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 27 th 2011 2011 Cisco and/or its affiliates. All rights reserved. 1
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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. 3
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Support Want to Video? talk? Call 1003 1003 OK 1002 OK VOICE MEDIA SIP SIP Hold 1 2 4 5 7 8 * 0 Video 3 6 9 # SIP SIP VIDEO MEDIA OK OK Hold 1 2 4 5 7 8 * 0 Video 3 6 9 # 2011 Cisco and/or its affiliates. All rights reserved. 5
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Now $7.44 billion by 2017 Data from Frost & Sullivan 2011 Cisco and/or its affiliates. All rights reserved. 7
Voice Communications Less Money Equal / Better quality Greater functionality 2011 Cisco and/or its affiliates. All rights reserved. 8
Worldwide Phenomenon Will happen One day, no PSTN It is Easy to implement 2011 Cisco and/or its affiliates. All rights reserved. 9
Unified Clients ITSP Unified Server Inc. Registrar and Location services SIP IP Phones sip sip sip sip sip Sip sip trunk ldap Gateway sip PBX Firewall / NAT naptr Directory DNS Messaging Server 2011 Cisco and/or its affiliates. All rights reserved. 10
TDM / PBX SIP Trunks TDM to SIP/RTP Gateway ITSP Data Asymmetric DSL Internet ISP 2011 Cisco and/or its affiliates. All rights reserved. 11
TDM SIP // PBX ITSP TDM to SIP/RTP Gateway Voice Switch IP Network Data Internet ISP 2011 Cisco and/or its affiliates. All rights reserved. 12
The road to compatibility 2011 Cisco and/or its affiliates. All rights reserved. 13
Your PBX ITSP Network SBC B2BUA REGISTER Secured SBC B2BUA SIP Registrar G.711 G.711 to G.729 G.729 SIP Signaling Media 2011 Cisco and/or its affiliates. All rights reserved. 14
dsladden@cisco.com 2011 Cisco and/or its affiliates. All rights reserved. 15
Enabling Business-to-Business Collaboration Enterprise Domain 1 Enterprise Domain 2 A IP IP A Changing Landscapes VoIP Islands to VoIP Interconnects Unified communications SIP Trunks to destinations beyond the Enterprise Enterprise Domain 1 A IP CUBE SBC Narrowband voice to Rich-media Interconnect SP VoIP SBC Extend rich-media collaboration to vendors, partners and customers A Cisco Unified Border Element (CUBE) provides b2b interconnectivity for secure rich-media services CUBE IP A Enterprise Domain 2 2011 Cisco and/or its affiliates. All rights reserved. 16
Avg. -40% Capture a 53% cost savings opportunity 2011 Cisco and/or its affiliates. All rights reserved. 17
1. TDM Trunking Yesterday A Campus CVP A Contact Center 2. TDM and IP Trunking Today Branch Offices A SP SIP CVP 3. IP Trunking Tomorrow Campus A Contact Center A CVP Branch Offices SP SIP A Campus Contact Center Branch Offices 2011 Cisco and/or its affiliates. All rights reserved. 18
Challenge 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. 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. Features of a Cisco SBC How can I ensure that I am compliant with my company s security policies when I implement SIP Trunking? 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. 2011 Cisco and/or its affiliates. All rights reserved. 19
Slide 19 mrf4 New one Mike Fratesi, 14/05/2008
Overview 2011 Cisco and/or its affiliates. All rights reserved. 20
An Integrated Network Infrastructure Service Cisco Unified Border Element Address Hiding H.323 and SIP interworking DTMF interworking SIP security Transcoding Note: An SBC appliance would have only these features CUBE TDM Gateway Voice and Video TDM Interconnect PSTN Backup Routing, FW, IPS, QoS WAN Interfaces Unified CM Conferencing and Transcoding RSVP Agent VXML SRST 2011 Cisco and/or its affiliates. All rights reserved. 21 GK Note: Some features/components may require additional licensing
ASR 1004/6 RP2 50-150 50-100 ASR 1001 20-35 17 CPS End of Life platforms Last IOS Release: 15.1.4M AS5000XM 3900 ISR G2 ASR 1002 3900E ISR G2 Introduced in Nov 10 8-12 3800 ISR 2800 ISR 2900 ISR G2 <5 2801 ISR 800/1861 ISR Introduced in Mar 11 4 <50 500-600 600-800 900-1000 1500-1700 2000-2500 10-12K 12-16K+ Active Voice Call (Session) Capacity 2011 Cisco and/or its affiliates. All rights reserved. 22
CUBE Session Capacity Summary Reference Platform CUBE Sessions C880/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 Introduced in March 2011 End of Life Platforms Last IOS Release: 15.1.4M ASR1001 introduced in RLS 3.2 in Nov 2010 2011 Cisco and/or its affiliates. All rights reserved. 23
Reduced Pricing for redundancy Platform Cisco 2901, 2911, 2921 ISR G2 Cisco 2951, 3925 ISR G2 Cisco 3945, 3925E, 3945E ISR G2 Cisco ASR1000 Single-Use Licenses FL-CUBEE-5 FL-CUBEE-25 FL-CUBEE-100 FL-CUBEE-5 FL-CUBEE-25 FL-CUBEE-100 FL-CUBEE-500 FL-CUBEE-5 FL-CUBEE-25 FL-CUBEE-100 FL-CUBEE-500 FL-CUBEE-1000 FLASR1-CUBEE-100P FLASR1-CUBEE-500P FLASR1-CUBEE-1KP FLASR1-CUBEE-4KP FLASR1-CUBEE-16KP Active-Standby B2B Redundancy Licenses FL-CUBEE-5-RED FL-CUBEE-25-RED FL-CUBEE-100-RED FL-CUBEE-5-RED FL-CUBEE-25-RED FL-CUBEE-100-RED FL-CUBEE-500-RED FL-CUBEE-5-RED FL-CUBEE-25-RED FL-CUBEE-100-RED FL-CUBEE-500-RED FL-CUBEE-1000-RED 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 2011 Cisco and/or its affiliates. All rights reserved. 24
Advanced Features 2011 Cisco and/or its affiliates. All rights reserved. 25
TDM SIP Trunks for PSTN Access SIP H.323 CUBE SIP Trunk SBC SP VOIP Servivces Networkbased Media Recording Solution Partner API MediaSense SIP RTP RTP CUBE SIP SBC SP IP Network IVR Integration for Contact Centers CVP vxml Server Media Server SIP CUBE SBC SP IP Network Business to Business Telepresence A CUBE SIP SP IP Network 2011 Cisco and/or its affiliates. All rights reserved. 26 SBC SIP CUBE A
Branch Office Campus Headquarters Telecommuter/SOHO MPLS MPLS Data Voice CUBE VPN Voice Data VPN V Voice Data Voice Voice Class 4/5 Switch SP IP Network TDM-based PSTN TDM IP Trunk Call Path 2011 Cisco and/or its affiliates. All rights reserved. 27
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 Centralized IP PSTN A CUBE Enterprise IP WAN Site-SP Media 2011 Cisco and/or its affiliates. All rights reserved. 28
Branch Office Campus Headquarters Telecommuter/SOHO MPLS MPLS CUBE Data CUBE VPN Voice Data VPN V Voice Data Voice Class 4/5 Switch SP IP Network TDM-based PSTN IP Trunk Call Path 2011 Cisco and/or its affiliates. All rights reserved. 29
Characteristics of Distributed 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 Operational Benefits 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 Challenges Distributed dial-peer management Distributed operational overhead Distributed IP PSTN A CUBE Enterprise IP WAN CUBE CUBE CUBE CUBE Site-SP Media 2011 Cisco and/or its affiliates. All rights reserved. 30
Characteristics of Hybrid 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. Benefits Adaptable to site specific requirements Optimizes BW use on Enterprise WAN Adaptable to regional SP issues Built-in redundancy strategy CUBE Hybrid IP PSTN A CUBE Enterprise IP WAN A CUBE CUBE Site-SP Media 2011 Cisco and/or its affiliates. All rights reserved. 31
Validated with service providers world-wide Tested with 3 rd party PBXs Standards based Cisco Interoperability Portal: www.cisco.com/go/interoperability 2011 Cisco and/or its affiliates. All rights reserved. 32
pamistry@cisco.com 2011 Cisco and/or its affiliates. All rights reserved. 33
Re-purpose your existing Cisco Voice Gateway s as Cisco s Session Border Controller Cisco Unified Border Element (CUBE) SIP/H.323 Trunks Digital/Analog Trunks x 1001 dial-peer voice 1 voip destination-pattern 1... session protocol sipv2 session target ipv4:<pbx_ip_addr> codec g711ulaw x 1001 SIP/H.323 Trunks Buy CUBE License Only CUB E dial-peer voice 2 pots destination-pattern 9T port 0/0/0:23 SIP Trunks SBC SIP SP Change POTS Call Leg to VoIP Call Leg dial-peer voice 1 voip destination-pattern 9T session protocol sipv2 session target ipv4:<pbx_ip_addr> codec g711ulaw dial-peer voice 2 voip destination-pattern 9T session protocol sipv2 session target ipv4:<sip_trunk_provider_ip_addr> codec g711ulaw 2011 Cisco and/or its affiliates. All rights reserved. 34
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) CUBE IP Media Flow-Through Signaling and media terminated by the Cisco Unified Border Element Transcoding and complete IP address hiding require this model CUBE IP Media Flow-Around Signaling and media terminated by the Cisco Unified Border Element Media bypasses the Cisco Unified Border Element 2011 Cisco and/or its affiliates. All rights reserved. 35
x1001 192.168.1.50 192.168.1.1 INVITE /w SDP c= 192.168.1.50 m=audio abc RTP/AVP 0 100 TRYING 180 RINGING 200 OK Internal Network c= 10.1.1.1 m=audio xyz RTP/AVP 0 10.1.1.1 CUBE 20.1.1.1 B2B User Agent External Network INVITE /w SDP c= 20.1.1.1 m=audio xxx RTP/AVP 0 100 TRYING 180 RINGING 200 OK 20.1.1.2 c= 20.1.1.2 m=audio uvw RTP/AVP 0 SBC SIP SP +1 408-526-6855 ACK ACK RTP (Audio) 192.168.1.50 10.1.1.1 20.1.1.1 20.1.1.2 2011 Cisco and/or its affiliates. All rights reserved. 36
Signaling Packets Voice Application Code Dial-Peer Voice Application Code L7 Protocol-independent memory structures holding call state and attributes (CLID, Called #, Codec ) Dial-peer Dial-peer Media Packets DSP (If invoked) SIP/H323 Protocol Stack SIP/H.323 Protocol Stack DTMF xlation Codec Filtering Xcoding Control SIP/H.323 Protocol Stack RTP Library TCP/UDP/TLS Voice stack IOS Infrastructure Physical Interfaces Ingress I/F RTP Library TCP UDP TLS DSP API DSP Hardware RTP Library TCP UDP TLS IOS Infrastructure (ACLs, FW, IPS, VPN) HW LAN/WAN Interfaces IOS Infrastructure Physical Interfaces Egress I/F Signaling Media 2011 Cisco and/or its affiliates. All rights reserved. 37
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 38
SIP CUBE SIP SBC SP IP Network SIP Trunk pointing to CUBE 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 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 39
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) 8 SBC s external IP address that is required for the SP to accept/authenticate calls (Source IP Address for INVITES) 9 Does SP require SIP Trunk registration for each DID? If yes, what is the username & password 10 Does SP require Digest Authentication? If yes, what is the username & password 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 40 Yes 20.1.1.1 No No
1. Turn CUBE Application ON voice service voip mode border-element license capacity 200 allow-connections sip to sip 3. Create a trusted list of IP addresses to prevent toll-fraud voice service voip ip address trusted list ipv4 10.1.1.50 ipv4 20.20.20.20 2. Global settings to meet SP s requirements and SIP Trunk towards SP if needed voice service voip sip early-offer forced header-passing error-passthru midcall-signaling passthru 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 41
Dial-peer static routing table mapping phone numbers to interfaces or IP addresses INBOUND & OUTBOUND CALLS H.323 or SIP CUBE SIP SBC SP IP Network 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 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 42
INBOUND DP FOR CALL FROM CUCM TO CUBE OUTBOUND DP FOR CALLS FROM CUBE TO CUCM dial-peer voice 100 voip description *** LAN side dial-peer *** incoming called-number 9T session protocol sipv2 destination-pattern [2-9]... voice-class sip bind control source gig0/0 voice-class sip bind media source gig0/0 session target ipv4:<cucm_address> codec g711ulaw dtmf-relay rtp-nte CUCM sending 9 + All digits dialed SP will be sending 10 digits inbound 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. 43
INCOMING WAN DIAL-PEER FOR CALLS FROM SP TO CUBE dial-peer voice 200 voip description *** WAN side Incoming DP *** incoming called-number [2-9]... session protocol sipv2 dtmf-relay rtp-nte Catch-all for all SP inbound calls OUTGOING WAN DIAL-PEER FOR CALLS TO SP FROM CUBE dial-peer voice 201 voip description *** WAN side dial-peer_long distance*** translation-profile outgoing Digitstrip_9 destination-pattern 91[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_provider IP address> dtmf-relay rtp-nte codec g729r8 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 2011 Cisco and/or its affiliates. All rights reserved. 44
SIP Provider Requirement Configure SIP Profiles 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 voice class sip-profiles 400 request INVITE sip-header Diversion modify sip:(.*>) sip:4085266855@sip.abc.com> request REINVITE sip-header Diversion modify sip:(.*>) sip:4085266855@sip.abc.com> 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 Apply to Dial-peer or Globally dial-peer voice 4000 voip description Incoming/outgoing SP voice-class sip profiles 400 voice service voip sip sip profiles 1000 See the difference Received: INVITE sip:2000@9.44.44.71:5060 SIP/2.0 User-Agent: Cisco-CUCM8.5 Diversion: <sip:3000@9.44.44.4>;privacy=off; reason=unconditional;screen=yes... m=audio 6001 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000... Sent: INVITE sip:2000@9.44.44.4:5060 SIP/2.0. User-Agent: Cisco CUCM8.5/IOS-15.1-3. Diversion: <sip:4085266855@sip.abc.com>; privacy=off;reason=unconditional;screen=yes. m=audio 32278 RTP/AVP 18 8 101 a=rtpmap:0 PCMU/8000.. 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 45
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. 46
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 2011 Cisco and/or its affiliates. All rights reserved. 47
Is CUBE Active? Is the call matching right Dial-peers? Are we sending the right SIP call to SP based on their requirements? show cube status CUBE-Version : 9.0 SW-Version : 15.2.1T, Platform 2911 HA-Type : none Licensed-Capacity : 200 debug voip ccapi inout Oct 26 18:59:01.146: //-1/66A6B1BF8013/CCAPI cc_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) debug ccsip messages Received: INVITE sip:912025552000@14.128.101.24:5060 SIP/2.0 Date: Wed, 26 Oct 2011 18:59:01 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY From: "Paul Hewson" <sip:1500@10.88.156.166>;tag=90d94d92-6ee4-45aa-9f18-2d09025c1ee4-27352390... 2011 Cisco and/or its affiliates. All rights reserved. 48
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 Some Network Management Tools: - Cisco Unified Operations Manager - Arcana Networks - Solarwinds CUBE can also send SNMP Traps to alert the network management tool of certain events like SIP Trunk failure, link down, high CPU etc.. SNMP Query SNMP Response Network Management Tool H.323 or SIP CUBE SIP SBC SP IP Network 2011 Cisco and/or its affiliates. All rights reserved. 49
Reference 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) Media Resources (DSPs) Voice Quality Trunk Utilization Call Arrival Rate Call Success/Failure SIP retries DSP Availability Transcoding util. MTP utilization Loss, delay, jitter IP SLA 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 CUBE 1.4: CISCO-VOICE-DIAL-CONTROL-MIB, cvcallratemonitor DIAL-CONTROL-MIB, dialctlpeerstatssuccesscalls, dialctlpeerstatsacceptcalls, dialctlpeerstatsfailcalls, dialctlpeerstatsrefusecalls CISCO-SIP-UA-MIB, csipstatserrclient, csipstatserrserver, csipstatsglobalfail CISCO-SIP-UA-MIB, csipstatsretry CISCO-DSP-MGMT-MIB, cdspcardresourceutilization, cdspdspfarmutilobjects CUBE 1.4: CISCO-DSP-MGMT-MIB, cdsptotavailtranscodesess, cdsptotunusedtranscodesess CUBE 1.4: CISCO-DSP-MGMT-MIB, cdsptotavailmtpsess, cdsptotunusedmtpsess CISCO-VOICE-DIAL-CONTROL-MIB, cvvoipcallactivetable 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 2011 Cisco and/or its affiliates. All rights reserved. 50
1. Configure capture profile! create profile ip traffic-export profile TAC mode capture bidirectional incoming access-list 123 outgoing access-list 123!! access-list to filter only SIP messages (port 5060) access-list 123 permit udp any any eq 5060 access-list 123 permit tcp any any eq 5060!! apply to an interface, default memory is 5M interface fa0/0 ip traffic-export apply TAC [size <bytes>] 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: Allows filtering of calling/called numbers when creating the flow graph 2. Capture traffic with these exec (enable) level commands Note: The exec cmds don t appear until a profile has been configured router#traffic-export interface fa0/0 clear router#traffic-export interface fa0/0 start <capture the problem> router#traffic-export interface fa0/0 stop IP Traffic Capture: http://www.cisco.com/en/us/docs/ios/12_4t/12_4t11/ht_rawip.html 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 51
Shipping Planned 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 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 52
Visit http://www.cisco.com/go/cube for more information on CUBE Email ask-cube@external.cisco.com for any CUBE related questions Visit http://cisco.thesipschool.com/webinar To get access to the Webinar To send any comments and feedback For discounted SSCA SIP training To get a copy of a working config Email cisco@thesipschool.com for your discount voucher. This offer is open to webinar attendees for a limited time only. 2011 Cisco and/or its affiliates. All rights reserved. 53