SIP phones stationary Cisco. Fundamentals of IP-telephony, basic principles, terms and protocols

IP-telephony technology is one of the most convenient and popular ways to transfer voice traffic. The technology allows you to cover all employees of the enterprise, regardless of the structure of the organization, the number of branches and other factors. IP telephony  provides voice communications over a data network using the IP protocol.

Cisco has long been specializing in the design and manufacture of network devices. Cisco IP Telephony Solutions  are one of the main components of a single unified communications system that provides users with all the benefits of a converged network for any organization.

Allows you to create a well-scalable and reliable communication environment, making the most of the benefits of existing LAN and WAN networks. Thanks to the introduction of IP-telephony, it is possible to increase the productivity of employees.

Cisco Systems offers a unified communications model that enables communication over any channel and is independent of the access devices used. The Unified Communication model allows you to combine different communication devices and different networks, so that territorial independence can be achieved, as well as the integration of communications with business processes. All this helps to optimize and increase the productivity and profitability of the business.

Cisco IP Telephony Equipment, presented in the section, helps to create a well-protected platform, optimized for data transfer, support for wireless technology and IP communications. This is an easy and economical way to connect independent networks and analog telephone gateways based on existing telephony equipment. Using its own network, the organization has the ability to maximize control over the quality of communication.

IP-telephony is increasingly being used in modern organizations. Using Cisco equipment, you can optimize your organization’s communications capabilities and increase productivity.

IP-telephony is a modern way of transmitting voice over the Voice-over-IP (VoIP) protocol, namely, over any data transmission networks using the TCP / IP protocol (Internet or local area network of an organization). IP telephony today supplants the traditional way of communication, because it has undeniable advantages, such as:

  • Call cost reduction. The ability to call remote offices is completely free (only Internet traffic is paid). IP-telephony providers provide lower tariffs for long-distance and international calls.
  • Flexible call management. IP-telephony allows you to create and manage different user groups - for example, to prohibit inter-city calls.
  • Ample opportunities in the field of telephone and computer network integration: installing free software, the user gets the opportunity to make phone calls by selecting a subscriber from the list of contacts in MS Outlook.
  • Remote access to the telephone network. An employee can connect to the corporate network even from home.
  • Mobility and ease of maintenance. No additional settings are required for the employee’s phone in the event of a change of workplace. Unified infrastructure for your network and telephony.
  • Monitoring telephone line congestion using special software.
  • Ease of administering IP-PBX, which even a system administrator who does not have high qualifications can handle.

Since 1998, CBS has been a certified partner of Cisco, a world leader in the production of equipment for IP-telephony, and has significant experience in project implementation, which includes:

  • Deployment of corporate telephony based on IP-telephony.
  • Design and creation of IP-telephony call centers and contact centers with the functions of voice mail, auto attendant (IVR), personalized routing of calls, audio conferences, etc.
  • Design and configuration of a fault-tolerant IP-telephony system.
  • Integration of telephone networks of remote offices into a single telephone network using Internet channels.

CBS carries out the design of corporate networks based on IP-telephony for Russian and international companies. In developing the IP-telephony project for the international company Jaffra Cosmetics, CBS network technology specialists worked closely with the engineers of the local provider in Mexico, as well as with specialists from the Cisco Technical Support Center in Belgium (TAC). The professionalism of CBS in the implementation of IP-telephony projects can be assessed by getting acquainted with the completed

  • Tutorial

Most recently, a client came to our organization who asked to configure an IP-PBX based on the Cisco 2921 platform and Cisco CallManager Express (CME) software. It’s not decent to scatter customers in our time, so we decided to provide this service.

I took up this matter not so much to cut the dough for the company, but to get to know how Ip-telephony is arranged for this Cisco itself, for the purpose of self-development, that is.

I got acquainted. The theme is pleasant; for me as a tsiskar it’s much more convenient to work than with Asterisk.

You can write a lot about CME features. Today I will tell you about its very basics.
  Let's try to solve a simple problem: there are a couple of phones - one works on SIP, the other on SCCP, you need to make them call each other. How to do it - under the cut.

CME initial setup
  Setting the dhcp server on the router is considered to be the zero point, since almost all phones are configured by default to obtain an IP address in this way. If someone has forgotten or did not know, then on Tsiska this is done like this:

CME # conf t
  CME # (config) ip dhcp pool voicepool
  CME # (config-ip-dhcp) network 192.168.101.0 255.255.255.0
  CME # (config-ip-dhcp) default-router 192.168.101.1
  CME # (config-ip-dhcp) option 150 ip 192.168.101.1
  CME # (config-ip-dhcp) lease 0 0 30

All settings seem to be obvious. Except, probably the 150th option. This option transfers the address of the tftp server in DHCP settings (I will not write about using the tftp server in this article, but this thing is very useful for working with phones)

By default, Cisco is very good and sympathetic to the operation of the SCCP protocol (due to the fact that this protocol belongs to Cisco itself), and is not very friendly with the more common SIP. So, for example, to simply allow calls between SIP phones connected to our PBX, you need to configure:

CME # conf t

  CME # (config-voice) allow-connections sip to sip

CME # conf t
  CME # (config) voice service voip
  CME # (config-voice) sip
CME # (config-voice-sip) registrar server expires max 3600 min 3600

Now you need to create the so-called codec class - a set of codecs to be used:

CME # conf t
  CME # (config) voice class codec 1
  CME # (config-voice) codec preference 1 g711alaw
  CME # (config-voice) codec preference 2 g711ulaw
  CME # (config-voice) codec preference 3 g729br8

The first attempt will be made to use the 711alaw codec, if not succeed, then g711ulaw, well, then intuitively.

These settings will be enough.

Setup for work with SCCP
  All settings regarding the operation of SCCP protocol occur in the section telephony-service

CME # conf t
  CME # (config) telephony-service
  CME # (config-telephony) max-ephones 10
  CME # (config-telephony) max-dn 10
  CME # (config-telephony) ip source-address 192.168.101.1 port 2000

ephone  - this is such a thing that, in simple terms, symbolizes a telephone, well, that is, the image of a telephone set in the concept of CME. Parameter max-ephones  responsible, respectively, for how many SCCP-phones can be registered on this PBX.

dn  - (directory number) this is such a thing that, in simple terms, symbolizes a telephone number. By analogy, it is clear what the parameter is responsible for max-dn.

By the way, it’s not very correct to indicate too many ephone  and dn, because under each of these "units" the CME will allocate RAM during initialization.

Plus, the number of supported ephone  and dn  Depends on the platform model and software version.
  ip source-address  sets the IP address (and as a result the interface) and the port on which requests from SCCP phones will be listened

Configuring CME to work with SIP phones
  All settings regarding the operation of the SIP protocol occur in the section voice register global. Here is the minimum we need:

CME # conf t
  CME # (config) voice register global
  CME # (config-voice) mode cme
  CME # (config-voice) source-address 192.168.101.1 port 5060
  CME # (config-voice) max-dn 50
  CME # (config-voice) max-pool 36

Two differences. The first is to execute the command mode cme, which, as it were, hints to the router that CME will work with SIP phones now (this is to the fact that not only the software we describe works with SIP protocol). The second - now instead ephonwe will operate voice pools

Register phones
  You can begin to configure the numbering and phone registration settings.
  The technology in both protocols is the same and simple enough - first it is created dn, then a “telephone” is created and its number (or several) is attached to the telephone.
In our example, for authorization of telephone sets on the server, we will use their MAC addresses. This is logical, since in practice a certain telephone is assigned to a certain person and a telephone number, and a person usually rarely moves his workplace in the office, and if he does, he transfers the telephone with him.
Registering a SCCP Phone
  Create a phone number:

CME # conf t
  CME # (config) ephone-dn 1
  CME # (config-ephone-dn) number 100
  CME # (config-ephone-dn) description Beauty-User
  CME # (config-ephone-dn) name The Real Man

We created the phone number 100, the description inside the CME is “Beauty-User”, the name displayed on the phone will be “The Real Man”

Create the image of the telephone. we attach it to the MAC address of the real device, we attach the number to it:

CME # conf t
  CME # (config) ephone 1
  CME # (config-ephone) mac-address B4A4.E328.BDEC
  CME # (config-ephone) button 1: 1

The binding of the telephone (or rather the telephone line of our telephone) is done by the team button 1: 1. The format is as follows - first we type the keyword "Button", then we indicate the number of this “button” (the first in our example), then we indicate the action to be performed (":" - a colon means that now we will attach some telephone number to the selected line) and then the parameter corresponding to the action - in In our case, we indicate the phone number (that is, the number of the pre-configured dn).

You probably saw in your life \\ in the pictures a ciscophone? Did you see there all kinds of buttons near the screen? This is the very same buttons. In fact, each of these buttons is responsible for its line. And you can perform many actions on them (not just bind to any number). But this is worthy of a separate article.

Meanwhile, we can safely turn on the phone with the specified MAC address in the network, wait until it loads, see on its screen near the first button number 100 and the cherished name - “The Real Man”. If we pick up the phone, we hear the treasured buzzer

Register a SIP Phone
  The directory number is created as follows:

CME # conf t
  CME # (config) voice register dn 1
  CME # (config-voice-register-dn) number 200

CME # conf t
  CME # (config) voice register pool 1
  CME # (config-voice-register-pool) id mac 1CDF.0F4A.152E
  CME # (config-voice-register-pool) number 1 dn 1
  CME # (config-voice-register-pool) voice-class codec 1
  CME # (config-voice-register-pool) username test password test

Snapping to the first line of the first number is a command number 1 dn 1.
  The team   voice-class codec 1 Define a set of valid codecs for the phone (we described this set at the beginning).
  Next team username \\ password  set authentication data.

In order for the phone to register, you need to go to its web face and in the settings of the first line specify the server address - 192.168.101.1 and authentication data. Save.
  Reboot.

  PROFIT!

The IP address of the phone can be found from the information from the dhcp pool by poppy using the show ip dhcp binding command, either in the phone’s settings - voice or visual menu.

It seems to be all. The minimum CME settings have been described so that two phones (SIP and SCCP) can register on it, having received the address through DHCP.

Sources of knowledge
  The primary source of knowledge to get you started was Jeremy Cioara's CCNA Voice video course. Pretty lively language (English truth) talks about IP-telephony in general and about the nuances of setting up this very CME. True, there is not a word touched on the setting of SIP phones.

A very good manual, of course, is on cisco.com. It is called the Cisco Unified Communications Manager Express System Administration Guide. Available

Currently there is a huge selection of software with which you can conduct telephone conversations over the Internet. In fact, the organization of voice data transmission over a network requires a minimum set of hardware: a computer connected to the network, appropriate software or application, a microphone and headphones. But for reputable companies, such a solution can hardly be called acceptable: it does not correspond to either the status of the organization or the range of tasks to be solved. The best solution in this case is the introduction of ip telephony, and Cisco Systems offers a huge selection of phones for organizing corporate communications.

Cisco IP Phones: Features,
technical characteristics, advantages

According to expert estimates, since 1997 the corporation has maintained its position as a leader in the market of VoIP telephony solutions. Thanks to what embedded technologies does she manage to achieve this? The thing is that Cisco IP Phones  They are an entire family of devices with an impressive array of features.

  July 19, 2011 at 01:51

The Basics of Cisco IP Telephony or Introducing Cisco CallManager Express

  • Development of communication systems
  • Tutorial

Most recently, a client came to our organization who asked to configure an IP-PBX based on the Cisco 2921 platform and Cisco CallManager Express (CME) software. It’s not decent to scatter customers in our time, so we decided to provide this service.

I took up this matter not so much to cut the dough for the company, but to get to know how Ip-telephony is arranged for this Cisco itself, for the purpose of self-development, that is.

I got acquainted. The theme is pleasant; for me as a tsiskar it’s much more convenient to work than with Asterisk.

You can write a lot about CME features. Today I will tell you about its very basics.
  Let's try to solve a simple problem: there are a couple of phones - one works on SIP, the other on SCCP, you need to make them call each other. How to do it - under the cut.

CME initial setup
  Setting the dhcp server on the router is considered to be the zero point, since almost all phones are configured by default to obtain an IP address in this way. If someone has forgotten or did not know, then on Tsiska this is done like this:

CME # conf t
  CME # (config) ip dhcp pool voicepool
  CME # (config-ip-dhcp) network 192.168.101.0 255.255.255.0
  CME # (config-ip-dhcp) default-router 192.168.101.1
  CME # (config-ip-dhcp) option 150 ip 192.168.101.1
  CME # (config-ip-dhcp) lease 0 0 30

All settings seem to be obvious. Except, probably the 150th option. This option transfers the address of the tftp server in DHCP settings (I will not write about using the tftp server in this article, but this thing is very useful for working with phones)

By default, Cisco is very good and sympathetic to the operation of the SCCP protocol (due to the fact that this protocol belongs to Cisco itself), and is not very friendly with the more common SIP. So, for example, to simply allow calls between SIP phones connected to our PBX, you need to configure:

CME # conf t

  CME # (config-voice) allow-connections sip to sip

CME # conf t
  CME # (config) voice service voip
  CME # (config-voice) sip
CME # (config-voice-sip) registrar server expires max 3600 min 3600

Now you need to create the so-called codec class - a set of codecs to be used:

CME # conf t
  CME # (config) voice class codec 1
  CME # (config-voice) codec preference 1 g711alaw
  CME # (config-voice) codec preference 2 g711ulaw
  CME # (config-voice) codec preference 3 g729br8

The first attempt will be made to use the 711alaw codec, if not succeed, then g711ulaw, well, then intuitively.

These settings will be enough.

Setup for work with SCCP
  All settings regarding the operation of SCCP protocol occur in the section telephony-service

CME # conf t
  CME # (config) telephony-service
  CME # (config-telephony) max-ephones 10
  CME # (config-telephony) max-dn 10
  CME # (config-telephony) ip source-address 192.168.101.1 port 2000

ephone  - this is such a thing that, in simple terms, symbolizes a telephone, well, that is, the image of a telephone set in the concept of CME. Parameter max-ephones  responsible, respectively, for how many SCCP-phones can be registered on this PBX.

dn  - (directory number) this is such a thing that, in simple terms, symbolizes a telephone number. By analogy, it is clear what the parameter is responsible for max-dn.

By the way, it’s not very correct to indicate too many ephone  and dn, because under each of these "units" the CME will allocate RAM during initialization.

Plus, the number of supported ephone  and dn  Depends on the platform model and software version.
  ip source-address  sets the IP address (and as a result the interface) and the port on which requests from SCCP phones will be listened

Configuring CME to work with SIP phones
  All settings regarding the operation of the SIP protocol occur in the section voice register global. Here is the minimum we need:

CME # conf t
  CME # (config) voice register global
  CME # (config-voice) mode cme
  CME # (config-voice) source-address 192.168.101.1 port 5060
  CME # (config-voice) max-dn 50
  CME # (config-voice) max-pool 36

Two differences. The first is to execute the command mode cme, which, as it were, hints to the router that CME will work with SIP phones now (this is to the fact that not only the software we describe works with SIP protocol). The second - now instead ephonwe will operate voice pools

Register phones
  You can begin to configure the numbering and phone registration settings.
  The technology in both protocols is the same and simple enough - first it is created dn, then a “telephone” is created and its number (or several) is attached to the telephone.
In our example, for authorization of telephone sets on the server, we will use their MAC addresses. This is logical, since in practice a certain telephone is assigned to a certain person and a telephone number, and a person usually rarely moves his workplace in the office, and if he does, he transfers the telephone with him.
Registering a SCCP Phone
  Create a phone number:

CME # conf t
  CME # (config) ephone-dn 1
  CME # (config-ephone-dn) number 100
  CME # (config-ephone-dn) description Beauty-User
  CME # (config-ephone-dn) name The Real Man

We created the phone number 100, the description inside the CME is “Beauty-User”, the name displayed on the phone will be “The Real Man”

Create the image of the telephone. we attach it to the MAC address of the real device, we attach the number to it:

CME # conf t
  CME # (config) ephone 1
  CME # (config-ephone) mac-address B4A4.E328.BDEC
  CME # (config-ephone) button 1: 1

The binding of the telephone (or rather the telephone line of our telephone) is done by the team button 1: 1. The format is as follows - first we type the keyword "Button", then we indicate the number of this “button” (the first in our example), then we indicate the action to be performed (":" - a colon means that now we will attach some telephone number to the selected line) and then the parameter corresponding to the action - in In our case, we indicate the phone number (that is, the number of the pre-configured dn).

You probably saw in your life \\ in the pictures a ciscophone? Did you see there all kinds of buttons near the screen? This is the very same buttons. In fact, each of these buttons is responsible for its line. And you can perform many actions on them (not just bind to any number). But this is worthy of a separate article.

Meanwhile, we can safely turn on the phone with the specified MAC address in the network, wait until it loads, see on its screen near the first button number 100 and the cherished name - “The Real Man”. If we pick up the phone, we hear the treasured buzzer

Register a SIP Phone
  The directory number is created as follows:

CME # conf t
  CME # (config) voice register dn 1
  CME # (config-voice-register-dn) number 200

CME # conf t
  CME # (config) voice register pool 1
  CME # (config-voice-register-pool) id mac 1CDF.0F4A.152E
  CME # (config-voice-register-pool) number 1 dn 1
  CME # (config-voice-register-pool) voice-class codec 1
  CME # (config-voice-register-pool) username test password test

Snapping to the first line of the first number is a command number 1 dn 1.
  The team   voice-class codec 1 Define a set of valid codecs for the phone (we described this set at the beginning).
  Next team username \\ password  set authentication data.

In order for the phone to register, you need to go to its web face and in the settings of the first line specify the server address - 192.168.101.1 and authentication data. Save.
  Reboot.

  PROFIT!

The IP address of the phone can be found from the information from the dhcp pool by poppy using the show ip dhcp binding command, either in the phone’s settings - voice or visual menu.

It seems to be all. The minimum CME settings have been described so that two phones (SIP and SCCP) can register on it, having received the address through DHCP.

Sources of knowledge
  The primary source of knowledge to get you started was Jeremy Cioara's CCNA Voice video course. Pretty lively language (English truth) talks about IP-telephony in general and about the nuances of setting up this very CME. True, there is not a word touched on the setting of SIP phones.

A very good manual, of course, is on cisco.com. It is called the Cisco Unified Communications Manager Express System Administration Guide. Available

Do you like the article? Share with friends: