Post on 23-Apr-2018
Implementation & Management of Cisco Unified Border
Element (CUBE) Enterprise Hussain Ali, CCIE# 38068 (Voice)
Technical Marketing Engineer
BRKUCC-2934
John Vickroy
Product Manager
• SIP Trunking and CUBE Overview
• CUBE Architecture (Physical & Virtual)
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE
• CUBE Management & Troubleshooting
• Futures & Key Takeaways
Agenda
CUBE Overview
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
Note: New platforms in this
release : Introducing ISR-4K
Series (IOS-XE 3.13.1+
recommended)
Introducing CUBE on
CSR
vCUBE [Performance
dependent on vCPU and
memory]
CUBE Session Capacity Summary
Platform CUBE Sessions
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
Introduced in Oct 2013
Introduced in July 2013
Introduced in May 2014
For Your
Reference
ISR G2 CUBE Ent ASR Parity
with ISR
ASR / ISR-4K*/vCUBE (CSR)*
CUBE Vers.
2900/ 3900 FCS CUBE Vers.
IOS XE Release FCS
9.0.1 15.3.1T 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
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
11.1.0 15.5(3)M July 2015 >95% 11.1.0 3.16 15.5(3)S July 2015
11.5.0 15.6(1)T Nov 2015 >95% 11.5.0 3.17 15.6(1)S Nov 2015
12.0.0 15.6(2)T Mar 2016 >95% 12.0.0 3.18 15.6(2)S Mar 2016
12.1.0 15.6(3)M July 2016 >95% 12.1.0 3.19 15.6(3)S July 2016
* IOS-XE3.13.1 or later recommended for all ISR-4K series and XE3.15 for vCUBE
CUBE Software Release Mapping
CUBE INTEROPERABILITY
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 based Verified by
CUBE ISR(G2/4K), ASR and CSR Licensing
Platform Single-Use Licenses Redundancy Licenses
( 1 SKU for Active/Standby Pair)
Cisco 881, 886, 887, 888, 892F, SPIAD FL-NANOCUBE N/A
ISR G2 (2901, 2911, 2921, 2951, 3925, 3945, 3925E, 3945E)
FL-CUBEE-5
FL-CUBEE-25
FL-CUBEE-100
FL-CUBEE-5-RED
FL-CUBEE-25-RED
FL-CUBEE-100-RED
ISR-4K (4321, 4331, 4351, 4431, 4451)
FL-CUBEE-5
FL-CUBEE-25
FL-CUBEE-100
FL-CUBEE-5-RED
FL-CUBEE-25-RED
FL-CUBEE-100-RED
Cisco ASR1001-X, 1002-X, 1004 RP2, 1006 RP2
FLASR1-CUBEE-100P
FLASR1-CUBEE-4KP
FLASR1-CUBEE-16KP
FLASR1-CUBEE-100R
FLASR1-CE-100R
FLASR1-CE-500R
FLASR1-CE-1KR
FLASR1-CUBEE-4K-R
FLASR1-CE-4KR
FLASR1-CUBEE-16KR
FLASR1-CE-16KR
vCUBE (CUBE on CSR 1000v)
APPX (No TLS/SRTP) or AX (All features) CSR licensing package
Same SKUs as ASR1K series Same SKUs as ASR1K series
http://www.cisco.com/c/en/us/products/collateral/unified-communications/unified-border-element/order_guide_c07_462222.html
For Your
Reference
• SIP Trunking and CUBE Overview
• CUBE Architecture (Physical & Virtual)
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE
• CUBE Management & Troubleshooting
• Futures & Key Takeaways
Agenda
CUBE Call Flow
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
Signaling and media terminated by the CUBE
Media bypasses the CUBE
Media Flow-Through
Signaling and media terminated by the CUBE
Transcoding and complete IP address hiding require this model
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, 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 voip destination-pattern 1000 session protocol sipv2 session target ipv4:1.1.1.10 codec g711ulaw
dial-peer voice 2 voip destination-pattern 2000 session protocol sipv2 session target ipv4:2.2.2.10 codec g711ulaw
Originating Endpoint –
1000
Terminating Endpoint –
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
CUBE
1000 1.1.1.1
2000 2.2.2.2
Incoming VoIP Call Leg Matches an Incoming Dial-peer
Outgoing VoIP Call Leg Matches an Outbound Dial-peer
INVITE /w SDP INVITE /w SDP
100 TRYING 100 TRYING
10.10.10.10 20.20.20.20
c= 1.1.1.1 m=audio abc RTP/AVP 0
c= 20.20.20.20 m=audio xxx RTP/AVP 0
180 RINGING 180 RINGING
200 OK 200 OK c= 2.2.2.2
m=audio uvw RTP/AVP 0 c= 10.10.10.10 m=audio xyz RTP/AVP 0
RTP (Audio)
ACK ACK
1.1.1.1 10.10.10.10 20.20.20.20 2.2.2.2
Understanding the Call flow
BYE BYE
200 OK 200 OK
Basic Show commands for Active Calls
CUBE# show call active voice brief 121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:2 Answer 2000 active dur 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 g729r8 TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a 121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:1 Originate 1000 active dur 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 g729r8 TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a Telephony call-legs: 0 SIP call-legs: 2 H323 call-legs: 0 Call agent controlled call-legs: 0 SCCP call-legs: 0 Multicast call-legs: 0 Total call-legs: 2 CUBE# show voip rtp connections VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 17 18 17474 6000 10.10.10.10 1.1.1.1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 Found 2 active RTP connections
CUBE Architecture ISR G2 vs ASR1K vs ISR 4K vs vCUBE (CUBE on CSR1000v)
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 IOS
Msg I/f
Control Plane
Data (Forwarding) Plane
ASR/ISR-4K Architecture
RP IOS
CPU
I/O I/O
Control
Plane
Data Plane
ISR G2 Architecture
Sig
na
ling
Sig
na
ling
Media
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 NICs Flash / Disk Memory Virtual 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)
Introducing vCUBE (CUBE on CSR 1000v) • CSR1000v is a virtual machine, running on x86 server (no specialized hardware) with
physical resources are managed by hypervisor and shared among VMs
• Can be installed either using an OVA file or deployed with an ISO image
• 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
• vMotion for vCUBE not supported today
• vCUBE Tested Reference Configurations [UCS base-M2-C460, C220-M3S, ESXi 5.1.0 & 5.5.0]
ASR, CSR & ISR-G2/4K Feature Comparison General SBC Features ASR1K ISR-G2 4300/4400 (XE3.13.1+) vCUBE (XE3.15+)
High Availability Implementation Redundancy-Group
Infrastructure HSRP Based
Redundancy-Group
Infrastructure
Redundancy-Group
Infrastructure
TDM Trunk Failover/Co-
existence Not Available Exists Exists Not Available
Media Forking XE3.8 15.2.1T XE3.10 Exists
Software MTP registered to
CUCM (Including HA Support) XE3.6 Exists Exists Exists
DSP Card SPA-DSP PVDM2/PVDM3 PVDM4 Not Available
Transcoder registered to CUCM Not Available Exists via SCCP Exists via SCCP (XE3.11) Not Available
Transcoder Implementation Local Transcoder Interface
(LTI)
SCCP or
LTI (starting IOS 15.2.3T) SCCP and LTI Not Available
Embedded Packet Capture Exists Exists Exists Exists
Web-based UC API XE3.8 15.2.2T Exists Exists
Noise Reduction & ASP Exists 15.2.3T Exists Not Available
Call Progress Analysis XE3.9 15.3.2T Exists Not Available
CME/SRST and CUBE co-
existence Not Available Exists XE3.11 Roadmap
SRTP-RTP Call flows Exists (NO DSPs needed) Exists (DSPs required) Exists (NO DSPs needed) Exists (No DSPs needed)
VXML GW Not Available Exists Not Available Not Available
vCUBE Installation using OVA
vCUBE – CSR1000v Installation with OVA
• Download CSR1000v OVA from cisco.com
vCUBE – Download XE3.15 image
vCUBE – Deploy OVA
vCUBE – Installation Cont’d
vCUBE – Installation Cont’d
vCUBE – Choose Form factor
vCUBE Installation Cont’d
vCUBE – Assign LAN, WAN, and VM Network
vCUBE Installation Cont’d
vCUBE Installation – Edit Settings to add Serial Port
vCUBE Installation – Edit Settings to add Serial Port
Serial Port – Connect via Network
Serial Port – Define URL
Serial Port – Verify Settings
vCUBE Installation – Power On VM
Install process takes some time
Install process takes some time
vCUBE – Initial Configuration
• Assign IP to VM Network Interface, Gig3 below, and enable console access with “platform console serial” CLI, and set enable password
vCUBE – Initial Configuration – Telnet into Router
Initial Configuration – Copy License File to Flash:
Initial Configuration – Install License File
Initial Configuration – Verify New Throughput Level and boot CSR to the correct package
vCUBE Initial Setup – Voice CLI is now accessible
• SIP Trunking and CUBE Overview
• CUBE Architecture (Physical & Virtual)
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE
• CUBE Management & Troubleshooting
• Futures & Key Takeaways
Agenda
High-density Dedicated
Gateways
Transitioning to Centralized SIP Trunking... Re-purpose your existing Cisco voice gateway’s as Session Border Controllers
SIP/H323/MGCP
Media
TDM PBX
SRST CME
A Enterprise Campus
Enterprise
Branch Offices
MPLS
BEFORE Media
SIP Trunks
SRST
Enterprise
Campus
IP PSTN A
TDM PBX
CME
MPLS
CUBE with High
Availability
Active
Standby
CUBE
CUBE
PSTN is now
used only for
emergency
calls over FXO
lines
AFTER
46
Enterprise
Branch Offices
• Step 1 – Configure IP PBX to route all calls (HQ and branch offices) to the edge SBC
• Step 2 – Get SIP Trunk details from the provider
• Step 3 – Enable CUBE application on Cisco routers
• Step 4 – Configure call routing on CUBE (Incoming & Outgoing dial-peers)
• Step 5 – Normalize SIP messages to meet SIP Trunk provider’s requirements
• Step 6 – Execute the test plan
Steps to transitioning...
Media
SRST
Enterprise
Campus
IP PSTN A
TDM PBX
CME
MPLS
Enterprise Branch
Offices
CUBE with High
Availability
Active
Standby
CUBE
CUBE
PSTN is now
used only for
emergency
calls over FXO
lines
SIP Trunk
SIP Trunk Pointing to CUBE
Step 1: Configure CUCM to route calls to the edge SBC
Standby
IP PSTN A
TDM PBX
SRST
CME
MPLS
Enterprise Branch Offices
Enterprise
Campus CUBE 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
Also see BRKUCC-2006
Step 2: Get details from SIP Trunk provider
Item SIP Trunk service provider requirement Sample 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
Step 3: Enable CUBE Application on Cisco routers
voice service voip
mode border-element license capacity 200
allow-connections sip to sip
2. Configure any other global settings to meet SP’s requirements
voice service voip
sip
early-offer forced
header-passing
error-passthru
3. Create a trusted list of IP addresses to prevent toll-fraud
voice service voip ip address trusted list Applications initiating signaling towards CUBE, e.g. CUCM, ipv4 20.20.20.20 CVP, Service Provider’s SBC ipv4 10.1.1.50 sip silent discard-untrusted Default configuration starting XE 3.10.1 /15.3(3)M1 to mitigate TDoS Attack
1. Enable CUBE Application
Step 4: Configure Call routing on CUBE
IP PSTN A
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-Peers LAN Dial-Peers
• 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
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
Outbound Dial-Peer for call legs from CUBE to SP
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
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
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
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
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
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 www.cisco.com/go/cube > Configure > Configuration Examples and TechNotes
Incoming Outgoing
INVITE
sip:5551000@sip.com:5060
user=phone SIP/2.0
INVITE
sip:5551000@sip.com: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:2222000020@9.13.24.6:5060
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
Normalize Inbound SIP Message (Example 1)
voice class sip-profiles 400
request INVITE sip-header Diversion modify “sip:” sip:1234@
dial-peer voice 4000 voip
description Incoming/outgoing SP
voice-class sip profiles 400 inbound
Sent:
INVITE sip:2000@9.44.44.4:5060 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 Dial-peer
or Globally
Sent:
INVITE sip:2000@9.44.44.4:5060 SIP/2.0
……….
User-Agent: SP-SBC
……….
Diversion: <sip:1234@abc.com>;
privacy=off;reason=unconditional;screen=yes
……….
m=audio 32278 RTP/AVP 18 8 101
a=rtpmap:0 PCMU/8000
………..
voice service voip
sip
sip profiles 400 inbound
CUBE
Requirement SIP 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
SIP Profile – Existing Feature Overview
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
SIP Profile Tagging Enhancement
New rule tagging mechanism has been 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
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
• 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
• 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
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
• 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….
• 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….
SIP Profile Support for Non-Standard Headers
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 Header
CUBE(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
ADD addition of the header
COPY Copy a header
MODIFY Modification of a header
REMOVE Removal of a header
The new “WORD”
option for specifying
unsupported headers
CUBE-Best-Session-in-SanDiego ?
Step 6: Execute the Test Plan
• Inbound and outbound Local, Long distance, International calls for G711 & G729 codecs (if supported by provider)
• Outbound calls to information and emergency services
• Caller ID and Calling Name Presentation
• Supplementary services like Call Hold, Resume, Call Forward & Transfer
• DTMF Tests
• Fax calls – T.38 and fallback to pass-through (if option available)
• SIP Trunking and CUBE Overview
• CUBE Architecture (Physical & Virtual)
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE
• CUBE Management & Troubleshooting
• Futures & Key Takeaways
Agenda
CUBE Dial-Peers Call Routing
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 SP SIP Trunk
CUBE
A
Outbound Calls Outbound WAN Dial-Peer Inbound LAN Dial-Peer
IP PSTN
Inbound WAN Dial-Peer Outbound LAN Dial-Peer Inbound Calls
Understanding Inbound Dial-Peer Matching Techniques
CUCM SIP Trunk SP SIP Trunk
CUBE
A
Inbound LAN Dial-Peer
IP
PSTN
Inbound WAN Dial-Peer Inbound Calls
Outbound Calls
Match based on Called
Number
Match based on Calling
number
1 Match 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:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
2
3
4
Priority
Understanding Inbound Dial-Peer Matching Techniques
dial-peer voice 5 voip
incoming called-number 654321
dial-peer voice 6 voip
answer-address 555
dial-peer voice 7 voip
destination-pattern 555
voice class uri 1001 sip
host ipv4:10.1.1.1
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 1 voip
incoming uri via 1001
dial-peer voice 2 voip
incoming uri request 2001
dial-peer voice 3 voip
incoming uri to 2001
dial-peer voice 4 voip
incoming uri from 1001
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
1
2
3
4
Priority
A
B
C
D
CUCM SIP Trunk SP SIP Trunk
CUBE
A
Inbound LAN Dial-Peer
IP
PSTN
Inbound WAN Dial-Peer Inbound Calls
Outbound Calls
Understanding Inbound Dial-Peer Matching Techniques
dial-peer voice 5 voip
incoming called-number 654321
dial-peer voice 6 voip
answer-address 555
dial-peer voice 7 voip
destination-pattern 555
voice class uri 1001 sip
host ipv4:10.1.1.1
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 1 voip
incoming uri via 1001
dial-peer voice 2 voip
incoming uri request 2001
dial-peer voice 3 voip
incoming uri to 2001
dial-peer voice 4 voip
incoming uri from 1001
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
1
2
3
4
A
B
C
D
Priority CUCM SIP Trunk SP SIP Trunk
CUBE
A
Inbound LAN Dial-Peer
IP
PSTN
Inbound WAN Dial-Peer Inbound Calls
Outbound Calls
Understanding Inbound Dial-Peer Matching Techniques
dial-peer voice 5 voip
incoming called-number 654321
dial-peer voice 6 voip
answer-address 555
dial-peer voice 7 voip
destination-pattern 555
voice class uri 1001 sip
host ipv4:10.1.1.1
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 1 voip
incoming uri via 1001
dial-peer voice 2 voip
incoming uri request 2001
dial-peer voice 3 voip
incoming uri to 2001
dial-peer voice 4 voip
incoming uri from 1001
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
1
2
3
4
A
B
C
D
Priority CUCM SIP Trunk SP SIP Trunk
CUBE
A
Inbound LAN Dial-Peer
IP
PSTN
Inbound WAN Dial-Peer Inbound Calls
Outbound Calls
Understanding Inbound Dial-Peer Matching Techniques
dial-peer voice 5 voip
incoming called-number 654321
dial-peer voice 6 voip
answer-address 555
dial-peer voice 7 voip
destination-pattern 555
voice class uri 1001 sip
host ipv4:10.1.1.1
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 1 voip
incoming uri via 1001
dial-peer voice 2 voip
incoming uri request 2001
dial-peer voice 3 voip
incoming uri to 2001
dial-peer voice 4 voip
incoming uri from 1001
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
1
2
3
4
A
B
C
D
Priority CUCM SIP Trunk SP SIP Trunk
CUBE
A
Inbound LAN Dial-Peer
IP
PSTN
Inbound WAN Dial-Peer Inbound Calls
Outbound Calls
dial-peer voice 5 voip
incoming called-number 654321
Understanding Inbound Dial-Peer Matching Techniques
dial-peer voice 6 voip
answer-address 555
dial-peer voice 7 voip
destination-pattern 555
voice class uri 1001 sip
host ipv4:10.1.1.1
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 1 voip
incoming uri via 1001
dial-peer voice 2 voip
incoming uri request 2001
dial-peer voice 3 voip
incoming uri to 2001
dial-peer voice 4 voip
incoming uri from 1001
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
1
2
3
4
A
B
C
D
Priority CUCM SIP Trunk SP SIP Trunk
CUBE
A
Inbound LAN Dial-Peer
IP
PSTN
Inbound WAN Dial-Peer Inbound Calls
Outbound Calls
Understanding Inbound Dial-Peer Matching Techniques
dial-peer voice 5 voip
incoming called-number 654321
dial-peer voice 6 voip
answer-address 555
dial-peer voice 7 voip
destination-pattern 555
voice class uri 1001 sip
host ipv4:10.1.1.1
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 1 voip
incoming uri via 1001
dial-peer voice 2 voip
incoming uri request 2001
dial-peer voice 3 voip
incoming uri to 2001
dial-peer voice 4 voip
incoming uri from 1001
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
1
2
3
4
A
B
C
D
Priority CUCM SIP Trunk SP SIP Trunk
CUBE
A
Inbound LAN Dial-Peer
IP
PSTN
Inbound WAN Dial-Peer Inbound Calls
Outbound Calls
Understanding Inbound Dial-Peer Matching Techniques
dial-peer voice 5 voip
incoming called-number 654321
dial-peer voice 6 voip
answer-address 555
dial-peer voice 7 voip
destination-pattern 555
voice class uri 1001 sip
host ipv4:10.1.1.1
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 1 voip
incoming uri via 1001
dial-peer voice 2 voip
incoming uri request 2001
dial-peer voice 3 voip
incoming uri to 2001
dial-peer voice 4 voip
incoming uri from 1001
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
1
2
3
4
A
B
C
D
Priority CUCM SIP Trunk SP SIP Trunk
CUBE
A
Inbound LAN Dial-Peer
IP
PSTN
Inbound WAN Dial-Peer Inbound Calls
Outbound Calls
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:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
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
Understanding Outbound Dial-Peer Matching Techniques
dial-peer voice 2 voip
destination-pattern 654321
carrier-id target orange
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 3 voip
destination uri 2001
dial-peer voice 4 voip
destination-pattern 654321
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 1 voip
destination uri 2001
carrier-id target orange
80
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
1
2
3
4
Priority
CUCM SIP Trunk SP SIP Trunk
CUBE
A
Outbound LAN Dial-Peer
IP
PSTN
Inbound Calls
Outbound Calls
Understanding Outbound Dial-Peer Matching Techniques
dial-peer voice 2 voip
destination-pattern 654321
carrier-id target orange
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 3 voip
destination uri 2001
dial-peer voice 4 voip
destination-pattern 654321
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 1 voip
destination uri 2001
carrier-id target orange
81
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
1
2
3
4
Priority
CUCM SIP Trunk SP SIP Trunk
CUBE
A
Outbound LAN Dial-Peer
IP
PSTN
Inbound Calls
Outbound Calls
Understanding Outbound Dial-Peer Matching Techniques
dial-peer voice 2 voip
destination-pattern 654321
carrier-id target orange
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 3 voip
destination uri 2001
dial-peer voice 4 voip
destination-pattern 654321
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 1 voip
destination uri 2001
carrier-id target orange
82
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
1
2
3
4
Priority
CUCM SIP Trunk SP SIP Trunk
CUBE
A
Outbound LAN Dial-Peer
IP
PSTN
Inbound Calls
Outbound Calls
Understanding Outbound Dial-Peer Matching Techniques
dial-peer voice 2 voip
destination-pattern 654321
carrier-id target orange
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 3 voip
destination uri 2001
dial-peer voice 4 voip
destination-pattern 654321
voice class uri 2001 sip
host ipv4:10.2.1.1
dial-peer voice 1 voip
destination uri 2001
carrier-id target orange
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
1
2
3
4
Priority
CUCM SIP Trunk SP SIP Trunk
CUBE
A
Outbound LAN Dial-Peer
IP
PSTN
Inbound Calls
Outbound Calls
CUBE Advanced Call Routing
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:654321@10.2.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
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
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
Additional Headers for Outbound Dial-Peer Matching
Match based on DIVERSION Header of incoming
INVITE
Match Based on URI of incoming INVITE message with
or without carrier-id target
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-
tag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........ Match based on CALLING Number
Match based on CALLED Number with or without
carrier-id target
SIP Trunk SP SIP Trunk
CUBE
A
Outbound LAN Dial-Peer
IP PSTN
Outbound WAN Dial-Peer
Inbound Calls
Outbound Calls
Match Based on FROM Header of incoming INVITE
Match Based on TO Header of incoming INVITE
Match Based on VIA Header of incoming INVITE
Match based on REFERRED-BY Header of incoming
INVITE
Introducing Outbound Dial-peer Provision Policy • Flexibility to choose how outbound dial-peers are selected
• Dynamically set the priority based on Inbound dial-peers
• Additional Inbound Leg Headers for Outbound Dial-peer Matching
VIA FROM TO DIVERSION REFERRED-BY Calling Number
• User-defined outbound dial-peer provision policy on a per incoming call bases
1. A provision policy contains two rules to save the match attributes and its precedence
2. Up to two match attributes can be defined from each rule of a provision policy
3. A provision policy setup will be used to match outbound dial-peers once it is associated to an incoming VoIP call.
• Outbound dial-peer match attributes
destination uri-via destination uri-diversion destination e164-pattern-map
destination uri-to destination uri-referred-by destination uri
destination uri-from destination calling destination-pattern
Dial-peer Provision Policy Configuration
1. Define Voice Class Dial-peer Provision Policy
CUBE(config)#voice class dial-peer provision-policy <tag>
CUBE(config-class)# description “Match outbound dial-peer based on this Criteria”
CUBE(config-class)#preference ?
<1-2> Preference order
CUBE(config-class)#preference 1 first-attribute second-attribute
called Match called number calling Match calling number
carrier-id Match carrier id diversion Match diversion uri
from Match from uri to Match to uri
uri Match destination uri via Match via uri
referred-by Match referred-by uri
voice class dial-peer provision-policy <tag>
description ‘Match outbound dial-peer based on criteria defined here’
preference 1 first-attribute second-attribute
preference 2 first-attribute second-attribute
Dial-peer Provision Policy Configuration – Cont’d
2. Associate Voice Class Provision Policy to an Incoming Dial-peer
dial-peer voice 1 voip
description Inbound Dial-peer
destination provision-policy <tag>
3. Define Outbound Dial-peer with match patterns based on attributes in a policy
CUBE(config)#dial-peer voice 2 voip
CUBE(config-dial-peer)#description Outbound Dial-peer
CUBE(config-dial-peer)#destination ?
calling Match destination calling number
e164-pattern-map Configure voice class to match destination e164-pattern-map
uri Configure voice class to match destination URI
uri-diversion voice class uri to match sip diversion header
uri-from voice class uri to match sip from header
uri-referred-by voice class uri to match sip referred-by header
uri-to voice class uri to match sip to header
uri-via voice class uri to match sip via header
Dial-peer Provision Policy Configuration – Cont’d
Configuring a match command for an outbound dial-peer according to the provision policy rule attribute configured
Provision Policy Rule Attribute Outbound Dial-peer Match command
Called destination-pattern pattern
destination e164-pattern-map pattern-map-class-id
Calling destination calling e164-pattern-map pattern-map-class-id
carrier-id carrier-id target
Uri destination uri uri-class-tag
Via destination uri-via uri-class-tag
To destination uri-to uri-class-tag
from destination uri-from uri-class-tag
diversion destination uri-diversion uri-class-tag
referred-by destination uri-referred-by uri-class-tag
For Your
Reference
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
Multiple Incoming Patterns Under Same Incoming/Outgoing Dial-peer
SIP Trunk SP SIP Trunk
CUBE
IP PSTN A
(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 Inbound 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
9192002010 5101001010 4081001010
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
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
Destination Dial-peer Group Configuration
voice class dpg 10000
description Voice Class DPG for DP Source SJ
dial-peer 1001 preference 1
dial-peer 1002 preference 2
dial-peer 1003
!
dial-peer voice 100 voip
description DP Source SJ w/voice class dpg
incoming called-number 1341
destination dpg 10000
dial-peer voice 1001 voip
description DPG 10000
destination-pattern 1341
session protocol sipv2
session target ipv4:10.1.1.1
!
dial-peer voice 1002 voip
description DPG 10000
destination-pattern 1341
session protocol sipv2
session target ipv4:10.1.1.2
!
dial-peer voice 1003 voip
description DPG 10000
destination-pattern 1341
session protocol sipv2
session target ipv4:10.1.1.3
1. Incoming Dial-peer
is first matched 2. Now the DPG associated
with the INBOUND DP is
selected
Media Manipulation
Audio Transcoding and Transrating
• Transcoding (12.4.20T)
• One voice codec to any other codec E.g. iLBC-G.711 or iLBC-G.729
• Support for H.323 and SIP
• CUCM 7.1.5 or later supports universal Transcoding
• Transrating (15.0.1M)
• Different packetizations of the same codec
• E.g. G.729 20ms to G.729 30ms
• Support for SIP-SIP calls
• No sRTP support with transrating
G.729 30 ms
CUBE
• Transcoding: G.711, G.723.1, G.726, G.728,
G.729/a, iLBC, G.722
• Transrating: G.729 20ms ↔ 30ms (AT&T)
Supported Codecs Packetization
(ms)
G.711 a-law 64 Kbps 10, 20, 30
G.711 µlaw 64 Kbps 10, 20, 30
G.723 5.3/6.3 Kbps 30, 60
G.729, G.729A, G.729B, G.729AB 8 Kbps
10, 20, 30, 40, 50, 60
G.722—64 Kbps 10, 20, 30
SP VoIP
Enterprise
VoIP
dial-peer voice 2 voip codec g729r8 bytes 30 fixed-bytes
iLBC, iSAC,
Speex IP Phones:
G.711, G.729 20 ms,
G.722
!Call volume (gain/loss) adjustment dial-peer voice 2 voip audio incoming level-adjustment x audio outgoing level-adjustment y
Configuration for SCCP based Transcoding (ISR-G2/4K)
voice-card 1
dspfarm
dsp services dspfarm
2. telephony-service configuration
telephony-service
sdspfarm units 1
sdspfarm transcode sessions 128
sdspfarm tag 1 CUBE-XCODE
max-ephones 10
max-dn 10
ip source-address
<CUBE_internal_IP> port 2000
3. sccp configuration
sccp local GigabitEthernet0/0
sccp ccm <CUBE_internal_IP> identifier 1 version 4.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register CUBE-XCODE
4. dspfarm profile configuration
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729r8
maximum sessions 10
associate application SCCP
1. Enabling dspfarm services
under voice-card
For Your
Reference
Configuration for LTI based Transcoding (ISR-G2/4K & ASR)
voice-card 0/1
dspfarm
dsp services dspfarm
2. dspfarm profile configuration
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729abr8
codec g729ar8
codec ilbc
maximum sessions 100
associate application CUBE
1. Enabling dspfarm services
under voice-card
Feature Notes:
• This uses Local Transcoding Interface to
communicate between CUBE and DSPs
• Also available on ISR-G2 starting IOS 15.2.3T
• Can only be used if CUBE invokes the DSP
for media services
• CUCM cannot invoke DSPs using this LTI
interface
Call Recording
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 1
recorder parameter
media-recording 20
dial-peer voice 1 voip
description dial-peer that needs to be forked
session protocol sipv2
media-class 1
dial-peer voice 20 voip
description dial-peer pointing to MediaSense
session protocol sipv2
session target ipv4:<Mediasense_IP>
• 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
SIP SIP
SIP
A
SP SIP
CUBE
RTP
RTP RTP
MediaSense
Needs to
match
Dial-peer based
Audio only Media Forking for an Audio/Video Call
media profile recorder 100
media-type audio media-recording 20
dial-peer voice 1 voip
description dial-peer that needs to be forked
session protocol sipv2
media-class 1
dial-peer voice 20 voip
description dial-peer pointing to MediaSense
session protocol sipv2
session target ipv4:<Mediasense_IP>
• 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
SIP SIP
SIP
A
SP SIP
CUBE
RTP
RTP RTP
MediaSense
CUBE Controlled Recording
media-class 1
recorder profile 100
• Support for forwarding any 3rd
party IP PBX GUID to the
recording server
CUCM (10.X or later) Controlled Recording
1. Enable HTTP on IOS ip http server
http client persistent
2. Enable the API on IOS uc wsapi
source-address [IP_Address_of_CUBE]
3. Enable XMF service within the API provider 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
• Selective Recording • Mobile/SNR/MVA Calls • Recording Call Preservation
• With XE3.13/IOS15.4(3)M, CUBE supports SRTP-SRTP, SRTP-RTP, RTP-SRTP recording. Feature on CUCM roadmap • CUBE HA not supported with CUCM controlled Recording
High Availability
CUBE HA Design Considerations on ISR-G2 for Box-to-Box Redundancy
• All signaling is sourced from/to the Virtual IP Address
• Lower address for both the interfaces (Gig0/0 and Gig0/1) should be on the same platform, which is used as a tie breaker for the HSRP Active state
• HSRP Group number should be unique to a pair/interface combination on the same L2
• Both interfaces of the same group have to be configured with the same priority
• Multiple HSRP interfaces require preemption with interface tracking to be configured
• No media-flow around, SDP-Passthru, or UC Services API support for CUBE HA
CUBE HA Design Considerations on ISR-G2 for Box-to-Box Redundancy – Cont’d
• Both platforms must be connected via the same physical Switch across all interfaces for CUBE HA to
work. Cannot have WAN terminated on CUBEs directly or Data HSRP on either side
• TDM or VXML GW cannot be collocated with CUBE HA
• 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.
• 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
• Upon failover, the ACTIVE CUBE goes through a reload
redundancy inter-device scheme standby SB voice service voip mode border-element allow-connections sip to sip redundancy ipc zone default association 1 no shutdown protocol sctp local-port 5000 local-ip 10.10.1.12 remote-port 5000 remote-ip 10.10.1.11
Define Redundancy scheme: Creates
interdependency b/w CUBE redundancy & HSRP
CUBE 1 CUBE 2
Turn on CUBE Redundancy
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
CUBE Configuration on ISR-G2 Box-to-Box Redundancy
redundancy inter-device scheme standby SB voice service voip mode border-element allow-connections sip to sip redundancy ipc zone default association 1 no shutdown protocol sctp local-port 5000 local-ip 10.10.1.11 remote-port 5000 remote-ip 10.10.1.12
interface GigabitEthernet0/0 ip address 10.10.1.11 255.255.255.0 standby version 2 standby 1 ip 10.10.1.13 standby delay minimum 30 reload 60 standby 1 name SB standby 1 preempt standby 1 track 2 decrement 10 standby 1 priority 50 interface GigabitEthernet0/1 ip address 128.107.60.71 255.255.255.0 standby version 2 standby 10 ip 128.107.60.73 standby delay minimum 30 reload 60 standby 10 preempt standby 10 track 1 decrement 10 standby 10 priority 50
interface GigabitEthernet0/0 ip address 10.10.1.12 255.255.255.0 standby version 2 standby 1 ip 10.10.1.13 standby delay minimum 30 reload 60 standby 1 name SB standby 1 preempt standby 1 track 2 decrement 10 standby 1 priority 50 interface GigabitEthernet0/1 ip address 128.107.60.72 255.255.255.0 standby version 2 standby 10 ip 128.107.60.73 standby delay minimum 30 reload 60 standby 10 preempt standby 10 track 1 decrement 10 standby 10 priority 50
Outside
interfaces:
HSRP group 10
Inside
interfaces:
HSRP group 1
CUBE 1 CUBE 2
CUBE Configuration on ISR-G2 Box-to-Box Redundancy
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
!
track 1 interface Gig0/0 line-protocol
track 2 interface Gig0/1 line-protocol
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
Configure Interface Tracking
• All signaling is sourced from/to the Virtual IP Address
• Lower address for all the interfaces (Gig0/0/0, Gig0/0/1, and Gig0/0/2) should be on the same platform
• Redundancy Interface Identifier, rii (HSRP Group number) should be unique to a pair/interface combination on the same L2
• Configuration on both the CUBEs must be identical including physical configuration and must be running on the same type of platform and IOS version. Loopback interfaces cannot be used as bind as they are always up.
• Multiple HSRP interfaces require interface tracking to be configured
CUBE
CUBE
CUBE-1
CUBE-2
GE 0/0/1 – 20.20.1.1
GE 0/0/1 – 20.20.1.2
20.20
.1.3
GE 0/0/0 – 10.10.1.1
GE 0/0/0 – 10.10.1.2
10.1
0.1
.3
CUCM
10.10.1.10
SP IP
Network
Y.Y.Y.Y
red
un
da
ncy
ri
i 1red
un
da
ncy
rii 2Keepalives
LAN Virtual IP
WAN Virtual IP
GE 0/0/2 – 3.3.1.1
GE 0/0/2 – 3.3.1.2
CUBE HA Design Considerations on ASR1K/ISR-4K for Box-to-Box Redundancy
CUBE HA Design Considerations on ASR1K/ISR-4K for Box-to-Box Redundancy – Cont’d
• Transcoding (LTI based) call flows are preserved starting XE 3.15. Requires same SPA-DSP capacity/placement
• Both platforms must be connected via the same physical Switch for CUBE HA to work. Cannot have WAN terminated on CUBEs directly or Data HSRP on either side
• Upon failover, starting XE3.11, the ACTIVE CUBE can be moved to PROTECTED state to avoid reload
• It is mandatory to use separate interface for redundancy (RG Control/data, Gig0/0/2). i.e interface used for traffic cannot be used for HA keepalives and checkpointing.
• CUBE B2B HA on ASR is not supported over a crossover cable connection for the RG-control/data link or across data centers
CUBE SIP Trunk Monitoring with OOD Options message
CUCM SIP Trunk SP SIP Trunk SP SIP
CUBE
A
OOD Options
200 OK
OOD Options
INVITE
DP 100 = ACTIVE
Timeout – no response
DP 100 = BUSYOUT
OOD Options
503 Service Unavailable
OOD Options
INVITE INVITE
200 OK 200 OK
dial-peer voice 100 voip
voice-class sip options-keepalive
up-interval 20 down-interval 20 retry 3
Three timers that can be configured:
• up-Interval: OPTIONS keepalive
timer interval for UP endpoint
• down-interval: OPTIONS keepalive
timer interval for DOWN endpoint
• retry: Retry count for OPTIONS
keepalive transmission
Warning: • Each dial-peer that has options
message configured sends out a
separate message.
• EEM Script can be used to busyout
other dial-peers
OOD OPTIONS Ping Keepalive Enhancement
• Each dial-peer that has OPTIONS message configured sends out a separate message, even if the session targets are same
• Network bandwidth and process runtime are wasted in CUBE and remote targets to sustain duplicate OOD OPTIONS Ping heartbeat keepalive connection
• Consolidate SIP OOD Options Ping connections by grouping SIP dial-peers with same OOD Options Ping setup
• New CLI : “voice class sip-keepalive-profile <tag>” is used to define OOD OPTIONS Ping setup
• Consolidated SIP OOD Options Ping connection will then be established with a target for multiple SIP dial-peers with the same target and OOD Options Ping profile setup
CUCM SIP Trunk SP SIP Trunk SP SIP
CUBE
A
OOD Options (DP 100)
200 OK
DP 100 : Session Target IPv4:1.1.1.1
INVITE INVITE (DP 100)
200 OK 200 OK
DP 200: Session Target IPv4:1.1.1.1
OOD Options (DP 200)
200 OK
DP 300: Session Target IPv4:1.1.1.1
OOD Options (DP 300)
200 OK
DP 400: Session Target IPv4:1.1.1.1
OOD Options (DP 400)
200 OK
OOD OPTIONS Ping Keepalive Enhancement - Configuration
• With OOD Options Ping Keepalive group, an options ping keepalive connection is established on per remote target base as opposed an options ping keepalive connection established per dial-peer basis
• Up to 10,000 “voice class sip-options-keepalive <tag>” can be defined per system
• Either legacy “sip options-keepalive” or the new “sip options-keepalive profile <tag>” can be configured on a dial-peer
voice class sip-options-keepalive 1
description UDP Options consolidation
down-interval 49
up-interval 180
retry 7
transport udp
dial-peer voice 1 voip
destination-pattern 6666
session protocol sipv2
session target ipv4:10.104.45.253
voice-class sip options-keepalive profile 1
dial-peer voice 2 voip
destination-pattern 5555
session protocol sipv2
session target ipv4:10.104.45.253
voice-class sip options-keepalive profile 1
Single OOD Option
Ping Group applied
to multiple dial-peers
with same session
targets
Sample Show command output
CUBE#sh voice class sip-options-keepalive 1
Voice class sip-options-keepalive: 1 AdminStat: Up
Description: UDP Options consolidation
Transport: udp Sip Profiles: 0
Interval(seconds) Up: 180 Down: 49
Retry: 7
Peer Tag Server Group OOD SessID OOD Stat IfIndex
-------- ------------ ---------- -------- -------
1 4 Active 9
2 4 Active 10
OOD SessID: 4 OOD Stat: Active
Target: ipv4:10.104.45.253
Transport: udp Sip Profiles: 0
Contact Center Features
Mid-call Xcoder Insert/Drop – Codec Renegotiation
Call arrives on G.729 SIP trunk
CVP connects call to speech recognition server that requires G.711. Since provider does not support G711 CUBE inserts transcoder
CVP xfers call to a remote agent that uses G.729
CUBE drops xcoder and e2e call becomes G.729 again
1
2
3
4
Transcoder Inserted
Transcoder Dropped
SP SIP
CVP
SIP
G.711
G.729
G.729 / G.711
1
Provider supports only
G.729 codec
CUBE
G.711
2
Call Xfer (signaling only)
3
G.729
4
REFER Handling for Contact Centers
• Enables CUBE to handle REFER messages more efficiently in contact center deployments
• CUBE can operate in either consume mode or pass-through mode
SIP SP
CVP
1. REFER
3. INVITE
Based on “Refer-To” header,
CUBE does outbound dial-peer
match and sends out an INVITE
message No supplementary-service sip refer
supplementary-service media-renegotiate
CUBE will pass across the
Refer message “as-is” without
any modification
A
CUBE
2. INVITE
SIP SP
CVP
1. REFER
A
CUBE
2. REFER
REFER Consumption
REFER Pass-through (Default mode)
REFER Handling Enhancement
• A new CLI, “refer consume”, has been added to the SIP dial peer.
• The final decision to consume or pass-through REFER is determined based on this new CLI option configured on the Refer-To dial-peer.
“supplementary-service sip refer”
Configured globally or
at inbound dial-peer
“refer consume”
Configured at dial-
peer that matches
‘refer-To’
Outcome
Yes (default) No (default) REFER Pass-through
Yes (default) Yes REFER Consume
No No (default) REFER Consume
No Yes REFER Consume
• SIP Trunking and CUBE Overview
• CUBE Architecture (Physical & Virtual)
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE
• CUBE Management & Troubleshooting
• Futures & Key Takeaways
Agenda
Serviceability
New CUBE Serviceability Features
Histogram for Call rate
Histogram for Concurrent calls
Histogram for Call duration
Histogram for SIP message rate
High/Low watermark for Call Rate
High/Low watermark for Concurrent calls
High/Low watermark for SIP message rate
Histogram for Call Failure Rate
High/Low watermark for Call Failure Rate
1122222357676678753222211111122247545789774322213311112245654598843333222
10
9 * *
8 * ** ***
7 * * *** * ***** * ##*
6 ******** * ***** ** *##*
5 *########* #* *####* *######*
4 *########* *#***####** *########*
3 **########** *#########** ** *########*****
2 ******#########***** ****##########**** ** ***########********
1 *######################################################################*
0....5....1....1....2....2....3....3....4....4....5....5....6....6....7..
0 5 0 5 0 5 0 5 0 5 0 5 0
Call switching rate / CPS (last 72 hours)
* = maximum calls/s # = average calls/s
Example:
show call history stats cps
Call Arrival Rate
Call History Stats – Graphical or Tabular form
show call history stats connected [table]
Last 60 sec, 60 minutes, 72 hours
Ability to sort dial-peers show run dial-peer sort
dial-peer (default) dial-peer sort dial-peer sort descending
dial-peer voice 4020 pots
destination-pattern 4020
port 0/2/0
!
dial-peer voice 5000 voip
destination-pattern 5...
session protocol sipv2
session target ipv4:1.4.65.5
!
dial-peer voice 5 pots
incoming called-number 1...
port 1/0/0:23
dial-peer voice 5 pots
incoming called-number 1...
port 1/0/0:23
!
dial-peer voice 4020 pots
destination-pattern 4020
port 0/2/0
!
dial-peer voice 5000 voip
destination-pattern 5...
session protocol sipv2
session target ipv4:1.4.65.5
dial-peer voice 5000 voip
destination-pattern 5...
session protocol sipv2
session target ipv4:1.4.65.5
!
dial-peer voice 4020 pots
destination-pattern 4020
port 0/2/0
!
dial-peer voice 5 pots
incoming called-number 1...
port 1/0/0:23
Dial Peer tag
New CUBE Serviceability Features
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-legs Connected
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
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
• 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 messages in debugs, issue the following command
debug ccsip non-call
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
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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