Voice Data, after the compression, is transferred via the network. For this continuous data stream of compressed voice signal, the conversation is divided into small data packets, before they are sent to the network. Then, these packets arrive at the specific destination network “nodes”, so-called routers, which direct the IP-packets to their final point, maybe via different paths.

The voice packets, before they are sent to the network line, they are first temporarily stored in a memory location called the buffer. At the destination end, voice packets pass through a digital to analog converter so that to be transformed back to human speech. For example, the voice packets delivery can be compared with normal mailing post, which addresses, mails and delivers individual packets of data.

To transfer data using VoIP we currently use the so-called Internet protocol version 4 (IPv4). The next version – IPv6 – was specified in the mid 90-ies, but the need to expand its use has not been taken seriously until the last few years. Unfortunately, the IP protocol works on the principle of “Best Effort” and thus it does not guarantee 100% packet delivery. This is the main reason that we do not always have good quality in VoIP. The newest IP Protocol version 6 offers the so-called “quality of service” under which the voice data transmission will have better quality.

In order to establish a connection between a VoIP telephone system and the traditional PSTN telephone network, we need to use the so-called Gateways. These Gateways are connected to the IP Data network as well as with the PSTN telephone network and transmit requests in both directions. In this case, IP-packets from one side are converted into digital voice stream on the PSTN side.

Since everything now runs over IP (our computer data, voice, video etc) we can have the integration of different types of data on to a single IP network. This concept is usually called “network convergence”. The meaning of convergence is that we have one common network (the IP Network) which transmits all kinds of information – voice, data, video, text and images.

Usually there is a single CUCME system in each LAN network, with several IP phones connected on the LAN switches. An enterprise with several sites connected over a private IP WAN network can establish full IP voice communications between sites by configuring H323 communication between each CUCME router. A simple example with a two-node topology is shown below.

CME-A node has local IP phones with numbering 500x and a WAN IP address of 1.1.1.1. On the other site, CME-B has local IP phones with numbering 600x and a WAN IP address of 2.2.2.2. By establishing H323 voip communication over the WAN (between 1.1.1.1 and 2.2.2.2) we can have full IP telephony conversations between the IP phones of both sites.

CAUTION: Because the actual VoIP RTP traffic communication between site A and site B will be running from one IP phone to another IP phone, there must be full IP routing established between the IP phone subnets.
The CUCME configuration to establish H323 between the two sites is shown below:

CME-A

CME-A#show running-config
dial-peer voice 6000 voip
destination-pattern 60..
session target ipv4:2.2.2.2
dtmf-relay h245-alphanumeric
codec g729r8

CME-B

CME-B#show running-config
dial-peer voice 5000 voip
destination-pattern 50..
session target ipv4:1.1.1.1
dtmf-relay h245-alphanumeric
codec g729r8

The dial-peer configuration on CME-A tells the system that in order to reach the destination pattern 60xx the session will be established with IP address 2.2.2.2 (i.e CME-B). The inverse applies for CME-B.

Note: Make sure to select one of the high compression codecs ( such as g729, g723) in order to save bandwidth for voice calls over the WAN network. Each VoIP conversation using a high compression codec (g729, g723) will use significantly less bandwidth compared with the traditional G711 codec.

Is IP Telephony Implemented Easily ?

Over time, most companies have acquired the expertise to implement IP Telephony solutions, either on existing corporate networks or from scratch. The main advantage to implementing VoIP applications is that they rely on network infrastructure which can be expanded gradually, depending on the needs of the business. Additionally, complimentary applications have been matured as well, such as call management software, so that the implementation of solutions and their use becomes more straightforward.

What happens in terms of voice quality ?

Traditionally the main problem of telephony on IP networks has been the quality of the voice. Since the same network carries different data packets (documents, other voice conversations etc.) we cannot always ensure that the packets carrying the voice conversation will all get together and on time at the other end in order to carry a real-time discussion. When you transfer a document, a web page, an email etc, we don’t care so much if one packet is delayed 1-2 seconds. In voice conversation however, delay works negatively on the quality of the voice. A solution to this problem would be the usage of high-capacity lines, combined with powerful routing equipment (eg routers and large enough switches). However they cost money. A better solution is the implementation of prioritization of voice packets with respect to other data. Gradually, as the cost of equipment and services drops, the quality of VoIP will be better and better. Finally, we must not forget that using certain technologies (e.g voice compression), we can increase the efficiency of communication lines and with appropriate settings in routers we can commit certain capacity from the network for voice communication. With that, voice transmission will be conducted as much as possible in real time, without delays and distortion.

Do we need special telephone handsets ?

There are special telephone handsets designed for VoIP communication that harness the potential of this technology. Such devices are available from most international manufacturers of telephony products as well as from third party manufacturers involved in related VoIP solutions. It is worth mentioning that using special equipment we can still use normal telephone devices. Alternatively, a company may consider the option of softphones. A softphone is essentially telephony software that is installed on a laptop or desktop computer and offer all the functionality of an IP telephone without the need for a hardware telephone device. Of course, the use of softphones depends upon the existence of a computer. Although the first softphones presented had poor voice quality and a great network load, now the technology is fairly mature and operational.

Can an IP Telephony System be connected to the public telephone network

There are special voice gateways which can connect an IP Telephony system with the public switched telephone network (PSTN) or other telephone networks. Using the voice gateway, a VoIP phone can call a legacy telephone line phone on the public telephone network and vice versa with no problems. Basically the voice gateway translates the IP packets from the IP Telephone system into TDM voice to be transmitted over the legacy PSTN network. Generally, regardless of the infrastructure that the IP Telephony system uses to carry out the conversation, ultimately it is a private telephone network, such as those implemented in corporate call centers, which is transparent to the public telephone network.

What are the benefits of IP Telephony and VoIP

The main advantages of VoIP and IP Telephony in general include:

• Single network infrastructure for data and telephony. Since the same infrastructure (communication lines and equipment) serve voice traffic and data traffic, we have significant economies of scale. Also, we achieve better management of telecommunications infrastructure.
• Maximum use of telecommunications infrastructure. The packet switched networks (e.g IP Networks) make better use of their bandwidth capacity in comparison with traditional circuit switched telephone networks since the line is not fully occupied for each call conversation therefore it can carry various data packets in addition to voice.
• Improved communication for remote workers. The use of IP telephony does not require the user to have a physical presence in the enterprise environment. If the user has an IP connection, he/she can take advantage of the features and functions of the enterprise telephone system, regardless of where the user is located.
• New services are introduced. The usage of a single infrastructure for both data and voice allows for the development of a new generation of services such as unified messaging that can contribute significantly to productivity growth.

Why companies are interested for IP Telephony

Since almost all companies have access to the Internet, they have already implemented their corporate networks over the IP protocol. Thus, they are given a first class opportunity to utilize the IP network infrastructure, which includes, in addition to the communication lines, other equipment such as routers, switches, etc. This IP network infrastructure can be used for telephony as well. Even if the IP telephony system is confined within the enterprise, the benefits are significant. When a company uses leased circuits to connect remote branches, the use of these circuits for both IP telephony and data connectivity provides substantial benefits and cost savings to the company.

Is IP Telephony the most economical solution for voice communication

Like any technology infrastructure investment, usage of VoIP and IP Telephony should be treated as a medium to long term business. According to studies, the use of packet switched networks for voice telephony is more economical than the networks that occupy the whole communication line for each conversation. And when we can serve phone calls through our corporate IP network – which in some cases is extended to different parts of the city, other cities or other countries – we certainly save money by not using the public telephone network. When routing phone calls over our own private IP network from New York to Los Angeles and the destination call is a PSTN number in Los Angeles, the call will be charged as local in Los Angeles (it will be routed from our voice gateway in Los Angeles to the PSTN). This is an example of a toll bypass cost saving. Companies should however consider the costs for the implementation of the IP telephony infrastructure, occurring in the increased bandwidth capacity to accommodate also voice traffic, in the extra equipment (e.g IP telephones), the additional software needed, etc. Overall, however, in medium to long term, telephony over IP networks has proved to be much more economical than traditional telephony solutions.

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.

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.

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

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 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.

The CME software can be downloaded from http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp (login required) and you can install it on the router flash. You need to download the CME software (comes as a single .zip compressed file) that is appropriate for the specific router IOS image you are intended to use. The ZIP file contains several individual files and several TAR archives. The individual files can be copied to flash using the regular “copy tftp flash” command and the TAR archives can be copied and extracted to flash using the “archive tar” command (more details later).

The recommended files that you need to install are the following:
 

•  Basic Files: This is a TAR archive containing the basic files you need to run the Cisco Unified Call Manager express. This archive contains also the phone firmware files required, although additional individual phone firmware files may be needed sometimes. The filename for this tar archive is “cme-basic-x.x.x.tar“.
• GUI Files: This is again a TAR archive containing only the files of the GUI management tool, which is a mouse-driven interface for provisioning phones and for general CME management after basic installation is complete. The filename for this tar archive is “cme-gui-x.x.x.tar“.
• Phone Firmware Files: Although the required phone firmware files are included in the Basic tar archive, you may need to add phone firmware files to support individual phone models that are not included in the basic package. Each firmware file is specific for each phone model and for the protocol it uses (i.e SCCP or SIP protocol). By default, new IP phones are shipped with an SCCP firmware image. If the firmware installed on an IP phone is older than the firmware loaded on the Call Manager router flash, the IP phone automatically upgrades its firmware and then registers with the Callmanager. The filename conventions used for phone firmware images are:
  • SCCP firmware: P003xxyyzzww or SCCPxxyyzzww
  • SIP firmware: P0S3-xx-y-zz or SIPxxyyzzww
  • For Java-based IP phones, such as the Cisco Unified IP Phone 7911, 7941, 7941GE, 7961, 796GE, 7970, and 7971, the firmware consists of multiple files including JAR and tone files.

Installation of CME software

After you download the .zip CME software file from Cisco, uncompress the file on a local TFTP server. You will get several individual files and several TAR archives. We assume the TFTP server is at 192.168.10.1 and has access to the CallManager router.

 For individual files:

Use the regular copy command to transfer the file from TFTP to the router’s flash:

Example:

Router# copy tftp://192.168.10.1/P00307020300.sbn flash:

For TAR archive files:

Use the archive command to transfer the files and extract them at the same time to the router’s flash:

Example:

To transfer the Basic Files tar archive (cme-basic-3.0.3.tar) to callmanager router:

Router# archive tar /xtract tftp://192.168.10.1/cme-basic-3.0.3.tar flash:

After all required files are installed, use the “show flash” command to list the files installed on flash memory.