Voice over IP Fundamentals

CHAPTER 3+5 Basic Telephony Signaling PSTN Services

Loop Start and Ground Start Signaling (Tip and Ring)• Tip – Ground side• Ring – Power side (-48V)

Trunking the PSTN to CME:

• Foreign Exchange Station (FXS) ports typically connect analog phones, fax machines, and modems to the CME router• Foreign Exchange Office (FXO) ports normally connect the PSTN to the CME router, or PBX system• Earth and Magneto (E&M) or Ear and Mouth connects from the PSTN directly to a PBX system

Digital Connections:

• Channel Associated Signaling (CAS) uses robbed bits from the voice data flow for signaling and control functions. Does affect the voice quality slightly (in-band-signaling)• Common Channel Signaling (CCS) uses a separate channel for all signaling and control functions (out-of-band signaling)

ISDN Features:

• ISDN Provides Circuit-based communications• New services extended to users• ISDN has two types, Basic Rate Interface (BRI) and Primary Rate Interface (PRI)• ISDN includes Direct-Inward-Dial (DID) and Direct-Outward-Dial (DOD)• ISDN can add additional channels• ISDN can transmit voice or data• ISDN uses a separate signaling channel• ISDN is compatible with SS7• ISDN enables the creation of VPNs

ISDN Interfaces:

• B Channel: 64 kbps No signaling information G.711 CODEC

• D Channel: All signaling information 9.6 kbps (64 kbps line)

ISDN Interfaces 1:

ISDN Interfaces 2:

ISDN Protocols:

• Q.921: Layer 2 Error-free and Secure Communications Link Access Procedures on D-Channel

(LAPD)• Q.931:

Layer 3 Call establishment Call Control Messages

PSTN Features:

• Features: Customer-originated trace Automatic callback Automatic recall Display features Calling number blocking Call screening Voice messaging Fax messaging Virtual Private Voice Networks Centrex services Call center services

Cisco Unified Communications Manager Express Platform Limitations:

Cisco Unified CME Features:

• System: Account codes and Call Detail Record (CDR) field entry Callback busy subscriber and camp-on Per-phone call coverage rules Call hold and retrieve Call park: personal and directed Call transfer and park recall Call park assign to extension Call pickup directed Call pickup local group Call pickup explicit group Call transfer: consultative and blind Call waiting Computer Telephony Integration (CTI) with Microsoft CRM and Outlook using Cisco IOS Software Telephony Services Provider (TSP) E911 with two emergency location numbers per zone, unlimited zones per site

Cisco Unified CME Features continued:

• System: Eight –party impromptu conferencing Directory services using XML Hunt-group dynamic login and logout Hunt-groups statistics: daily and hourly Intercom Meet-me conferencing (32 party) Music on Hold (MoH): internal or external source Night service bell or call forwarding Overlay extensions for enhanced call coverage Called-name display for overlay extensions Paging: internal through IP phones or to external system Per-call caller ID blocking Secondary dial tone Standards-based network call transfer and call forwarding using H.450 Additional system speed-dial option through XML service


• System: Time-of-day and day-of-week call blocking Customizable called-name display Support of SRST fallback service phone auto-registration Basic automatic call distribution (B-ACD) (three queues) with auto-attendant and call statistics Display of number of calls in queue on IP Phone Agent login and logout of B-ACD hunt group Integration with Cisco Unified Contact Center Express 5.0 for advanced call center features with support for up to 50 agents, agent supervisors, call recording, silent monitoring, and reporting features. Secure Real-Time Protocol (SRTP), providing media encryption for calls on the IP network Secure Voice IP Phone certificate authentication and provisioning plus secure device signaling using Transport layer Security (TLS)


• Phone Features: Maximum 250 phones per system Up to 34 line appearances per phone Attendant console functions using Cisco Unified IP Phone Expansion Module 7914 Fast transfer: blind or consult Busy lamp Silent ringing options Automatic line selection for outbound calls Call forward on busy, no answer, and all (internal or external Call-forward-all restriction control Do not disturb (DND) Feature ring with DND set IP Phone display of DND state Dial-plan pattern load on SIP phones Diversion of calls directly to voice mail

Cisco Unified CME Features:

• Phone Features: Customization of softkeys Enable and disable call-waiting notification per line Call waiting with overlay directory number Call-waiting ring Dual or 8 call line appearances per button After-hours toll-bar override Auto-answer with headset European date formats Hook flash pass-through across analog PSTN trunks Idle URL: periodically push messages or graphics on IP Phones Last-number redial Live record to Cisco Unity Express mailbox Local name directory lookup On-hook dialing Station speed dial with configuration changes from IP Phone


• Phone Features: System speed dial for 10,000 numbers Silent and feature ring options SIP-based line-side subscribe, providing basic presence of phone status Transfer to voice mail softkey Call barge with privacy on shared lines Access features using softkeys or feature access codes Remote teleworker IP phone support Dynamic hunt-group join or leave Support for analog phones using Cisco ATA 186 Analog Telephone Adapter or Cisco VG224 Analog Phone Gateway in SCCP mode Support for fax machines on Foreign Exchange Station (FXS) ports or ATA using H.323, SCCP, or SIP XML application services on Cisco Unified IP display phones


• Phone Features: Station-to-station video with voice using Cisco Unified Video Advantage or Cisco Unified IP Phone 7985G endpoints Extension mobility within the single site Wideband audio (G.722) and iLBC codec


• Trunk Features: Analog Foreign Exchange Office (FXO) loop and ground start Ear and Mouth (E&M) Basic Rate Interface (BRI) and Primary Rate Interface (PRI) support (NI2, 4ESS, 5ESS, EuroISDN, DMS100, and DMS250) and several other switch types currently supported in Cisco IOS software Called ID name and number Automatic number identification (ANI) Digital trunk support Direct Inward Dialing (DID) Direct outward dialing E1 R2 support Dedicated trunk mapping to phone button H.323 trunks with H.450 support H450.12 automatic detection of H.450 support for remote H.323 endpoints

Virtual Trunking Protocol:

• Switches exchange VLAN information automatically• VTP Domain Names and Passwords are case sensitive• VTP Modes are server, client or transparent• VTP Server allows the creation or deletion of VLAN’s throughout system. VLAN information is saved in switch memory• VTP Client allows only the acceptance of VLAN’s from the server. Information is not stored in memory.• VTP Transparent mode allows the creation or deletion of VLAN’s of local significance only. VLAN information is stored in switch memory. Will pass VTP information to other switches within the same domain.

Virtual Trunking Protocol continued:

• Each VTP change increased VTP revision number. Highest revision number is distributed through out system• Configuration:

Switch(config)#vtp mode serverSwitch(config)#vtp domain PHONE_NETWORKSwitch(config)#vtp password VOICEPA55Switch(config)#endSwitch#

Virtual Trunking Protocol continued:

• Interface trunking modes: Dynamic desirable: Cisco default. Will become trunk depending on mode and device attached. Dynamic auto: Will become a trunk depending on mode and device attached but will not actively try to negotiate a trunk link. Trunk: Will be in trunk mode but will negotiate with either dynamic auto, dynamic desirable using Dynamic Trunking Protocol (DTP). Access: Not in a trunk mode. Gives access to one Data VLAN and one Voice VLAN only. Nonegotiate: Disables DTP messages on interface


• Trunk Features: H.323-to-H.323 hairpin call routing for non-H.450-compliant H.323 endpoints SIP trunks and RFC 2833 support Transcoding with G.711, G.729a, and iLBC ISDN Q.SIG supplementary services of basic calls, including call forwarding busy, no answer, all; calling name and line identification (CLIP and CNIP); connected line and name identification (COLP and CONP); message waiting indicator (MWI) and message center support; MWI passthrough QSIG-to-TDM voice mail


• Voice Mail Features: Integrated voice mail and auto-attendant solution with Cisco Unity Express Integration with Cisco Unity Voice Mail and Cisco Unity Unified Messaging, or third-party voice mail integration (H.323, SIP or dual-tone multifrequency [DTMF])

CME Integration:

Cisco Licensing for Cisco Unified CME:

• IOS license: The CME router must be licensed to run a version of Cisco IOS that is capable of supporting the CME software• Feature license: The feature license grants the CME router the ability to support a specific number of IP Phones.• Phone user licenses: each phone must have a license

Cisco Unified CME Models:

• PBX Model


• Keyswitch Model


• Hybrid Model

Installing Cisco Unified CME:

• Router Router(config)#copy tftp://172.16.2.5/c2800nm-adventerprisek9_ivs-mz.124-24.T2.bin flash:


• CME Basic files GUI files XML template files MOH files Script files Miscellaneous files


• CME Installation File


• Extract CME Installation File Router#archive tar /xtract tftp://172.16.2.5/cme-full-7.1.0.1.tar flash:

Loading cme-full-7.1.0.1.tar from 172.16.2.5 (via fastEthernet0/0):!bacdprompts/ (directory) 0 (bytes)extracting bacdprompts/app-b-acd-2.1.2.2-ReadMe.txt (18836 bytes) extracting bacdprompts/app-b-acd-2.1.2.2.tcl (24985 bytes)

IP Phone Boot:

1. IP Phone connects to switchport2. Switchport senses and supplies PoE3. Via CDP phone receives voice VLAN information4. Phone sends DHCP request on voice VLAN and receives IP address, Mask and default-Gateway5. Once addressed the phone contacts TFTP server (Option 150) and downloads configuration files6. Phone contacts first call processing center (CME Router) and registers. If unable to contact will contact additional centers as listed in configuration


• Verify download:

CME_Voice#dir flash: CME_Voice#dir flash:/phone CME_Voice#dir flash:/phone/7940-7960

CME_Voice#dir flash:/phone/7940-7960Directory of flash:/phone/7940-7960/

99 -rw- 129824 Nov 17 2009 21:18:18 -04:00 P00308000500.bin 100 -rw- 458 Nov 17 2009 21:18:18 -04:00 P00308000500.loads 101 -rw- 705536 Nov 17 2009 21:18:22 -04:00 P00308000500.sb2 102 -rw- 130228 Nov 17 2009 21:18:24 -04:00 P00308000500.sbn

256491520 bytes total (92770304 bytes free)CME_Voice#

Create e-phone:

• Maximum Phones and Numbers:

CME_Voice(config)#telephony-service CME_Voice(config-telephony)#max-ephones 24 CME_Voice(config-telephony)#max-dn 48 CME_Voice(config-telephony)#exit

Create e-phone:

• TFTP services for Firmware loads:

CME_Voice(config)#tftp-server flash:/phone/7940-7960/ P00308000500.sbn alias P00308000500.sbn CME_Voice(config)#telephony-service CME_Voice(config-telephony)#load 7960-7940 P00308000500.loads

Create e-phone:

• Source address:

CME_Voice(config)#interface lo0 CME_Voice(config-if)#ip address 1.0.0.1 255.255.255.255 CME_Voice(config)#telephony-service CME_Voice(config-telephony)#ip source-address 1.0.0.1

Create e-phone:

• Create CNF files:

CME_Voice(config)#telephony-service CME_Voice(config-telephony)#create cnf-files

Create e-phone:

• Verify CNF files:

CME_Voice#show telephony-service tftp-bindings

CME_Voice#sh telephony-service tftp-bindingstftp-server system:/its/united_states/7960-tones.xml alias United_States/7960-tones.xmltftp-server system:/its/united_states/7960-font.xml alias English_United_States/7960-font.xmltftp-server system:/its/vrf1/XMLDefault7940.cnf.xml alias SEP000F24A97457.cnf.xmltftp-server system:/its/vrf1/XMLDefault7940.cnf.xml alias SEP0012435D4217.cnf.xmltftp-server system:/its/vrf1/XMLDefault7940.cnf.xml alias SEP000AB776A988.cnf.xml

Create e-phone:

• Inspect Generic Configuration file:

CME_Voice#more system:/its/vrf1/XMLDefault7940.cnf.xml

CME_Voice#more system:/its/vrf1/XMLDefault7940.cnf.xml<device><versionStamp>{7940 Feb 16 2010 14:23:20}</versionStamp><devicePool><callManagerGroup><members><member priority="0"><callManager><ports><ethernetPhonePort>2000</ethernetPhonePort></ports><processNodeName>10.15.0.1</processNodeName>

Create e-phone:

• Phone has Registered:

End of Chapter 3 & 5

Voice over IP Fundamentals

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