design & implementation of sip trunking using cisco’s ... · pdf filedesign &...
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© 2011 Cisco and/or its affiliates. All rights reserved. 1
Design & Implementation of SIP Trunking using Cisco’s Session Border Controllers
Graham Francis – CEO, The SIP School
Darryl Sladden – Technical Marketing Manager, Cisco
Pashmeen Mistry – Technical Marketing Engineer, Cisco
October 27th 2011
© 2011 Cisco and/or its affiliates. All rights reserved. 2
© 2011 Cisco and/or its affiliates. All rights reserved. 3
• Founded in April 2000
• 5300+ Students
• Provide the Industry recognised SSCA® SIP Certification program, endorsed by the TIA + more.
• eLearning in modular format
• Unique as content evolves as SIP evolves
• Connected with Cisco to provide SIP foundation training
• http://cisco.thesipschool.com / Discount codes later.
• Now, let’s talk about why we’re all here today and we’ll start with SIP
© 2011 Cisco and/or its affiliates. All rights reserved. 4
© 2011 Cisco and/or its affiliates. All rights reserved. 5
6644 55
9977 88
##** 00
VideoVideo
3311 22
HoldHold
6644 55
9977 88
##** 00
VideoVideo
3311 22
HoldHold
SIPSupport Video?
SIP OK
Call 1003
VOICE MEDIA
1003 OK 1002 OKSIPWant to talk?
SIP OK
VIDEO MEDIA
© 2011 Cisco and/or its affiliates. All rights reserved. 6
© 2011 Cisco and/or its affiliates. All rights reserved. 7
Data from
Frost & Sullivan
Now $7.44 billion by 2017
© 2011 Cisco and/or its affiliates. All rights reserved. 8
•Voice Communications
•Less Money
•Equal / Better quality
•Greater functionality
© 2011 Cisco and/or its affiliates. All rights reserved. 9
•Worldwide Phenomenon
•Will happen
•One day, no PSTN
• It is Easy to implement
© 2011 Cisco and/or its affiliates. All rights reserved. 10
Unified Clients
Unified Server
Inc.Registrar and
Location services
SIP IP Phones
DirectoryDNS
Messaging Server
GatewayPBX
Firewall / NAT
ITSP
ldap
naptr
sip
sip
sip sip sip
sip
sipSip trunk
© 2011 Cisco and/or its affiliates. All rights reserved. 11
Internet ISP
Data
Asymmetric DSL
TDM / PBX
ITSP
TDM to SIP/RTP
Gateway
SIP Trunks
© 2011 Cisco and/or its affiliates. All rights reserved. 12
Data
TDM / PBX
TDM to SIP/RTPGateway
Voice
Internet ISP
ITSP
Switch
SIP / PBX
IP Network
© 2011 Cisco and/or its affiliates. All rights reserved. 13
The road to compatibility
© 2011 Cisco and/or its affiliates. All rights reserved. 14
ITSP
Network
Your PBX
SIP Registrar
G.711 G.729G.711 to G.729
Media
SIP Signaling
REGISTER
Secured
SBC
B2BUA
SBC
B2BUA
© 2011 Cisco and/or its affiliates. All rights reserved. 15
© 2011 Cisco and/or its affiliates. All rights reserved. 16
• Changing Landscapes –VoIP Islands to VoIP Interconnects
• Unified communications SIP Trunks to destinations beyond the Enterprise
IPA
IPA
Enterprise Domain 1 Enterprise Domain 2
Narrowband voice to Rich-media Interconnect
A A
Enterprise Domain 1 Enterprise
Domain 2SP VoIPSBC SBCCUBE CUBE
� Extend rich-media collaboration to vendors, partners and customers
� A Cisco Unified Border Element (CUBE) provides b2b interconnectivity for secure rich-media services
IP IP
Enabling Business-to-Business Collaboration
© 2011 Cisco and/or its affiliates. All rights reserved. 17
Capture a 53% cost savings opportunity
Avg.
-40%
© 2011 Cisco and/or its affiliates. All rights reserved. 18
A
ACVP
Branch Offices
Campus Contact Center
A
ACVP
SP SIP
A
ACVP
SP SIP
1. TDM Trunking – Yesterday
2. TDM and IP Trunking – Today
3. IP Trunking – TomorrowCampus Contact Center
Campus Contact Center
Branch Offices
Branch Offices
© 2011 Cisco and/or its affiliates. All rights reserved. 19
� I have multiple PBXs that all need to have SIP Trunking enabled in order to get the best Return on Investment (ROI).
� I would like to centralize all of my SIP Trunking in a single location.
� SIP Trunking is complex new technology, how do I make Trouble shooting easier.
� How can I ensure that I am compliant with my company’s security policies when I implement SIP Trunking ?
Challenge Impact of an SBC
� Allows you to have a single interconnect point to your Service Provider across multiple disparate systems.
� Allows you to scale your SIP Trunk solution while only connecting to one device.
� Allows a single point of troubleshooting for your SIP Trunks. A device that is supported by Cisco allows you to have one vendor support your entire solution.
� SBC’s ensures security on SIP Trunks. An SBC from a trusted vendor such as Cisco incorporates security in all aspects from an embedded firewall to administrative control on changes.
Features of a Cisco SBC
Slide 19
mrf4 New oneMike Fratesi, 14/05/2008
© 2011 Cisco and/or its affiliates. All rights reserved. 20
Overview
© 2011 Cisco and/or its affiliates. All rights reserved. 21
An Integrated Network Infrastructure Service
VXML
SRSTRSVP Agent
Cisco Unified Border Element
� Address Hiding
� H.323 and SIP interworking
� DTMF interworking
� SIP security
� Transcoding
Unified CM Conferencing and
Transcoding
GK
TDM Gateway
� Voice and Video TDM Interconnect
� PSTN Backup
Routing, FW, IPS, QoS
WAN Interfaces
Note: An SBC appliance wouldhave only these features
CUBE
Note: Some features/components may require additional licensing
© 2011 Cisco and/or its affiliates. All rights reserved. 22
2800 ISR
3800 ISR
2900 ISR G2
AS5000XM
ASR 1004/6 RP2
Active Voice Call (Session) Capacity
CP
S
<5
8-12
50-150
12-16K+<50 500-600 600-800 900-1000
3900 ISR G217
1500-1700
ASR 1002
2801 ISR
3900E ISR G2
2000-2500
20-35
4
800/1861 ISR
ASR 1001
10-12K
50-100
End of Life platforms
Last IOS Release: 15.1.4MIntroduced
in Nov ‘10
Introduced in Mar ‘11
© 2011 Cisco and/or its affiliates. All rights reserved. 23
CUBE Session Capacity Summary
Platform CUBE SessionsC880/C890 SKUs 5-25
1861 5-15
2801 55
2811 110
2821 200
2851 225
3825 400
3845 500
AS5000XM 600
2901 100
2911 200
2921 400
2951 600
3925 800
3945 950
3925E 2100
3945E 2500
ASR1002/1004/1006 RP1 1750
ASR1001 10000
ASR1004/1006 RP2 16000
ASR1001 introduced in RLS 3.2 in Nov 2010
Introduced in March 2011
Reference
End of Life PlatformsLast IOS Release:
15.1.4M
© 2011 Cisco and/or its affiliates. All rights reserved. 24
Platform Single-Use LicensesActive-Standby B2B
Redundancy Licenses
Cisco 2901, 2911, 2921 ISR G2
FL-CUBEE-5
FL-CUBEE-25
FL-CUBEE-100
FL-CUBEE-5-RED
FL-CUBEE-25-RED
FL-CUBEE-100-RED
Cisco 2951, 3925 ISR G2
FL-CUBEE-5
FL-CUBEE-25
FL-CUBEE-100
FL-CUBEE-500
FL-CUBEE-5-RED
FL-CUBEE-25-RED
FL-CUBEE-100-RED
FL-CUBEE-500-RED
Cisco 3945, 3925E, 3945E ISR G2
FL-CUBEE-5
FL-CUBEE-25
FL-CUBEE-100
FL-CUBEE-500
FL-CUBEE-1000
FL-CUBEE-5-RED
FL-CUBEE-25-RED
FL-CUBEE-100-RED
FL-CUBEE-500-RED
FL-CUBEE-1000-RED
Cisco ASR1000
FLASR1-CUBEE-100P
FLASR1-CUBEE-500P
FLASR1-CUBEE-1KP
FLASR1-CUBEE-4KP
FLASR1-CUBEE-16KP
FLASR1-CUBEE-100R
FLASR1-CUBEE-500R
FLASR1-CUBEE-1K-R
FLASR1-CUBEE-4K-R
FLASR1-CUBEE-16KR
More info in the CUBE Ordering Guide: http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/order_guide_c07_462222.html
Reduced Pricing for redundancy
© 2011 Cisco and/or its affiliates. All rights reserved. 25
Advanced Features
© 2011 Cisco and/or its affiliates. All rights reserved. 26
Network-based Media
Recording Solution
Business to Business
Telepresence
SIP Trunks for PSTN Access
IVR Integration for Contact
Centers
SIP
H.323
SP VOIPServivces
SIP Trunk
SBC
TDM
CUBE
SP IP Network
SIP
SBCCUBE
SIP
MediaSense
RTP
RTP
Partner API
CUBE
SIPSP IP
NetworkSBC
CVPvXML Server
Media Server
A SP IP Network
SIP SIP ASBC
CUBECUBE
© 2011 Cisco and/or its affiliates. All rights reserved. 27
Telecommuter/SOHO
V V V V
SP IP Network
TDM-based PSTN
Class 4/5 Switch
MPLS
Campus HeadquartersBranch Office
TDM Trunk Call Path
Voice
Voice
Voice
IP Trunk Call Path
Voice
Voice
CUBE
MPLS
Data Data
Data
VPNVPN
© 2011 Cisco and/or its affiliates. All rights reserved. 28
Characteristics of Centralized Operational Benefits Challenges
� Central Site is the only location
with SIP session connectivity to
IP PSTN
� Voice services delivered to
Branch Offices over the
Enterprise IP WAN (usually
MPLS)
� Media traffic hairpins through
central site between SP and
branches
� Centralizes Physical
Operations
� Centralizes Dial-Peer
Management
� Centralizes SIP Trunk
Capacity
� Increased campus and branch bandwidth, CAC, latency; media optimization
� HA in campus (single point of failure)
� Survivability (backup branch call processing)
� Emergency services
� Legal/Regulatory, Geographical
Site-SP Media
Centralized
A
IP PSTN
Enterprise
IP WAN
CUBE
© 2011 Cisco and/or its affiliates. All rights reserved. 29
Telecommuter/SOHO
V V V V
SP IP Network
TDM-based PSTN
Class 4/5 Switch
MPLS
Campus HeadquartersBranch Office
MPLS
DataData
Data
VPNVoice
Voice
IP Trunk Call Path
Voice
CUBECUBE
VPN
© 2011 Cisco and/or its affiliates. All rights reserved. 30
Characteristics of
Distributed
Operational Benefits Challenges
� Each site has direct connection
for SIP sessions to SP
� Takes advantage of SP session
pooling, if offered by SP
� Media traffic goes direct from
each branch site to the SP
� Leverages existing branch
routers
� No media hair-pinning thru any
site.
� Lower latency on voice or video
� Built-in Redundancy strategy
� Quickest transition from existing
TDM
� Distributed dial-peer management
� Distributed operational overhead
Site-SP Media
A
Distributed
Enterprise
IP WANCUBE
CUBECUBECUBECUBE
IP PSTN
© 2011 Cisco and/or its affiliates. All rights reserved. 31
Characteristics of Hybrid Benefits
� Connection to SP SIP service is determined on a site
by site basis to be either direct or routed through a
regional site.
� Decision to route call direct or indirect based on
various criteria
� Media traffic goes direct from site to SP or hairpins
through another site, depending on branch
configuration.
� Adaptable to site specific requirements
� Optimizes BW use on Enterprise WAN
� Adaptable to regional SP issues
� Built-in redundancy strategy
CUBE
Hybrid
A
CUBE
A
CUBE
Enterprise
IP WANCUBE
IP PSTN
Site-SP Media
© 2011 Cisco and/or its affiliates. All rights reserved. 32
Cisco Interoperability Portal:www.cisco.com/go/interoperability
� Validated with service providers world-wide
� Tested with 3rd party PBXs
� Standards based
© 2011 Cisco and/or its affiliates. All rights reserved. 33
© 2011 Cisco and/or its affiliates. All rights reserved. 34
Digital/Analog Trunks
SIP/H.323 Trunks
dial-peer voice 2 voipdestination-pattern 9Tsession protocol sipv2session target ipv4:<SIP_Trunk_Provider_IP_Addr>codec g711ulaw
dial-peer voice 2 potsdestination-pattern 9Tport 0/0/0:23
dial-peer voice 1 voipdestination-pattern 9Tsession protocol sipv2session target ipv4:<PBX_IP_Addr>codec g711ulaw
dial-peer voice 1 voipdestination-pattern 1...session protocol sipv2session target ipv4:<PBX_IP_Addr>codec g711ulaw
x 1001
CUBE
SBC
SIP TrunksSIP/H.323 Trunks
x 1001
SIP SP
Change POTS
Call Leg to
VoIP Call Leg
Re-purpose your existing Cisco Voice Gateway’s as Cisco’s Session Border Controller – Cisco Unified Border Element (CUBE)
Buy CUBE License
Only
© 2011 Cisco and/or its affiliates. All rights reserved. 35
IP
CUBE
CUBE
� 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 required for transcoding (calls with dissimilar codecs)
IP
Media Flow-Around
� Signaling and media terminated by the Cisco Unified Border Element
� 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
© 2011 Cisco and/or its affiliates. All rights reserved. 36
CUBE
x1001+1 408-526-6855
INVITE /w SDPINVITE /w SDP
100 TRYING
100 TRYING
10.1.1.1 20.1.1.1
c= 192.168.1.50m=audio abc RTP/AVP 0
c= 20.1.1.1m=audio xxx RTP/AVP 0
180 RINGING
180 RINGING
200 OK200 OK
c= 20.1.1.2m=audio uvw RTP/AVP 0c= 10.1.1.1
m=audio xyz RTP/AVP 0
RTP (Audio)
ACK ACK
192.168.1.50 10.1.1.1 20.1.1.1 20.1.1.2
192.168.1.1
192.168.1.50
SIP SP
SBC
20.1.1.2
Internal Network
ExternalNetwork
B2B User Agent
© 2011 Cisco and/or its affiliates. All rights reserved. 37
SIP/H.323Protocol Stack
SIP/H.323 Protocol Stack
Ingress I/F Egress I/FHW LAN/WAN Interfaces
IOS Infrastructure (ACLs, FW, IPS, VPN)
TCP UDP TLS TCP UDP TLSDSP Hardware
DSP API
DTMF xlationCodec FilteringXcoding Control
Dial-peer Dial-peer
Voice Application CodeL7 Protocol-independent memory structures holding call
state and attributes (CLID, Called #, CodecM)
Signaling
RTP LibraryRTP Library
Media
Signaling
Packets
Physical Interfaces
IOS Infrastructure
TCP/UDP/TLS Voice
stack
SIP/H323 Protocol Stack
Dial-Peer
Voice Application Code
Media
Packets
Physical Interfaces
IOS Infrastructure
RTP Library
DSP (If invoked)
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 38
� Step 0 – Configure IP PBX to route calls to the edge SBC
� Step 1 – Get SIP Trunk details from the Provider
� Step 2 – Turn CUBE Application ON on Cisco routers
� Step 3 – Configure Call routing on CUBE (Incoming & Outgoing Dial-Peers)
� Step 4 – Normalize SIP messages to meet SIP Trunk Provider’s requirements
� Step 5 – Execute the Test Plan
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 39
� Configure CUCM to route calls to CUBE via a SIP/H323 Trunk
� Make sure all different patterns of calls – local, long distance, international, emergency, informational etc.. are pointing to CUBE
SP IP Network
SIP
SBCCUBE
SIP Trunk pointing to CUBE
SIP
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 40
Item SIP Trunk service provider requirement Sample
Response
1 SIP Trunk IP Address (Destination IP Address for INVITES) 20.1.1.2
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
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 41
1. Turn CUBE Application “ON”1. Turn CUBE Application “ON”
2. Global settings to meet SP’s requirements and SIP Trunk towards SP if needed2. Global settings to meet SP’s requirements and SIP Trunk towards SP if needed
voice service voip
mode border-element license capacity 200
allow-connections sip to sip
voice service voip
sip
early-offer forced
header-passing error-passthru
midcall-signaling passthru
voice service voipip address trusted list
ipv4 10.1.1.50ipv4 20.20.20.20
3. Create a trusted list of IP addresses to prevent toll-fraud3. Create a trusted list of IP addresses to prevent toll-fraud
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 42
� Dial-peer – “static routing” table mapping phone numbers to interfaces or IP addresses
SP IP Network
SIPH.323 or SIP
SBCCUBE
INBOUND & OUTBOUND CALLS
LAN Dial-Peers
WAN Dial-Peers
� LAN Dial-Peers – Dial-Peers that are facing towards the IP PBX for sending & 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
© 2011 Cisco and/or its affiliates. All rights reserved. 43
dial-peer voice 100 voipdescription *** LAN side dial-peer ***incoming called-number 9T session protocol sipv2destination-pattern [2-9].........voice-class sip bind control source gig0/0voice-class sip bind media source gig0/0session target ipv4:<CUCM_Address>codec g711ulawdtmf-relay rtp-nte
CUCM sending 9 + All digits dialed
SP will be sending 10 digits inbound
INBOUND DP FOR CALL FROM CUCM TO CUBEOUTBOUND DP FOR CALLS FROM CUBE TO CUCM
Note: If more than 1 CUCM exists, you will have to create multiple such LAN dial-peers with “preference CLI” for CUCM redundancy
© 2011 Cisco and/or its affiliates. All rights reserved. 44
Catch-all for all SP inbound calls
dial-peer voice 201 voipdescription *** WAN side dial-peer_Long distance***translation-profile outgoing Digitstrip_9destination-pattern 91[2-9].........session protocol sipv2voice-class sip bind control source gig0/1voice-class sip bind media source gig0/1session target ipv4:<SIP_Trunk_Provider IP address>dtmf-relay rtp-ntecodec g729r8
dial-peer voice 200 voipdescription *** WAN side Incoming DP ***incoming called-number [2-9].........session protocol sipv2dtmf-relay rtp-nte
INCOMING WAN DIAL-PEER FOR CALLS FROM SP TO CUBE
OUTGOING WAN DIAL-PEER FOR CALLS TO SP FROM CUBE
DP for sending long distance calls to SP
Note: Separate outgoing DP to be created for Local, International, Emergency, Informational calls etc. Thus, for WAN Inbound & Outbound DP are separate
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 45
voice class sip-profiles 400
request INVITE sip-header Diversion modify “sip:(.*>)” “sip:[email protected]>”
request REINVITE sip-header Diversion modify “sip:(.*>)” “sip:[email protected]>”
request ANY sip-header User-Agent modify “User-Agent:(.*)” “User-Agent: Cisco CUCM8.5/IOS-15.1-3”
response ANY sip-header Server modify “Server:(.*)” “Server: Cisco CUCM8.5/IOS-15.1-3”
dial-peer voice 4000 voip
description Incoming/outgoing SP
voice-class sip profiles 400
Received: INVITE sip:[email protected]:5060 SIP/2.0SSSUser-Agent: Cisco-CUCM8.5
SSSDiversion: <sip:[email protected]>;privacy=off;
reason=unconditional;screen=yesSS...m=audio 6001 RTP/AVP 0 8 18 101a=rtpmap:0 PCMU/8000SS...
Configure
SIP Profiles
Apply to
Dial-peer or
Globally
See the
difference
Sent: INVITE sip:[email protected]:5060 SIP/2.0SSS.User-Agent: Cisco CUCM8.5/IOS-15.1-3
SSS.Diversion: <sip:[email protected]>;
privacy=off;reason=unconditional;screen=yesSSS.m=audio 32278 RTP/AVP 18 8 101a=rtpmap:0 PCMU/8000SSS..
voice service voip
sip
sip profiles 1000
SIP Provider
Requirement
1. For Call Forward & Transfer scenarios back to PSTN, the Diversion header should match the
registered DID of your network
2. The User-Agent field in all SIP messages should state the version of PBX and of SBC that is being
used
© 2011 Cisco and/or its affiliates. All rights reserved. 46
� Inbound and outbound Local, Long distance, International calls for G711 & G729 codecs
� 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)
© 2011 Cisco and/or its affiliates. All rights reserved. 47
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:2 Answer 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 g729r8 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
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:1 Originate 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 g729r8 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
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 RemoteIP1 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
© 2011 Cisco and/or its affiliates. All rights reserved. 48
Is CUBE Active ? show cube status
Is the call matching
right Dial-peers ?
Are we sending the
right SIP call to SP based
on their requirements ?
debug voip ccapi inout
debug ccsip messages
CUBE-Version : 9.0SW-Version : 15.2.1T, Platform 2911HA-Type : noneLicensed-Capacity : 200
Oct 26 18:59:01.146: //-1/66A6B1BF8013/CCAPIcc_api_call_setup_ind_common:.................
Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=TRUE,.................
Outgoing Dial-peer=100, Params=0x26E8574, Progress Indication=NULL(0)
Received:INVITE sip:[email protected]:5060 SIP/2.0Date: Wed, 26 Oct 2011 18:59:01 GMTAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYFrom: "Paul Hewson" <sip:[email protected]>;tag=90d94d92-6ee4-45aa-9f18-2d09025c1ee4-27352390................
© 2011 Cisco and/or its affiliates. All rights reserved. 49
SP IP Network
SIPH.323 or SIP
SBCCUBE
SNMP Response
SNMP Query
� Network Management Tools can be used to monitor key CUBE statistics like SIP Trunk status, Trunk utilization, Call Arrival Rate, Call Success/Failure count, voice quality metrics etc..
� Network Management Tools can send SNMP Queries to CUBE
� CUBE responds to the SNMP queries with real time values of the monitored objects
� CUBE can also send SNMP Traps to alert the network management tool of certain events like SIP Trunk failure, link down, high CPU etc.. Network
Management Tool
Some Network Management Tools:- Cisco Unified
Operations Manager
- Arcana Networks- Solarwinds
Some Network Management Tools:- Cisco Unified
Operations Manager
- Arcana Networks- Solarwinds
© 2011 Cisco and/or its affiliates. All rights reserved. 50
Area Information Method
Router Health CPU, Memory, I/f� CISCO-PROCESS-MIB, cpmCPUTotal5minRev
� CISCO-MEMORY-POOL-MIB, ciscoMemoryPoolTable
� IF-MIB, IfEntry
SIP Trunk Status SIP Trunk Status � SIP OOD Options Ping, CLI dial-peer status
Traffic Reports (Calls, Sessions, Capacity Planning, Errors)
Trunk Utilization
� CUBE 1.4: CISCO-VOICE-DIAL-CONTROL-MIB, cvCallVolume
� Older CUBE: DIAL-CONTROL-MIB, callActive
� CISCO-DIAL-CONTROL-MIB, cCallHistoryTable
� CUBE 8.5: SIP RAI Trunk Utilization
Call Arrival Rate � CUBE 1.4: CISCO-VOICE-DIAL-CONTROL-MIB, cvCallRateMonitor
Call Success/Failure
� DIAL-CONTROL-MIB, dialCtlPeerStatsSuccessCalls, dialCtlPeerStatsAcceptCalls, dialCtlPeerStatsFailCalls, dialCtlPeerStatsRefuseCalls
� CISCO-SIP-UA-MIB, cSipStatsErrClient, cSipStatsErrServer, cSipStatsGlobalFail
SIP retries � CISCO-SIP-UA-MIB, cSipStatsRetry
Media Resources (DSPs)
DSP Availability � CISCO-DSP-MGMT-MIB, cdspCardResourceUtilization, cdspDspfarmUtilObjects
Transcoding util.� CUBE 1.4: CISCO-DSP-MGMT-MIB, cdspTotAvailTranscodeSess,
cdspTotUnusedTranscodeSess
MTP utilization� CUBE 1.4: CISCO-DSP-MGMT-MIB, cdspTotAvailMtpSess,
cdspTotUnusedMtpSess
Voice QualityLoss, delay, jitter � CISCO-VOICE-DIAL-CONTROL-MIB, cvVoIPCallActiveTable
IP SLA � CISCO-RTTMON-RTP-MIB, rttMonJitterStatsTable , rttMonLatestJitterOperTable
More info in CUBE Management and Manageability Specification at:http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/white_paper_c11-613550.html
Reference
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 51
! create profileip traffic-export profile TAC mode capture
bidirectionalincoming access-list 123outgoing access-list 123
!! access-list to filter only SIP messages (port 5060) access-list 123 permit udp any any eq 5060access-list 123 permit tcp any any eq 5060!! apply to an interface, default memory is 5Minterface fa0/0
ip traffic-export apply TAC [size <bytes>]
router#traffic-export interface fa0/0 clearrouter#traffic-export interface fa0/0 start
<capture the problem>router#traffic-export interface fa0/0 stop
2. Capture traffic with these exec (enable) level commands
1. Configure capture profile
IP Traffic Capture: http://www.cisco.com/en/US/docs/ios/12_4t/12_4t11/ht_rawip.html
3. Export the pcap file to a server
router#traffic-export interface fa0/0 copy ftp://x.x.x.x/capture.pcap
4. Display ladder diagram (with Wireshark)
Note: The exec cmds don’t appear until a profile has been configured
Note: Allows filtering of calling/called numbers when creating the flow graph
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 52
� High Availability with Inbox & Box to Box Redundancy� Resiliency by alternative routing of INVITEs� Media Forking for recording of calls� Media Enhancements through DSP’s such as Noise
Reduction and Acoustic Shock Prevention� Video Call Handling via CUBE� Additional Audio Codecs such as G722 and wide-band
codecs� Additional SIP messages handled via Trunking specifically
Presence Indicators� SIP Trunks to Webex for Cloud Connected Audio
ShippingShipping
PlannedPlanned
© 2011 Cisco and/or its affiliates. All rights reserved. 53
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