For many years now, telephony voice services for businesses and enterprises have been provided by using legacy PBX systems connected to the Public Switched Telephone Network (PSTN) using TDM connections (T1/E1 ISDN PRI lines or BRI or analog lines). This is shown on the figure below:

Newer telephony systems adopted the IP technology on the internal LAN, but they still used TDM connectivity (ISDN PRI/BRI and analog lines) to connect to the legacy PSTN network as shown below:

The newest trend is to go all-IP using SIP TRUNKING to connect your business office to the Telephony Service Provider network. A SIP Trunk allows the company to replace the traditional TDM fixed lines (PRI, BRI etc) with just a normal IP connection towards the service provider. This solution offers significant cost savings to the enterprise as you avoid costly BRI/PRI lines. Also, voice/data traffic can be converged on a single IP connection. This scenario is shown below:

The Cisco Call Manager Express product can be used as the telephony SIP trunk gateway between the local IP telephony network and the IP Telephony Service Provider. Calls from and to PSTN will be handled by a SIP PROXY server located in the Service Provider network.

 A sample Call Manager Express configuration for SIP trunking is shown below (a snippet of the complete configuration is shown):

voice service voip
   allow-connections sip to sip
   sip
       registrar server expires max 3600 min 3600
       localhost dns:mycompany.test.com

voice class codec 1
 codec preference 1 g711ulaw

!— Inbound Translation Rule
!—  for Auto Attendant pilot number “500″
voice translation-rule 1
 rule 1 /5552222100/ /500/

voice translation-profile AutoAttendant
!— Applied to the inbound dial-peers for AA
 translate called 1

!— SIP Trunk Configuration —
dial-peer voice 1 voip
 description **Incoming Call from SIP Trunk**
 translation-profile incoming AutoAttendant
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 incoming called-number .%
 dtmf-relay rtp-nte
 no vad

dial-peer voice 2 voip
 description **Outgoing Call to SIP Trunk**
  destination-pattern 9……….
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad

dial-peer voice 3 voip
 description **International Outgoing Call to SIP Trunk**
  destination-pattern 9011T
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad

!— SIP UA Configuration —
sip-ua
 authentication username 5552222100 password 075A701E1D5E415447425B
 no remote-party-id
 retry invite 2
 retry register 10
 retry options 0
 timers connect 100
 registrar dns: mycompany.test.com expires 3600
 sip-server dns: mycompany.test.com
  host-registrar
!

The Cisco Unified CallManager Express (CME) solution not only has the benefit of voice-data integration on a single platform, but offers also flexible deployment options. The Cisco CME on its basic form consists of a router on which the callmanager software is installed, plus several telephony devices. The CME router acts as a gateway between the Public Switched Telephone Network (PSTN) and your local IP telephony network. IP Phones or other legacy telephony devices can be connected on the Call Manager Express router (either directly using FXS ports, or on the local LAN switch). The figure below shows a basic small-office/medium-office CME network topology (figure is from Cisco):

The typical CME deployment above uses a single callmanager router with few legacy telephony devices (normal telephones and a Fax machine) connected directly on the router itself (on FXS ports), plus few IP Phones connected on the local LAN switch. All these phones are controlled by the CME router.

The Cisco CME software uses the following basic building blocks:
 

• Ephone: This is configured in software (using IOS commands on the router) and represents a physical telephone. The MAC address of each physical phone is configured using the ephone configuration commands.
• Directory Number: This is again a software concept that represents the line that connects a voice channel to a phone. A directory number represents a virtual voice port in the Cisco Unified CME system.

Call Manager Express Call Handling Modes

Before deploying a Call Manager Express system you must decide how the system will handle calls. There are three call handling models: PBX model, KeySwitch model or Hybrid model.

PBX Model:
This is the simplest and most popular call manager mode of operation. Each internal telephone has its own unique directory number (extension number) as shown in the diagram below.

Incoming PSTN calls are usually routed by the CME router to a central receptionist (or auto-attendant) which then delivers the calls to the appropriate requested extension number. There is also the option of having Direct Inward Dialing (DID) lines towards the PSTN which allows incoming PSTN calls to be directly routed to specific internal extensions. An example of DID is when calls coming to number 555-838-1001 will be routed directly to Extension 1001, calls coming to number 555-838-1002 will be routed to Extension 1002 etc.

It is recommended for this model that you configure directory numbers as dual-lines so that each button that appears on an IP phone can handle two concurrent calls. Dual-line directory numbers enable your configuration to support call waiting, call transfer with consultation, and three-party conferencing (G.711 only).

Keyswitch Model:
In this model there is no central receptionist telephone. Rather, all telephones have an identical configuration in which each phone is able to answer any incoming PSTN call on any line. An example is shown below:

The keyswitch model is configured by creating a set of directory numbers (Extension numbers) that correspond one-to-one with your PSTN lines. Then you configure your PSTN ports to route incoming calls to those directory numbers. When an incoming PSTN call arrives (e.g on Extension 1001), then ALL telephones will ring on line 1001. Any user can then pick-up the ringing line by just pressing the button corresponding to that line.

 Hybrid Model:
In this model, each IP phone can have both PBX and Keyswitch configurations. Each telephone can have unique extension numbers (PBX model) and also shared lines numbers (keyswitch model).

A hybrid model is shown above. Extension numbers 1001, 1002, 1003 are shared lines, and Extensions 1004, 1005, 1006 are unique private numbers for each user.

If you are preparing for your CCNA certification then you might have been in the dilemma which path you should take for becoming a Cisco CCNA certified professional. As we know, Cisco is offering two “routes” for obtaining a CCNA.

The “quick route” requires you to take just a single composite exam (640-802 CCNA) which covers topics from Interconnecting Cisco Networking Devices parts 1 and 2 (ICND1 and ICND2) courses. The 640-802 exam is 90 minutes duration with 50 to 60 questions.

The second route to CCNA requires candidates to take two exams:

• 640-822 ICND1 (90 minutes duration with 50 to 60 questions)
• 640-816 ICND2 (75 minutes duration with 45 to 55 questions)

The cost of the two certification paths is the same. That is, the cost of taking the single composite exam (640-802) is the same as taking the two-exam option.

So, which path is the best? This really depends on each individual candidate. In my opinion, people that are already working in the networking field and have some experience with routers, switches, cabling etc, its better to take the single-exam path to avoid the hassle and stress of the two-exam option. On the other hand, people that are just starting now in networking are better off to go for the two-exam option. This option has also the advantage of giving you an extra certification, since you will get the CCENT (Cisco Certified Entry Networking Technician) certification by passing the 640-822 ICND1 exam.