Although this is a Cisco networks dedicated blog, I decided to start a series of tutorial posts about a general technology which is not directly related to Cisco but it is a field in which Cisco is again a major player. This is IP Telephony and Voice over IP (VoIP). The two terms, IP Telephony and VoIP, are related around the same concept but in my opinion they are not exactly the same thing. Many people refer to these two terms interchangeably but they are not exactly the same. So, before moving on lets clarify the difference between IP Telephony and VoIP.

IP Telephony Vs VoIP

IP telephony has to do mainly with digital telephony systems (LAN based IP PBX systems) which use the IP protocol entirely for voice communication. All components of the IP telephony system use digitized voice which is transferred as IP packets through an IP network (usually the LAN network). The telephone handsets (VoIP phones) translate the analogue voice signal into digital voice (binary voice) which is transferred as IP packets from one phone to another. The call control system is usually a software based (softswitch) server which handles all call signaling, call routing, IP phone management etc, again using IP protocol for transport. So think about IP telephony as a bigger concept.

VoIP on the other hand is a subset of IP Telephony. Basically, VoIP is the technology which is used by IP Telephony as the vehicle to transport phone calls. VoIP is the technology in which the analogue voice signal is digitized (analog to digital conversion) and becomes binary numbers in order to be transferred by the IP protocol. VoIP is the basis for the implementation and functionality of an IP Telephony system. VoIP can also be used by legacy TDM based PBX systems to transport voice calls over an IP WAN network or even over the Internet. Special voice gateways are used to connect to the legacy PBX telephone system on one end and to the IP network on the other end in order to translate the TDM voice stream into IP voice packets.

So to summarize, IP Telephony is the overall concept of the modern form of voice communication which harnesses the power and features of VoIP technology in order to offer the overall experience of communicating effectively and with lots of extra features.

Now that we described the difference between IP Telephony and VoIP, let’s see more details about the two concepts:

1. More details about Voice over IP

The term VoIP or Voice over IP refers to the transfer of voice packets over networks based on Internet technology and, more specifically, the IP Protocol. The IP protocol on which the whole Internet is based on was created to implement the transmission of data in the form of data packets. This means that when a data document is transferred over the Internet is cut into small IP packets and sent over the network. When the document reaches its destination, the packets are joined again thus recreating the original document. The same logic applies if the data transferred corresponds to a voice conversation. The voice is digitized, chopped into packets of data transferred over the network via the IP protocol. At the destination the packets are rejoined to recreate the voice stream. Here we should make clear that VoIP refers to the transfer of voice over any IP network. Such a network is the Internet of course, but when considering VoIP it does not necessarily mean that we carry voice over the Internet only. It can be any IP-based network (such as a private corporate WAN network).

2. Packet based (IP Telephony) Vs Circuit Switched Telephone Systems

IP Telephony systems are those using entirely IP packets for voice communication, as explained before. In contrast to packet switched telephone systems (those based on IP protocol), conventional telephone systems apply the logic of direct connection between the two communicating voice parties through a dedicated circuit reserved exclusively for each contact. Thus the term Circuit switched telephone systems. In packet switched systems, however, the same communication line can be used to simultaneously pass different kinds of packets. Thus, the voice packets of one or more conversations may travel through the same route as other packets transferring data, video etc. This is the main difference between traditional telephony which is implemented to the public switched telephone network (PSTN) and telephony implementation on IP networks (or more generally to packet switched networks).

More on IP telephony and VoIP on a future post. Stay tuned.

The Cisco CallManager Express is a product under the Unified Communications Products suite of Cisco. In the past it was known as CCME (Cisco Call Manager Express) but now the new name is Cisco Unified Communications Manager Express.

It is an IP Telephony system (IP PBX) for small to medium size businesses of up to 250 IP phones capacity. Basically, a CallManager Express system is a normal Cisco Integrated Services Router (models 1800, 2800, 3800) which has the CallManager software installed on the router’s flash memory. The router hosting the callmanager system can work also as normal Internet Border router or as WAN Router connecting to other enterprise sites. The CallManager software provides call control and IP telephony functionality to internal IP phones. For connectivity to the PSTN network, voice interface cards can be installed on the CallManager router (such as voice BRI, PRI etc).

In this post we will describe three common deployment models for a CallManager Express system as it is implemented in real world enterprise environments. The three deployment models are Single Site, Multi Site with Distributed Call Processing, and Multi Site with Centralized Call Processing.

Single Site Deployment Model

This is the most common scenario and is usually found in smaller business environments. See the picture below:

Basically a single CallManager Router system is installed which usually provides also the Internet connectivity for the office. If you are a little flexible with your budget, I would recommend installing a firewall in front of the CallManager router to protect it from Internet attacks. All IP Telephony services are provided on the LAN network for internal IP Voice communication. Any call beyond the LAN uses the PSTN network. There are no telephony services provided over an IP WAN.

Characteristics and Best Practices

• Maximum of 250 IP phones can be supported.
• Arrange your internal switch to have two VLANs (one for Voice and one for Data Traffic).
• Use G.711 codec for all IP phone calls on the LAN (80kbps bandwidth per call) for best voice quality.
• You can also install a Voice Mail card on the router to offer voice mail functionality to users.
• Use appropriate Voice Interface Cards on router for PSTN connectivity.
• You can use dual router for redundancy if needed.
• Try to avoid connecting the CallManager router directly to the Internet. Use a firewall as border internet device.
• Dial Plan is simplified. If DID (Direct Inward Dialing) is required, then arrange your dial plan and internal IP phone numbering accordingly.

Multi Site with Distributed Call Processing Model

The multi site model consists of two or more independent sites, each with its own CallManager Express system installed (distributed call processing) as shown in the figure below.

All the sites are interconnected over an IP WAN which can be offered via Leased Lines, Frame Relay, ATM, MPLS Layer2/3 VPN, IPSEC VPN over the Internet etc. All sites have also local PSTN connectivity which can serve as backup to the WAN telephony connectivity or for local inbound and outbound PSTN calls. 

Characteristics and Best Practices

• PSTN Call cost savings when using the IP WAN for calls between sites.
• Bypass long distance call charges (toll bypass) by routing calls through remote site callmanager systems which are closer to the PSTN number dialed. For example, you have one site in New York and one in California. Calls from NY to California can be routed over the IP WAN towards Cal office and then get out to PSTN from the Cal office.
• No loss in functionality for IP WAN failure because there are independent Call processing units in each site.
• Recommended to install a GateKeeper (Cisco IOS gatekeeper) to provide call admission control and dial-plan resolution.
• Use G.729 or G.723 codec for IP calls over the WAN to save bandwidth.
• Use a SIP proxy if you are using SIP instead of H323.

Multi Site with Centralized Call Processing Model

This implementation scenario is suitable for an Enterprise that has a big central office with several smaller branches. One centralized CallManager system can be installed to the Central Site offering call processing and IP Telephony service to both the central site as well as to the remote small branches. The remote branches are equipped only with IP phones (no callmanager system). This is shown in the figure below:

The remote branches are connected to the central site over an IP WAN or even using IPSEC VPN over the Internet. The IP phones located to the remote sites should have IP connectivity to the Central CallManager system, where they are registered. PSTN access is offered only on the Central Site. That is, the call of a remote branch user calling a PSTN number is routed over the WAN to the Central Site and then routed out to the PSTN.

Characteristics and Best Practices

• Cost savings in hardware (only one central callmanager express)
• Easier to manage (centralized management for all IP phones).
• Disadvantage in redundancy since remote sites depend heavily on the availability of WAN lines.
• Use G.729 or G.723 for inter-site calls.
• Savings in PSTN line costs.
• Remote sites must not have many IP phones (10-20 maximum).

All the above deployment models apply also for the other Cisco IP Telephony solution, the Cisco Unified Communications Manager system which is for bigger implementations compared to the Express solution.

Before showing you how to configure a basic IP phone on Cisco CallManager Express (CCME), you need first to understand the concepts of ephone and ephone-dn.

In CCME, “ephone” (short for Ethernet Phone) refers to the physical IP phone device, and is configured with the Ethernet MAC address of the IP phone. The MAC address of the IP phone uniquely identifies the device on the network and is found on a sticker on the underside of the IP phone or from the phone’s shipping carton label.

The ephone directory number (ephone-dn) refers to the phone lines that are associated with the ephone device. The ephone-dn parameter basically configures the telephone device number. Also, the ephone-dn can use the “dual-line” option which will allow the IP phone to handle two simultaneous calls. The dual-line option also provides a way for the phone to support call waiting, conferencing, call transfer with consultation etc.

Configuration:

In the following configuration we will configure a Cisco 7960 IP phone with two directory numbers 2100 and 2200 on the first two line buttons of the telephone.

CCME#show running-config

!Tell the router that the phone firmware P00303020214.bin is located in Flash

tftp-server flash:P00303020214.bin

!Configure the IP Telephony DHCP range
ip dhcp pool Voice
network 10.1.1.0 255.255.255.0
default-router 10.1.1.1
option 150 ip 10.1.1.1

interface FastEthernet0/0
ip address 10.1.1.1 255.255.255.0

telephony-service
ip source-address 10.1.1.1
load 7960-7940 P00303020214
max-ephones 24
max-dn 24
create cnf-files

!Configure the first directory number 2100
ephone-dn 10 dual-line
number 2100

!Configure the second directory number 2200
ephone-dn 11 dual-line
number 2200

!Configure the 7960 phone and assign ephone-dn numbers to buttons 1 and 2
ephone 1
mac-address 000d.aa45.3f6e
type 7960
button 1:10 2:11

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
!