config)#voice class sip-profiles 1
CUBE(config-class)#?
VOICECLASS configuration commands:
exit Exit from voice class configuration mode
help Description of the interactive help system
no Negate a command or set its defaults
request sip request
response sip response
rule Specify the rule
CUBE(config-class)#rule ?
<1-1073741823> Specify the rule tag
before The rule to be inserted before
CUBE(config-class)#rule 1 ?
request sip request
response sip response
The new
keyword “rule”
“tag” to be
provided with
rule keyword
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
• For inserting a rule between two rules, “before” option has been provided
SIP Profile Tagging – Configuration Cont’d
CUBE(config)#voice class sip-profiles 1
CUBE(config-class)#rule before ?
<1-1073741823> Specify the rule tag
CUBE(config-class)#rule before 3 ?
request sip request
response sip response
• If rule <tag> option is used to configure a SIP Profile rule, then this rule can be deleted by specifying just the tag number instead of specifying the entire rule configuration.
CUBE(config)#voice class sip-profiles 1
CUBE(config-class)#no rule before <tag>
For inserting a rule
between two rules, the new before keyword
is being introduced
BRKCOL-2125 34
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Configuration Example
• For tagging the rules:
voice class sip-profiles 1
rule 1 request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”
rule 2 request INVITE sip-header Supported Add “Supported: ”
• For inserting a rule between two rules using “before” option:
rule before 2 request INVITE sip-header To Modify “(.*)” “\1;temp=abc”
voice class sip-profiles 1
rule 1 request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”
rule 2 request INVITE sip-header To Modify “(.*)” “\1;temp=abc”
rule 3 request INVITE sip-header Supported Add “Supported: ”
before
option
The new rule has
been inserted
between #1 and
#3
BRKCOL-2125 35
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
• Auto-Upgrade : Exec command - “voice sip sip-profiles upgrade”
• Suppose we have the following rules configured:
request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”
request INVITE sip-header Supported Add “Supported: ”
request REGISTER sip-header Contact Modify “(.*)” “\1;temp=abc”
• After auto upgrade, the rules will be automatically upgraded as follows:
rule 1 request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”
rule 2 request INVITE sip-header Supported Add “Supported: ”
rule 3 request REGISTER sip-header Contact Modify “(.*)” “\1;temp=abc”
Configuration Example continued….
BRKCOL-2125 36
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
• Auto-Downgrade : Exec command - “voice sip sip-profiles downgrade”
• Suppose we have the following rules configured:
rule 1 request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”
rule 2 request INVITE sip-header Supported Add “Supported: ”
rule 3 request REGISTER sip-header Contact Modify “(.*)” “\1;temp=abc”
• After auto downgrade, the rules will be automatically downgraded as follows:
request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”
request INVITE sip-header Supported Add “Supported: ”
request REGISTER sip-header Contact Modify “(.*)” “\1;temp=abc”
Configuration Example continued….
BRKCOL-2125 37
SIP Profile Support for Non-Standard Headers
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
SIP Profile support for Non-Standard Headers Introducing support for adding/copying/removing/modifying non-
standard SIP headers using SIP profiles
A new 'WORD' option has been added to the SIP Profiles CLI chain to allow the user to configure any non-standard SIP HeaderCUBE(config)#voice class sip-profiles 1
CUBE(config-class)#request INVITE sip-header ?
Accept-Contact SIP header Accept-Contact
…….
Via SIP header Via
WORD Any other SIP header name
WWW-Authenticate SIP header WWW-Authenticate
CUBE(config-class)#request INVITE sip-header WORD ?
ADD addition of the header
COPY Copy a header
MODIFY Modification of a header
REMOVE Removal of a header
CUBE(config-class)#request INVITE sip-header WORD ADD “MyCustomHeader : Hussain Ali”
The new “WORD”
option for specifying
unsupported headers
• SIP Trunking and CUBE Overview
• CUBE Architecture (Physical & Virtual)
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE (Call Recording, Multi-Tenancy, HA)
• CUBE Serviceability
• Futures & Key Takeaways
Agenda
CUBE Dial-PeersCall Routing
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Understanding Dial-Peer matching Techniques:LAN & WAN Dial-Peers
• LAN Dial-Peers – Dial-peers that are facing towards the IP PBX for sending and receiving calls to & from the PBX
• WAN Dial-Peers – Dial-peers that are facing towards the SIP Trunk provider for sending & receiving calls to & from the provider
CUCM SIP Trunk ITSP SIP Trunk
CUBE
A
Outbound Calls Outbound WAN Dial-PeerInbound LAN Dial-Peer
IP PSTN
Inbound WAN Dial-PeerOutbound LAN Dial-PeerInbound Calls
BRKCOL-2125 42
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Understanding Inbound Dial-Peer Matching Techniques
CUCM SIP Trunk SP SIP Trunk
CUBE
A
Inbound LAN Dial-Peer
IP
PSTN
Inbound WAN Dial-PeerInbound Calls
Outbound Calls
Match based on
Called Number
Match based on
Calling number
1Match Based on URI
of an incoming
INVITE message
Default Dial-Peer = 0
Exact Pattern
match
Host Name/IP
Address
User portion of
URI
Phone-number of
tel-uri
Received:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:[email protected]";;branch=z9hG4bK-23955-1-0
From: "555" <sip:[email protected]:5060>;tag=1
To: ABC <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: sip:[email protected]:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
2
3
4
Priority
BRKCOL-2125 43
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Understanding Outbound Dial-Peer Matching Techniques
Match based on
Called Number &
carrier-id target
Match Based on URI
of incoming INVITE
message & carrier-id
target
Received:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:[email protected]";;branch=z9hG4bK-23955-1-0
From: "555" <sip:[email protected]:5060>;tag=1
To: ABC <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: sip:[email protected]:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
Match based on URI
of an incoming
INVITE message
Match based on
Called number
CUCM SIP Trunk SP SIP Trunk
CUBE
A
Outbound LAN Dial-Peer
IP
PSTN
Outbound WAN Dial-Peer
Inbound Calls
Outbound Calls
Exact Pattern
match
Host Name/IP
Address
User portion of
URI
Phone-number of
tel-uri
1
2
3
4
Exact Pattern
match
Host Name/IP
Address
User portion of
URI
Phone-number of
tel-uri
Priority
CSCua14749 – Carrier-id CLI not working on XE based
platforms
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Destination Server Group• Supports multiple destinations (session targets) be defined in a group and applied to
a single outbound dial-peer
• Once an outbound dial-peer is selected to route an outgoing call, multiple destinations within a server group will be sorted in either round robin or preference [default] order
• This reduces the need to configure multiple dial-peers with the same capabilities but different destinations. E.g. Multiple subscribers in a cluster
voice class server-group 1
hunt-scheme {preference | round-robin}
ipv4 1.1.1.1 preference 5
ipv4 2.2.2.2
ipv4 3.3.3.3 port 3333 preference 3
ipv6 2010:AB8:0:2::1 port 2323 preference 3
ipv6 2010:AB8:0:2::2 port 2222
* DNS target not supported in server group
dial-peer voice 100 voip
description Outbound DP
destination-pattern 1234
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
session server-group 1
BRKCOL-2125 45
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Multiple Incoming Patterns Under Same Incoming/Outgoing Dial-peer
SIP Trunk SP SIP Trunk
CUBE
IP PSTNA
(408)100-1010
(510)100-1010
(919)200-2010
(919)200-2000
(510)100-1000
(408)100-1000
voice class e164-pattern-map 300
e164 919200200.
e164 510100100.
e164 408100100.
dial-peer voice 1 voip
description Inbound DP via Calling
incoming calling e164-pattern-map 300
codec g729r8
voice class e164-pattern-map 400
url flash:e164-pattern-map.cfg
dial-peer voice 2 voip
description Outbound DP via Called
destination e164-pattern-map 400
codec g711ulaw
! This is an example of the contents of E164 patterns text file stored in flash:e164-pattern-map.cfg
919200201051010010104081001010
Site A
Site B
Site C
Site A
Site B
Site C
G729 Sites
G711 Sites
Provides the ability to combine multiple
incoming called OR calling numbers
on a single inbound voip dial-peer,
reducing the total number of inbound
voip dial-peers required with the same
routing capability
Up to 5000 entries in a text file
BRKCOL-2125 46
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Destination Dial-peer Group
• Allows grouping of outbound dial-peers based on an incoming dial-peer, reducing existing outbound dial-peer provisioning requirements
• Eliminates the need to configure extra outbound dial-peers that are sometimes needed as workarounds to achieve desired call routing outcome
• Multiple outbound dial-peers are saved under a new “voice class dpg <tag>”. The new “destination dpg <tag>” command line of an inbound voip dial-peer
can be used to reference the new dpg (dial-peer group)
• Once an incoming voip call is handled by an inbound voip dial-peer with an active dpg, dial-peers of a dpg will then be used as outbound dial-peers for an incoming call
• The order of outgoing call setups will be the sorted list of dial-peers from a dpg, i.e, the destination-patterns of the outgoing dial-peers is not relevant for selection
BRKCOL-2125 47
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Destination Dial-peer Group Configuration
voice class dpg 10000
description Voice Class DPG for SJ
dial-peer 1001 preference 1
dial-peer 1002 preference 2
dial-peer 1003
!
dial-peer voice 100 voip
description Inbound DP
incoming called-number 1341
destination dpg 10000
dial-peer voice 1001 voip
destination-pattern 8888
session protocol sipv2
session target ipv4:10.1.1.1
!
dial-peer voice 1002 voip
destination-pattern 8888
session protocol sipv2
session target ipv4:10.1.1.2
!
dial-peer voice 1003 voip
destination-pattern 8888
session protocol sipv2
session target ipv4:10.1.1.31. Incoming Dial-peer is first
matched 2. Now the DPG associated with
the INBOUND DP is selected
BRKCOL-2125 48
External/PSTN Call Recording
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
External/PSTN Call Recording Options• CUBE Controlled (Dial-peer based ORA)
• Based on Open Recording Architecture, metadata sent in Cisco Proprietary format from CUBE to Recorder
• Dial-peer controlled, IP-PBX independent
• Source of recorded media (RTP only) is always CUBE (External calls only). For SRTP-RTP calls, apply media forking CLI on the RTP leg only.
• Records both audio and video calls and supported with CUBE HA (Inbox or box-2-box)
• CUBE Controlled (Dial-peer based SIPREC)
• Based on SIPREC (RFC 6341, 7245, Metadata-draft-17, Protocol-draft-15), CUBE sends metadata in XML format
• Dial-peer controlled, IP-PBX independent
• Source of recorded media (RTP only) is always CUBE (External calls only). For SRTP-RTP calls, apply media forking CLI on the RTP leg only.
• Records both audio and video calls and supported with CUBE HA (Inbox or box-2-box)
• CUCM NBR (Network Based Recording)
• CUCM Controlled, requires CUCM 10+ and UC Services API be enabled on CUBE
• Recording triggered by CUCM and this mode records only Audio calls
• Source of Recorded Media can be CUBE or Endpoint (BiB), CUBE as source desired for PSTN calls
BRKCOL-2125 50
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
CUBE Controlled Recording Option – Media Forking
• Call agent independent
• Configured on a per Dial-peer level to fork RTP
Cisco MediaSense
(authentication disabled w/o UCM)
Cisco Search/Play demo app or Partner
Application
media class 9
recorder parameter
media-recording 950
dial-peer voice 901 voip
description dial-peer that needs to be forked
session protocol sipv2
media-class 9
dial-peer voice 950 voip
description dial-peer pointing to MediaSense
destination-pattern 9999 ! Dummy
session protocol sipv2
session transport tcp
session target ipv4:<Mediasense_IP>
! Bind on this DP mandatory
• CUBE sets up a stateful SIP session
with MediaSense server
• After SIP dialog established, CUBE
forks the RTP and sends it for
MediaSense to record
• With XE 3.10.1, Video calls supported
and CUBE HA for audio calls
SIPSIP
SIP
A
SP SIP
CUBERTP RTP
MediaSense
Needs to
match
Dial-peer based – Open Recording Architecture (ORA)
Cisco Proprietary Metadata
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Audio only Media Forking for an Audio/Video Call
media profile recorder 100
media-type audiomedia-recording 950
dial-peer voice 1 voip
description dial-peer that needs to be forked
session protocol sipv2
media-class 1
• MediaSense 10+ or any recording server can decline the video stream and choose to have only the audio
stream recorded by setting the video port as 0 in the SDP answer
• CUBE can be configured to offer only audio streams to be recorded even if the call that is being recorded
is an audio/video call
SIPSIPA
SP SIP
CUBERTP RTP
MediaSense
CUBE Controlled Recording – Dial-peer based
media-class 1
recorder profile 100
• Support for forwarding any 3rd
party IP PBX GUID to the
recording server by use of SIP
ProfilesSIP
Cisco Proprietary Metadata
dial-peer voice 950 voip
description dial-peer pointing to MediaSense
destination-pattern 9999 ! Dummy
session protocol sipv2
session transport tcp
session target ipv4:<Mediasense_IP>
! Bind on this DP mandatory
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
CUBE Controlled Recording Option - SIPREC
• SIP Profiles can additionally be used to forward 3rd party IP PBX Call Identifier to the Recorder for Correlation
SIPREC Compliant Recorder
• SIP is used as a protocol between CUBE and the recording server, where CUBE acts as the recording
client and any third party recorder acts as the recording server
• Along with SDP, metadata information is passed by CUBE to the recording server in XML format
SIPSIPA
SP SIP
CUBERTP RTP
Recorder
Dial-peer based – SIPREC Standard
XML Metadata
• Metadata includes the
communication session details of
audio or video calls and also
identifies the participants of the call
media class 9
recorder parameter siprec
media-recording 950
dial-peer voice 901 voip
description dial-peer that needs to be forked
session protocol sipv2
media-class 9
dial-peer voice 950 voip
description dial-peer pointing to MediaSense
destination-pattern 9999 ! Dummy
session protocol sipv2
session transport tcp
session target ipv4:<Mediasense_IP>
! Bind on this DP mandatory
Needs to
match
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
CUCM (10.X or later) Controlled Recording
1. Enable HTTP on IOSip http server
http client persistent
2. Enable the API on IOSuc wsapi
source-address [IP_Address_of_CUBE]
3. Enable XMF service within the APIprovider xmf
remote-url 1 http://CUCM:8090/ucm_xmf
no shutdown
Gateway/CUBE Recording
Enabled
1. 2.
3.
4.
5.
[1] – [3]: An external call is answered by user with IP phone
[4] – [5]: CUCM sends forking request over HTTP to CUBE, which
sends two media streams towards the Recording Server
UC Services API – Network Based Recording
• Recording not preserved on failover in CUBE HA• Selective Recording, Mobile/SNR/MVA Calls• Recording Call Preservation
Now Supports Inbound CVP (Survivability.tcl) Call Recording [IOS 15.6(1)T, IOS-XE 3.17]
BRKCOL-2125 54
Multiple Non-Authenticated SIP Trunks on a CUBE
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Non-Authenticated SIP Trunking to more than one Service Provider
A
TDM PBX
SRST
CME
MPLS
Enterprise
Branch Offices
Enterprise
Campus
Active
CUBE
SIP SP-1(10.10.10.2)
SIP SP-2(20.20.20.2)
Large enterprises are deploying more
than one SIP Trunk provider for:
• Alternate call routing
• Load balancing
dial-peer voice 20 voip
description “Secondary path to SIP SP-2”
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target ipv4:20.20.20.2
preference 2
voice-class sip options-keepalivevoice-class sip bind control source-interface loopback2
voice-class sip bind media source-interface loopback2
interface loopback1
ip address 10.10.10.1 255.255.255.0
interface loopback2
ip address 20.20.20.1 255.255.255.0
dial-peer voice 10 voip
description “Primary path to SIP SP-1”
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target ipv4:10.10.10.2
voice-class sip options-keepalivevoice-class sip bind control source-interface loopback1
voice-class sip bind media source-interface loopback1
SIP SP-1’s
network
SIP SP-2’s
network
NOTE: Dual SPs can be used for outbound calls, but to be utilized for inbound calls, arrangements between SPs required
BRKCOL-2125 56
Multiple Authenticated/Registered SIP Trunks on a CUBE
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Multiple Instances of SIP-UA on a CUBE
Existing Implementation, prior to IOS 15.6(2)T and IOS-XE 16.3.1
• CUBE Configuration generally consists of • Global – Everything under voice service voip
• Call Routing – Dial-peers (Any configuration under dial-peers always overrides Global config)
• SIP User Agent Config – Everything under sip-ua, applicable globally on the platform
• No provision to configure specific bind/credentials/outbound proxy for different registrar
• No provision to configure specific configs (e.g. timers, retry) for different tenants
• Unable to handle authentication challenge for more than one trunk that have the same SIP realm
BRKCOL-2125 58
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Introducing Tenants on CUBE
• Every Registrar/User Agent/ITSP connected to CUBE can be considered a Tenant to CUBE
• Allows specific global configurations (CLI under sip-ua) for multiple tenants such as specific SIP Bind for REGISTER messages
• Allows differentiated services for different tenantsBRKCOL-2125 59
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
“Voice class Tenant” Overview
Prior to Multi Tenancysip-uaregistrar 1 ipv4:60.60.60.60:9051 expires 3600registrar 2 ipv4:70.70.70.70:9052 expires 3600credentials username aaaa password 7 06070E204D realm aaaa.comcredentials username bbbb password 7 110B1B0715 realm bbbb.com
voice service voip outbound-proxy ipv4:10.64.86.35:9057bind control source-interface GigabitEthernet0/1
With Voice Class Tenant (Multi-Tenancy)voice class tenant 1registrar 1 ipv4:60.60.60.60:9051 expires 3600credentials username aaaa password 7 06070E204D realm aaaa.comoutbound-proxy ipv4:10.64.86.35:9057bind control source-interface GigabitEthernet0/0
voice class tenant 2registrar 1 ipv4:70.70.70.70:9052 expires 3600credentials username bbbb password 7 110B1B0715 realm bbbb.comoutbound-proxy ipv4:10.64.86.40:9040bind control source-interface GigabitEthernet0/1
E164 - aaaa
E164 - bbbb
Registrar - 1
Registrar - 2
E164 - aaaa
E164 - bbbb
Registrar - 1
Registrar - 1
OB Proxy 1 & Bind-1
OB Proxy 2 & Bind-2
• Most configs under “sip-ua” and “voice service voip” added in “voice class tenant <tag>”, e.g. Registrar and Credentials CLI under tenant using different bind and outbound proxy
Global OB Proxy and Bind
BRKCOL-2125 60
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Authenticating Multiple trunks with same Realm
With Voice Class Tenant (Multi-Tenancy)voice class tenant 1registrar 1 dns:cisco.com expires 3600credentials number +1234 username aaaa@cisco password 0 AAAA realm cisco.comauthentication username aaaa@cisco password 7 AAAA realm cisco.com
voice class tenant 2registrar 1 dns:cisco.com expires 3600credentials number +6789 username bbbb@cisco password 0 BBBB realm cisco.comauthentication username bbbb@cisco password 7 BBBB realm cisco.com
• Requirement : To register two different authenticated numbers/usernames to different registrars, but with the same realm
• Prior to IOS 15.6(2)T / IOS-XE 16.3.1, CUBE could register multiple trunks only with different realms as the “authentication” command only accepted different realms. If the realms were the same, it just overwrote the username and password
• Now each credential/authentication pair can be defined under its own voice class tenant so that the same realm can be used for authentication
BRKCOL-2125 61
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Configuring Voice Class Tenant• Configure voice class tenantvoice class tenant 1
registrar 1 ipv4:10.64.86.35:9052 expires 3600
credentials username aaaa password 7 06070E204D realm aaaa.com
credentials number bbbb username bbbb password 7 110B1B0715 realm bbbb.com
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
copy-list 1
outbound-proxy ipv4:10.64.86.35:9055
early-offer forced
• Apply tenant to the desired dial-peer
dial-peer voice 1 voip
destination-pattern 111
session protocol sipv2
session target ipv4:10.64.86.35:9051
session transport udp
voice-class sip tenant 1 Apply Tenant to a Dial-peer
Add new voice class tenant
BRKCOL-2125 62
Multi-VRF Aware CUBE
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Introduction to Multi-VRF
ATT
VZN
SPT
• Virtual Routing and Forwarding (VRF) is an IP technology that allows for multiple instances of a routing table to coexist on the same router at the same time as opposed to a single global route table, allowing for multiple virtual networks within a single network entity to isolate between media and data virtual networks
• Multi-VRF allows for the use of only one router to accomplish the tasks that multiple routers usually perform
• Prior to IOS 15.6(2)T / IOS-XE 16.3.1, CUBE only supports a single VRF for Voice [voice vrf vrfname]
BRKCOL-2125 64
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Multi-VRF and CUBE Enterprise
BRKCOL-2125 65
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Multi-VRF Aware Call Routing on CUBE
• CUBE allows intra and inter VRF routing of voice and video calls without the need of Route Leaks improving security at the network level
• Overlapped IP addressing and Dial Plan with Multi VRF feature provides seamless integration of networks
• Show command outputs enhanced to display the VRF ID’s for active voice and video calls
• Provision to configure RTP port ranges for each VRF and allocation of Local RTP ports based upon VRF. Listen sockets on UDP, TCP and TLS transports based on the VRF
BRKCOL-2125 66
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Multi-VRF Design Considerations
• It is strongly recommended to deploy CUBE 11.5.2 or later [IOS 15.6(3)M, IOS-XE 16.3.1] for Multi-VRF aware call routing as inbound dial-peers are filtered based on the incoming VRF FIRST and then followed by the regular inbound dial-peer matching. This ensures no potential routing issues will exist for incoming INVITES or any out-of-dialog messages such as REGISTER, OPTIONS, NOTIFY, etc
• Dial-peer bind statements are mandatory as the VRF association to a dial-peer is based upon the interface sip bind and both Control and Media on a dial-peer has to bind with the same VRF
• Whenever global sip bind interface associated with a VRF is added/modified/removed, user should restart the sip services under “voice service voip sip call service stop/no call service stop”
• Default incoming dial-peer (dial-peer 0) match is not supported with VRF
BRKCOL-2125 67
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Understanding Inbound Dial-Peer Matching Techniques
CUCM SIP Trunk SP SIP Trunk
CUBE
A
Inbound LAN Dial-Peer
IP
PSTN
Inbound WAN Dial-PeerInbound Calls
Outbound Calls
Match based on
Called Number
Match based on
Calling number
1Match Based on URI
of an incoming
INVITE message
Default Dial-Peer = 0
Exact Pattern
match
Host Name/IP
Address
User portion of
URI
Phone-number of
tel-uri
Received:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:[email protected]";;branch=z9hG4bK-23955-1-0
From: "555" <sip:[email protected]:5060>;tag=1
To: ABC <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: sip:[email protected]:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
2
3
4
Priority
0
Filter dial-peers based
on incoming VRF if
configured and then 1
to 3 below
BRKCOL-2125 68
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Multi-VRF Design Considerations – Cont’d
• Whenever destination server group is used with VRF, ensure that the server group should have the candidates (i.e. session targets) belonging to the same network as that of sip bind on the dial-peer where the server-group is configured. Sample Configuration in notes section below
• Dial-peer group feature or COR (Class of Restriction) lists can be used to restrict call routing to the same or group of VRFs (e.g. Overlapping Dial plans)
• The DSP resources are a global pool and not reserved on a per VRF basis. It is used on a first come first serve basis
BRKCOL-2125 69
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Multi-VRF Feature Restrictions
• CUBE + CME co-located with VRF and TDM-SIP gateway are not supported
• IPV6 with VRF is not supported on CUBE. Only IPv4 is supported with VRF
• Multi-VRF calls across CUBE are supported in SIP-SIP flow-through mode only and not supported in flow-around mode.
• Media Anti-trombone is not supported with VRF
• Legacy global voice vrf and Multi VRF doesn’t co-exist. Customers using global voice vrf have to remove the CLI in order to use Multi VRF feature
• UC Services API (CUCM NBR Recording) is not VRF aware. Works globally for all call recordings and will not separate the call notification on a per VRF basis
• With Single/Multi VRF configured, DNS request will be at global (i.e. no vrf is associated with the DNS request)
For Your
Reference
BRKCOL-2125 70
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
CUBE Multi VRF - Basic Configuration
CUBE
VRF 1 VRF 2
ip vrf vrf1rd 1:1
interface GigabitEthernet0/0/0ip address 7.44.44.13 255.255.0.0 ip vrf forwarding vrf1
dial-peer voice 100 voipvoice-class sip bind all interface GigE0/0/0
ip vrf vrf2rd 2:2
interface GigabitEthernet0/0/1ip address 6.44.44.13 255.255.0.0 ip vrf forwarding vrf2
dial-peer voice 200 voipvoice-class sip bind all interface GigE0/0/1
1. Configure VRF
2. Apply VRF under the interface/sub-interface
3. Bind the VRF associated interface to the dial-peer (VRF association by dial-peer bind CLI)
• Up to 54 different VRFs supported in 15.6(3)M and IOS-XE 16.3.1 or later releases
Gig0/0/0 Gig0/0/1
BRKCOL-2125 71
© 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
CUBE Multi VRF – Inbound dial-peer match
ip vrf vrf1rd 1:1
!
interface GigabitEthernet0/0/0ip address 7.44.44.13 255.255.0.0 ip vrf forwarding vrf1
!
dial-peer voice 100 voipvoice-class sip bind all interface GigE0/0/0incoming called-number 2000
ip vrf vrf2rd 2:2
!
interface GigabitEthernet0/0/1ip address 6.44.44.13 255.255.0.0 ip vrf forwarding vrf2
!
dial-peer voice 200 voipvoice-class sip bind all interface GigE0/0/1incoming called-number 2000
Inbound match based on VRF where SIP INVITE received
For VRF 1, dial-peer 100 is matched
For VRF 2, dial-peer 200 is matched
CUBE
VRF 1 VRF 2INVITE
INVITE
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CUBE Multi VRF – Inter/Intra VRF Routing
ip vrf vrf1rd 1:1
interface GigabitEthernet0/0/0ip address 7.44.44.13 255.255.0.0 ip vrf forwarding vrf1
dial-peer voice 1 voipvoice-class sip bind all interface GigE0/0/0incoming called-number 3000
dial-peer voice 100 voipvoice-class sip bind all interface GigE0/0/0destination-pattern 2000session-target ipv4: 10.1.1.1
dial-peer voice 1000 voipvoice-class sip bind all interface GigE0/0/0incoming called-number 2000
ip vrf vrf2rd 2:2
interface GigabitEthernet0/0/1ip address 6.44.44.13 255.255.0.0 ip vrf forwarding vrf2
dial-peer voice 2 voipvoice-class sip bind all interface GigE0/0/1incoming called-number 2000
dial-peer voice 200 voipvoice-class sip bind all interface GigE0/0/1destination-pattern 3000session-target ipv4:10.2.2.2
dial-peer voice 2000 voipvoice-class sip bind all interface GigE0/0/1incoming called-number 3000Intra VRF
Routing
Inter VRF
Routing
CUBE
VRF 1 VRF 2
VR
F1
INVITE sip:[email protected] INVITE sip:[email protected]
VR
F1
INVITE sip:[email protected] INVITE sip:[email protected]
VR
F2
VR
F2
VR
F1 V
RF
2
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CUBE Multi VRF – Routing w/ Overlapped Dial Plan
Route Inter or
Intra VRF calls
based on
outbound
dial-peer groups
ip vrf vrf1
interface GigabitEthernet0/0/0ip address 7.44.44.13 255.255.0.0 ip vrf forwarding vrf1
voice class dpg 100dial-peer 11 preference 1
dial-peer voice 1 voipvoice-class sip bind all interface GigE0/0/0incoming called-number 2000destination dpg 100
dial-peer voice 11 voipvoice-class sip bind all interface GigE0/0/0destination-pattern 8888session-target ipv4:10.1.1.1
ip vrf vrf2
interface GigabitEthernet0/0/1ip address 6.44.44.13 255.255.0.0 ip vrf forwarding vrf2
voice class dpg 200dial-peer 22 preference 1
dial-peer voice 2 voipvoice-class sip bind all interface GigE0/0/1incoming called-number 2000destination dpg 200
dial-peer voice 22 voipvoice-class sip bind all interface GigE0/0/1destination-pattern 2000session-target ipv4:10.2.2.2
INVITE
INVITE
INVITE
INVITE
VRF1 VRF2
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CUBE Multi VRF – Call Routing w/ Overlapped IP
ip vrf vrf1
interface GigabitEthernet0/0/0ip address 7.44.44.13 255.255.0.0 ip vrf forwarding vrf1
dial-peer voice 1 voipvoice-class sip bind all interface GigE0/0/0incoming called-number 1000destination dpg 100
voice class dpg 100dial-peer 11 preference 1
dial-peer voice 11 voipvoice-class sip bind all interface GigE0/0/0destination-pattern 1000session-target ipv4:10.1.1.1
ip vrf vrf2
interface GigabitEthernet0/0/1ip address 7.44.44.13 255.255.0.0 ip vrf forwarding vrf2
dial-peer voice 2 voipvoice-class sip bind all interface GigE0/0/1incoming called-number 2000destination dpg 200
voice class dpg 200dial-peer 22 preference 1
dial-peer voice 22 voipvoice-class sip bind all interface GigE0/0/1destination-pattern 2000session-target ipv4:10.1.1.1
INVITE
INVITE
INVITE
INVITE
VRF1 VRF2
Overlapped
local IP
Overlapped
Remote IP
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CUBE Multi Tenant Configuration Example
ip vrf vrf1 rd 1:1
interface GigabitEthernet0/0/0ip address 7.44.44.13 255.255.0.0 ip vrf forwarding vrf1
voice class tenant 1registrar ipv4:10.1.1.5 expires 3600credentials username vrf1 password 7 104F081804 realm vrf1.commax-forwards 57retry invite 7timers trying 100bind all source-interface GigabitEthernet0/0/0
dial-peer voice 1 voipvoice-class sip bind all interface GigabitEthernet0/0/0incoming called-number 2000 voice class sip tenant 1
dial-peer voice 11 voipvoice-class sip bind all interface GigabitEthernet0/0/0destination-pattern 2000session-target ipv4: 10.1.1.1voice-class sip tenant 1
ip vrf vrf2rd 2:2
interface GigabitEthernet0/0/1ip address 6.44.44.13 255.255.0.0 ip vrf forwarding vrf2
voice class tenant 2registrar ipv4:10.2.2.5 expires 3600credentials username vrf1 password 7 104F081804 realm vrf2.commax-forwards 58retry invite 5timers trying 200bind all source-interface GigabitEthernet0/0/1
dial-peer voice 2 voipvoice-class sip bind all interface GigabitEthernet0/0/1incoming called-number 3000 voice class sip tenant 2
dial-peer voice 22 voipvoice-class sip bind all interface GigabitEthernet0/0/1destination-pattern 3000session-target ipv4: 10.2.2.2voice-class sip tenant 2
CUBE
VRF 1 VRF 2
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High Availability
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CUBE High Availability Options
CUSP CUSP
• L2 Box-to-Box redundancy• ISR G2/4K (Stateful failover)
• ASR 1001-X/2-X/4/6 (Stateful failover)
• Local redundancy (Both routers must be physically located on the same Ethernet LAN)
• Not supported across data centers
• Only 1 RP and 1 ESP in ASR1006
• Preserves both signaling and media
• Clustering with load balancing• All platforms
• Load balancing by
• SP call agent
• Cisco Unified SIP Proxy
• Local and geographical redundancy
• Inbox redundancy• ASR 1006, preserves signaling & media• Stateful failover
• Local redundancyASR(config)#redundancy
ASR-RP2(config-red)#mode sso
ASR-RP2(config-red)#end
Active
Virtual IP
CUBE
CUBE
Virtual IP
Standby
SIP SP
SIP SP
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CUBE HA Design Considerations on ISR-G2 for Box-to-Box Redundancy
• Anytime a platform is reloaded in a CUBE-HA relationship, it always boots up as Standby
• All signaling/media is sourced from/to the Virtual IP Address
• Multiple Traffic (SIP/RTP) interfaces (Gig0/0, Gig0/1) require preemption and interface tracking
• HSRP Group number should be unique to a pair/interface combination on the same L2
• All interfaces of the same group have to be configured with the same priority
• No media-flow around or UC Services API (CUCM NBR) support for CUBE HA
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• Lower IP Address for ALL the interfaces (Gig0/0, Gig0/1, Gig0/2) should be on the same platform, which is used as a tie breaker for the HSRP Active state
• Multiple HSRP Groups/Interfaces/sub-interfaces can be used on either LAN or WAN side
• Upto 6 multimedia lines in the SDP are checkpointed for CUBE HA
• SDP Passthru (upto 2 m-lines) calls are also checkpointed starting IOS 15.6(1)T
• TDM or SRST or VXML GW cannot be collocated with CUBE HA
• Both platforms must be connected via a physical Switch across all likewise interfaces for CUBE HA to work, i.e. Gig0/0 of CUBE-1 and CUBE-2 must terminate on the same switch and so on
CUBE HA Design Considerations on ISR-G2 for Box-to-Box Redundancy – Cont’d
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• Cannot have WAN terminated on CUBEs directly or Data HSRP on either side. Both Active/Standby must be in the same Data Center
• Both the CUBEs must be running on the same type of platform and IOS version and identical configuration. Loopback interfaces cannot be used for bind as they are always up. Sub-interfaces are supported for all interfaces. Port Channels are supported for all interfaces from IOS 15.6(3)M
• CUBE HA only checkpoints SIP/RTP Traffic. Support for Survivability.tcl preservation was added in 15.6(2)T for CVP deployments
CUBE HA Design Considerations on ISR-G2 for Box-to-Box Redundancy – Cont’d
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• CCB (courtesy callback) feature is not supported if a callback was registered with CVP and then a switchover was done on CUBE. The CCB will not work in these scenarios.
• Recommended to configure TCP session transport for the SIP trunk between CVP and CUBE
• LTI based transcoding called flows including SRTP/RTP interworking preserved starting 15.5(2)T. Requires same PVDM3 chip capacity on both active and standby in the same slot/subslot. CPA calls (prior to being transferred to the agent), SCCP based media resources, Noise Reduction, ASP, transrating calls are not checkpointed
• SRTP - RTP, SRTP - SRTP and SRTP passthru supported on ISR-G2
CUBE HA Design Considerations on ISR-G2 for Box-to-Box Redundancy – Cont’d
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• CUBE HA with HSRP is supported with VRFs configured
• Traffic interfaces (SIP/RTP) can have VRFs configured but HSRP interface [ipc zone default config –Gig0/2 above] cannot have any VRF associated with it. This means for every CUBE HA deployment where VRFs are being utilized for SIP/RTP interfaces, at least three interfaces are required. Otherwise, any of the LAN interfaces (Gig0/0 above) can be used as an HSRP interface
• VRF ID’s will be check pointed for the calls before and after switchover. VRF Configurations in both active and standby routers have to be identical. This includes VRF based rtp port range as well
• Upon failover, the previously ACTIVE CUBE goes through a reload by design, preserving signaling/media. Thus, running config should always be saved to avoid losing it due to the reload
CUBE HA Design Considerations on ISR-G2 for Box-to-Box Redundancy – Cont’d
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CUBE Configuration on ISR-G2 Box-to-Box Redundancy
CUBE-1> enable CUBE-1# configure terminal CUBE-1(config)# ip vrf LAN-VRF CUBE-1(config)# rd 1:1 CUBE-1(config)# ip vrf WAN-VRFCUBE-1(config)# rd 2:2
CUBE-2> enable
CUBE-2# configure terminal
CUBE-2(config)# ip vrf LAN-VRF
CUBE-2(config)# rd 1:1
CUBE-2(config)# ip vrf WAN-VRF
CUBE-2(config)# rd 2:2
Configure VRFs
on the platform
(if applicable)
CUBE 1 CUBE 2
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CUBE Configuration on ISR-G2 Box-to-Box Redundancy
interface GigabitEthernet0/0description “Enterprise LAN”ip vrf forwarding LAN-VRFip address 10.10.1.12 255.255.255.0standby version 2standby 1 ip 10.10.1.13standby delay minimum 30 reload 60standby 1 preemptstandby 1 track 2 decrement 10standby 1 track 3 decrement 10standby 1 priority 50
Inside interfaces :
HSRP group 1
VRF ID : LAN-VRF
(if applicable)
Interface can be
utilized as an HSRP
interface if no VRFs
are required or
configured
CUBE 1 CUBE 2
interface GigabitEthernet0/0description “Enterprise LAN”ip vrf forwarding LAN-VRFip address 10.10.1.11 255.255.255.0standby version 2standby 1 ip 10.10.1.13standby delay minimum 30 reload 60standby 1 preemptstandby 1 track 2 decrement 10standby 1 track 3 decrement 10standby 1 priority 50
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CUBE Configuration on ISR-G2 Box-to-Box Redundancy
interface GigabitEthernet0/1description “Enterprise WAN”ip vrf forwarding WAN-VRFip address 128.107.66.77 255.255.255.0standby version 2standby 10 ip 128.107.66.79standby delay minimum 30 reload 60 standby 10 preemptstandby 10 track 1 decrement 10standby 10 track 3 decrement 10standby 10 priority 50
interface GigabitEthernet0/1
description “Enterprise WAN”
ip vrf forwarding WAN-VRF
ip address 128.107.66.78 255.255.255.0standby version 2standby 10 ip 128.107.66.79standby delay minimum 30 reload 60 standby 10 preemptstandby 10 track 1 decrement 10standby 10 track 3 decrement 10standby 10 priority 50
Outside
interfaces :
HSRP group 10
VRF ID : WAN-
VRF (if
applicable)
CUBE 1 CUBE 2
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CUBE Configuration on ISR-G2 Box-to-Box Redundancy
interface GigabitEthernet0/2description “HSRP Interface”ip address 1.1.1.1 255.255.255.0standby version 2standby 100 ip 1.1.1.3standby delay minimum 30 reload 60 standby 100 preemptstandby 100 name CUBEHAstandby 100 track 1 decrement 10standby 100 track 2 decrement 10standby 100 priority 50
!track 1 interface Gig0/0 line-protocol track 2 interface Gig0/1 line-protocoltrack 3 interface Gig0/2 line-protocol
interface GigabitEthernet0/2
description “HSRP Interface”
ip address 1.1.1.2 255.255.255.0standby version 2standby 100 ip 1.1.1.3standby delay minimum 30 reload 60 standby 100 preemptstandby 100 name CUBEHAstandby 100 track 1 decrement 10standby 100 track 2 decrement 10
standby 100 priority 50
!track 1 interface Gig0/0 line-protocol track 2 interface Gig0/1 line-protocoltrack 3 interface Gig0/2 line-protocol
Configure Interface
Tracking (for line protocol
on corresponding
interfaces of the platform
HSRP interfaces :
HSRP group 100
CANNOT HAVE VRFs
associated
CUBE 2CUBE 1
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CUBE Configuration on ISR-G2 Box-to-Box Redundancy
redundancy inter-devicescheme standby CUBEHA
voice service voipmode border-elementallow-connections sip to sipredundancy
ipc zone defaultassociation 1no shutdownprotocol sctp
local-port 5000local-ip 1.1.1.1
remote-port 5000remote-ip 1.1.1.2
redundancy inter-devicescheme standby CUBEHA
voice service voipmode border-elementallow-connections sip to sipredundancy
ipc zone defaultassociation 1no shutdownprotocol sctp
local-port 5000local-ip 1.1.1.2
remote-port 5000remote-ip 1.1.1.1
Define Redundancy scheme: Creates
interdependency b/w CUBE
redundancy & HSRP
Turn on CUBE Redundancy
HSRP Interface - IPC configuration :
Allows the ACTIVE CUBE to tell the
STANDBY about the state of the calls.
CONFIG SHOULD BE APPLIED on the
LAN SIDE (to avoid SPLIT BRAIN) and a
NON-VRF associated interface
CANNOT HAVE VRFs
associated with this interface
CUBE 1 CUBE 2
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CUBE Configuration on ISR-G2 Box-to-Box Redundancy
dial-peer voice 100 voip
description TO SERVICE PROVIDER
destination-pattern 9T
session protocol sipv2
session target ipv4:y.y.y.y
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
!
dial-peer voice 200 voip
description TO CUCM
destination-pattern 555….
session protocol sipv2
session target ipv4:10.10.1.10
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
!
ip rtcp report interval 3000
!
gateway
media-inactivity-criteria all
timer receive-rtcp 5
timer receive-rtp 86400
Bind traffic destined to the outside (SP SIP trunk)
to the outside Physical interface.
This ensures that all RTP and SIP packets are
created with the virtual IP associated with the
respective physical interface.
CUBE HA does not work with loopback interfaces
as they are always up
Configuration on Active and Standby
Bind traffic destined to the inside (CUCM or IP
PBX) to the inside Physical interface.
This ensures that all RTP and SIP packets are
created with the virtual IP associated with the
respective physical interface.
Configure media inactivity feature to clean up any
calls that may not disconnect after a failover
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CUBE HA Design Considerations on ASR1K/ISR-4K/vCUBE for Box-to-Box Redundancy
• Uses Redundancy Group (RG) Infrastructure Protocol
• GE0/0/0 and GE0/0/1 are referred to as traffic (SIP/RTP) interfaces and GE0/0/2 is RG (Redundancy Group) Control/data interface
• Starting IOS-XE 16.3.1, Port channel is supported for both RG Control/data and traffic interfaces
• When configuration is applied and saved, the platform must go through a reload cycle
• RG Control/Data Interfaces (GE0/0/2) must be connected through a Switch and not a Crossover Cable
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Additional Supported options for CUBE HA
• The RG control data interfaces can be a sub interface that is part of the same port channel used for voice traffic. This will go to switch D and E
thereby eliminating the need for additional switches for RG control/data. This is provided there is sufficient bandwidth for voice + RG
data/control on the port channel (for example when using 10G)
• Multiple ITSPs or multiple trunks from the same ITSP can be terminated on the same CUBE ENT HA (ISR G2, ISR 4K, ASR 1K, vCUBE) pair
• Port Channel(s) can be used on the WAN/ITSP side as well as shown for the LAN side in the above diagram with L2 and CE router redundancy
CUBE-1
CUBE-2
CUCM
WAN Edge
PortChannel34
Gig0/0/3
PortChannel34
Gig0/0/4
Gig0/0/3
Switch A Switch B
CUBE
CUBE
Gig0/0/2.100 – ITSP 1
Gig0/0/2.100
Gig0/0/0
Gig0/0/0
Gig0/0/4
red
un
dan
cy rii 3
red
un
dan
cy rii 2 Switch C
Gig0/0/2.200
Gig0/0/2.200 – ITSP 2
Gig0/0/1
Gig0/0/1
Po
rtC
han
ne
l2P
ort
Ch
ann
el2
Switch D Switch E
red
un
dan
cy
rii 1
ITSP 1
ITSP 2
vPC
vPC
Video Suppression
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Video Suppression
• When CUBE receives video capabilities as part of SDP, it passes them across by default
• This feature adds a mechanism on CUBE to allow only audio and image (for T.38 fax) media capabilities and drop all other media capabilities like video, application m-lines etc. while routing calls to service providers
»Only supported for SIP-SIP calls not in SDP Passthru mode
A SP SIP
INVITE w/ audio, video, application
CUBE(config)#voice service voip
CUBE(conf-voi-serv)#sip
CUBE(conf-serv-sip)#audio forced
CUBE(conf-serv-sip)#dial-peer voice 100 voip
CUBE(conf-serv-sip)#description “Outgoing Dial-peer”
CUBE(config-dial-peer)#voice-class sip audio forced
SBCCUBE
INVITE w/ audio only
Video Endpoints
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• SIP Trunking and CUBE Overview
• CUBE Architecture (Physical & Virtual)
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE (Call Recording, Multi-Tenancy, HA)
• CUBE Serviceability
• Futures & Key Takeaways
Agenda
Serviceability
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Cisco Unified Border Element Basic Call Flow
1. Incoming VoIP setup message from originating endpoint
2. This matches inbound VoIP dial peer 1 for characteristics such as codec, VAD, DTMF method, protocol, etc.
3. Match the called number to outbound VoIP dial peer 2
4. Outgoing VoIP setup message
Incoming VoIP Call Outgoing VoIP Call
dial-peer voice 1 voipdestination-pattern 1000session protocol sipv2 session target ipv4:1.1.1.1codec g711ulaw
dial-peer voice 2 voipdestination-pattern 2000session protocol sipv2session target ipv4:2.2.2.2codec g711ulaw
Originating Endpoint -
1000
TerminatingEndpoint –
2000
CUBE
voice service voip
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
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CUBE
10001.1.1.1
20002.2.2.2
Incoming VoIP Call LegMatches an Incoming Dial-peer
Outgoing VoIP Call LegMatches an Outbound Dial-peer
INVITE /w SDP INVITE /w SDP
100 TRYING100 TRYING
VRF1 – 10.10.10.10 20.20.20.20 – VRF2
c= 1.1.1.1m=audio abc RTP/AVP 0
c= 20.20.20.20m=audio xxx RTP/AVP 0
180 RINGING180 RINGING
200 OK200 OK c= 2.2.2.2
m=audio uvw RTP/AVP 0c= 10.10.10.10m=audio xyz RTP/AVP 0
RTP (Audio)
ACKACK
1.1.1.1 10.10.10.10 20.20.20.20 2.2.2.2
Understanding the Call flow
BYE BYE
200 OK200 OK
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Basic Show commands for Active CallsCUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 activedur 00:00:14 tx:0/0 rx:0/0IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 activedur 00:00:14 tx:0/0 rx:0/0IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF2
Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF 1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections
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Basic Show commands for Active CallsCUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 activedur 00:00:14 tx:0/0 rx:0/0IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 activedur 00:00:14 tx:0/0 rx:0/0IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF2
Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF 1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections
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Basic Show commands for Active CallsCUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 activedur 00:00:14 tx:0/0 rx:0/0IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 activedur 00:00:14 tx:0/0 rx:0/0IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF2
Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF 1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections
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Basic Show commands for Active CallsCUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 activedur 00:00:14 tx:0/0 rx:0/0IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 activedur 00:00:14 tx:0/0 rx:0/0IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF2
Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF 1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections
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Basic Show commands for Active CallsCUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 activedur 00:00:14 tx:0/0 rx:0/0IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 activedur 00:00:14 tx:0/0 rx:0/0IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF2
Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF 1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections
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Basic Show commands for Active CallsCUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 activedur 00:00:14 tx:0/0 rx:0/0IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 activedur 00:00:14 tx:0/0 rx:0/0IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF2
Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF 1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections
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Basic Show commands for Active CallsCUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 activedur 00:00:14 tx:0/0 rx:0/0IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 activedur 00:00:14 tx:0/0 rx:0/0IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF2
Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF 1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections
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Basic Show commands for Active CallsCUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF2
Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF 1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections
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Basic Show commands for Active CallsCUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF2
Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF 1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections
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Basic Show commands for Active CallsCUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF2
Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF 1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections
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Basic Show commands for Active CallsCUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 activedur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 activedur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF2
Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF 1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections
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Basic Show commands for Active CallsCUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 activedur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 activedur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF2
Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF 1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections
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Basic Show commands for Active CallsCUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 activedur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 activedur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF2
Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF 1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections
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Basic Show commands for Active CallsCUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 activedur 00:00:14 tx:0/0 rx:0/0IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 activedur 00:00:14 tx:0/0 rx:0/0IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a VRF:VRF2
Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF
1 17 18 17474 6000 10.10.10.10 1.1.1.1 NO VRF1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 NO VRF2
Found 2 active RTP connections
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Total Number of Active Concurrent Calls
Router# show call active total-calls
Total Number of Active Calls : 10
Total Number of Active Calls
A single call can have multiple call-legs. To determine the total number of active calls from call-legs is challenging
CLI added to display the value of current number of active (connected) calls on CUBE
The table defines the relation between call-legs and number of active calls
Call Flow Call-legsConnected
call
Basic call (audio/video) 2 1
Transferred call (Refer
handling)
3 2
Transcoded call (SCCP) 4 1
Calls after rotary/hunt 2 + x 1
Forwarded calls (CUBE
handling)
3 1
Forked call (media forking) 3 2
Forked call (signaling forking) 2 1
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Avoiding Non-call-context Debug Logs
• Many times SIP debugs contain unrelated debugs that are not useful in debugging issues related to call failures
• Starting CUBE 10.0.1, non-call-context debugs will not be printed when debug ccsip is issued
• This applies to messages originating from CUBE. Non-call context INBOUND messages towards CUBE will still be printed when debug ccsip is issued.
• If a message is not part of any call, that debug will not be printed
• Affected messages: OPTIONS, REGISTER, SUBSCRIBE/NOTIFY
• To see the above OUTBOUND messages in debugs, issue the following command
debug ccsip non-call
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Debugging Made Easier
Router# debug ccsip feature < audio | cac |
config | control | dtmf | fax | line | misc |
misc-features | parse | registration | sdp-
negotiation | sdp-passthrough | sip-profiles
| sip-transport | srtp | supplementary-
services | transcoder | video >
Categorize Debugs based on Functionality
Categorization based on Functionality
1. Audio/video/sdp/control
2. Configuration /sip-transport
3. CAC
4. DTMF/FAX/Line-side
5. Registration
6. Sdp - passthrough
7. Sip-profile/SRTP/transcoder
Example: enabling DTMF and audio debugs only with default log level is considered.
CUBE#sh debugging
CCSIP SPI: SIP info debug tracing is enabled (filter is OFF)
CCSIP SPI: audio debugging for ccsip info is enabled (active)
CCSIP SPI: dtmf debugging for ccsip info is enabled (active)
May 21 17:54:53.377: //444/5FE632EB8479/SIP/Info/verbose/32/sipSPI_ipip_store_channel_info: dtmf negotiation done, storing
negotiated dtmf = 0,
May 21 17:54:53.377: //444/5FE632EB8479/SIP/Info/info/2/sipSPIUpdateCallEntry:
Call 444 set InfoType to SPEECH
DTMF(32) debug code
Audio(2) debug code
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Debugging Made Easier
CUBE# show cube debug category codes
Categorize Debugs based on Functionality
This CLI is used to collect the predefined debug features category codes , which helps in analysis of debugs manually.
|-----------------------------------------------
| show cube debug category codes values.
|-----------------------------------------------
| Indx | Debug Name | Value
|-----------------------------------------------
| 01 | SDP Debugs | 1
| 02 | Audio Debugs | 2
| 03 | Video Debugs | 4
| 04 | Fax Debugs | 8
| 05 | SRTP Debugs | 16
| 06 | DTMF Debugs | 32
| 07 | SIP Profiles Debugs | 64
| 08 | SDP Passthrough Deb | 128
| 09 | Transcoder Debugs | 256
| 10 | SIP Transport Debugs | 512
| 11 | Parse Debugs | 1024
| 12 | Config Debugs | 2048
| 13 | Control Debugs | 4096
| 14 | Mischellaneous Debugs| 8192
| 15 | Supp Service Debugs | 16384
| 16 | Misc Features Debugs| 32768
| 17 | SIP Line-side Debugs | 65536
| 18 | CAC Debugs | 131072
| 19 | Registration Debugs | 262144
|-----------------------------------------------
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Diagnostic Signature Lookup Tool (DSLT)On-demand Troubleshooting Automation
• SIP Trunking and CUBE Overview
• CUBE Architecture (Physical & Virtual)
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE (Call Recording, Multi-Tenancy, HA)
• CUBE Management & Troubleshooting
• Futures & Key Takeaways
Agenda
Introducing Cisco Business Edition 4000 (BE4K)
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BE4K – Simple, cloud-managed office communications
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Features
Packaged Collaboration Offer - Voice and Voicemail solution for SMBs (200 users)
Based on ISR4321 router running CUCME and virtualized CUE.
Support for the latest IP Phones (78xx, 88xx, Jabber, ATA-190)
Rapid Installation – Preconfigured before shipment
Smart licensing
ISR 4321
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BE4K – Contd.
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Voice & Voicemail
Cisco OEM Cloud Management.
Cloud Deployed - configuration, administration and management.
Rich Feature Set – Auto-attendant, Flexible call routing, Park, Hold, Transfer, Hunt groups, Conferencing, Voicemail.
Single or multiple sites.
Fixed configuration SKU for ease of ordering.
Migration options include BE6000 / BE7000, Cisco Spark. No throw-away capex.
Add-on Spark and WebEx for messaging and meetings.
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Key Takeaways• It is a manageable transition from existing TDM based networks to SIP
networks using these network design techniques
• Enterprise SBC (Cisco Unified Border Element - CUBE) is an essential component of a UC solution providing;
• Security, Session Management, Interworking, Demarcation
• Over 18,000 Enterprise customers all over the Globe
• Proven interoperability with 3rd party PBX vendors and different service providers around the world (more than 165 countries)
• Now is the time to deploy SIP Trunking in either a Centralized or a Distributed solution to save money, simplify your topology and setup your infrastructure for future services
• Complete feature Presentations, Lab Guide, Free Hands-on Lab access & Application Notes :
»https://cisco.box.com/cubeBRKCOL-2125 121
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Complete Your Online Session Evaluation
Don’t forget: Cisco Live sessions will be available for viewing on-demand after the event at CiscoLive.com/Online
• Please complete your Online Session Evaluations after each session
• Complete 4 Session Evaluations & the Overall Conference Evaluation (available from Thursday) to receive your Cisco Live T-shirt
• All surveys can be completed via the Cisco Live Mobile App or the Communication Stations
122BRKCOL-2125
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Cisco Spark Ask Questions, Get Answers, Continue the Experience
Use Cisco Spark to communicate with the Speaker and fellow participants after the session
Download the Cisco Spark app from iTunes or Google Play
1. Go to the Cisco Live Berlin 2017 Mobile app
2. Find this session
3. Click the Spark button under Speakers in the session description
4. Enter the room, room name = BRKCOL-2125
5. Join the conversation!
The Spark Room will be open for 2 weeks after Cisco Live
123Presentation ID
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Continue Your Education
• Demos in the Cisco campus
• Walk-in Self-Paced Labs
• Lunch & Learn
• Meet the Engineer 1:1 meetings
• Related sessions
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Thank You