Basic Concepts in IP Telephony and CUCM Lab

Basic Concepts in IP Telephony and CUCM Lab

Basic IP Telephony concepts and how to access Cisco Unified Communications Manager (CUCM) free environment from Cisco Devnet.

Static IPSec Tunnel VPN Configuration between Cisco Routers
Packet Tracer IP Telephony Lab using Communications Manager Express (CME)
Establish WAN eBGP Connection

Introduction

We will cover some basic IP Telephony concepts in this article and also demonstrate how to access Cisco Unified Communications Manager (CUCM) free lab from Cisco Devnet.

Hardware and Components

Call Processing/Call Control: Cisco Unified Communications Manager (CUCM) in Figure 1 or Cisco Unified Communications Express (CME) performs call processing and call control functions. It is the brain of the IP telephony system, and all other components are logically connected to the call control unit.

Figure 1 demonstrates a CUCM cluster, with 1 publisher and 1 subscriber. Publisher holds the read-write copy of database; while Subscriber holds the read-only copy of database. Subscriber replicates data from the Publisher. For example, when we configure a phone profile, the change occurs on the Publisher and is replicated to the Subscriber.  When a phone registers on CUCM, it can registers on either the Subscriber or the Publisher for redundancy.

Endpoints: IP phones in Figure 1 are endpoints.

Voice Gateway: Voice gateway sits on the border of our Voice over IP (VoIP) world and the traditional analogue or digital telephony world (Figure 1). It converts traditional telephony traffic into IP for transmission over a data network. A voice gateway is not required, if external calls will not be placed nor received. We can use a standard Cisco ISR router with the voice module, operation system with voice services and appropriate license as the voice gateway.

Application: There are also various unified communications application servers, adding rich features to IP telephony system, such as messaging, call centre services, voice mail etc.

Figure 1: IP Telephony basic components and protocol

In addition to the basic hardware components outlined above, you will see the following protocols often in the IP telephony world (Figure 1). In general, the voice protocols can be grouped under two categories as below: voice control and voice data. Voice control protocols look after call administration, registration, call signalling, call setup and call tear down.

Voice Control

  • SIP – stands for Session Initiation Protocol. SIP  is designed similar to the HTTP request/response transaction model, and works with both User Datagram Protocol (UDP) and Transmission Control Protocol (TCP). SIP is a vendor neutral protocol. Figure 2 demonstrates a SIP phone boot up and registration process. SIP adopts HTTP status code, “200 OK” for example. A complete list of HTTP status code is available from Wiki.
  • SCCP – stands for Skinny Call Control Protocol. SCCP is a Cisco proprietary protocol, which means SCCP is available if you use Cisco phones and CUCM.

Figure 2: SIP phone boot up and register on CUCM  (Source: Cisco, 2017)

  • H.323 –  is more rigid than SIP but promotes standardised multimedia communications, such as videoconferencing. It can be used as voice gateway protocol (Figure 1).
  • MGCP – stands for Media Gateway Control Protocol. It is used as voice gateway protocol (Figure 1).

Voice Data

  • RTP – stands for Real-time Transport Protocol, which delivers audio and video media stream over IP networks. It typically uses UDP. The media stream doesn’t detour via CUCM, but occurs directly between the two end points – the caller and the receiver. In the external call scenario, the RTP stream will occur between the internal end point and the voice gateway.

Cisco Devnet Sandbox – CUCM

Cisco Devnet Sandbox provides free CUCM environment and much more. The access link is: https://devnetsandbox.cisco.com/RM/Topology. Cisco ID logon is required.

Figure 3: Cisco Devnet Sandbox


I started the CUCM 11.5 lab with the topology as below. It includes 1 CUCM Publisher, 1 CUCM Subscriber, 1 IM&P server and 1 Active Directory/DNS server on Windows 2012.

Figure 4: Cisco Devnet Sandbox – CUCM 11.5 Enviornment

After you reserve the environment, it will take about 15 minutes to set up. You will receive an instruction email regarding how to access the environment upon the environment is initialised.

I access the environment via Cisco AnyConnect VPN, which you can download from https://developer.cisco.com/site/devnet/sandbox/anyconnect/. The instruction email will include VPN access details including IP, username and password.

After we VPN into the environment, open a web browser and access the CUCM via 10.10.20.1. Have a browse for now and we will come to feature configuration in subsequent articles.

Figure 5: Cisco CUCM 11.5


I also installed Cisco Jabber on my computer, which VPNed into Cisco Devnet environment. I registered Jabber to the CUCM Publisher to work as a soft phone (Figure 6). Figure 5 shows device “CSFUSER001”, from IP 192.168.6.1, is register with the CUCM Publisher. Cisco Unified Client Services Framework (CSF) device refers to Cisco Jabber installed on Mac or Windows. Please refer to On-Premises for Cisco Jabber 11.5 for other device types.

We define CUCM local user to log into Jabber, which can be configured under “User Management” in CUCM.

Figure 6: Cisco Jabber as soft phone

Reference

Cisco, 2017, IP Phone, SCCP & SIP Phone Registration Process with CUCM, https://supportforums.cisco.com/t5/collaboration-voice-and-video/ip-phone-sccp-amp-sip-phone-registration-process-with-cucm/ta-p/3109183

Cisco, 2017, On-Premises Deployment for Cisco Jabber 11.5, https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/11_5/CJAB_BK_D00D8CBD_00_deployment-installation-guide-cisco-jabber115/CJAB_BK_D00D8CBD_00_deployment-installation-guide-cisco-jabber115_chapter_01000.html

COMMENTS

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  • comment-avatar
    Eddie Zhang 2 months

    very thorough explanation!Thanks… Is there any simulation lab available?

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