TechBee — Cisco IP Telephony Partner
Enterprise-grade Cisco Unified Communications architecture — call processing, IP phones, and WAN design — deployed and supported by TechBee across the UAE.
TechBee is one of the leading Cisco IP Telephony Partners in Dubai. IP telephony refers to communication services such as phone, fax, and voice-messaging that are delivered over a packet-switched IP network rather than a circuit-switched PSTN.
VoIP converts voice into IP packets and routes them to corresponding locations using voice-enabled routers. Users are typically unaware that VoIP has been implemented in the network; they continue to use their traditional phones, which are connected to a PBX. The PBX, on the other hand, is connected to a voice-enabled router that serves as an entrance point to VoIP rather than the PSTN or another PBX.
Traditional phones are replaced with IP phones, and PBXs are replaced with the Cisco Unified Communications Manager, a server for call control and signalling. Voice-to-IP conversion is handled by the IP phone itself, therefore voice-enabled routers aren’t required in the workplace network. If a link to the PSTN is necessary, a voice-enabled router or other gateway that forwards calls to the PSTN must be installed.
Converting an analogue speech signal into a digital format, then compressing and translating the digital signal into IP packets for transmission across an IP network, are the main processes in placing an IP telephone call. At the receiving end, the process is reversed.
The IP telephony architecture includes four distinct components: infrastructure, call processing, applications, and client devices.
Data link layer and multilayer switches, as well as voice-enabled routers, are used to connect endpoints to the IP and PSTN networks. Switched 10/100 Ethernet ports connect endpoints to the network. Power over Ethernet (PoE) ports on switches detect the presence of IP devices that require inline power, such as Cisco IP phones and wireless access points, and supply it. Conversions between the circuit-switched PSTN and IP networks are performed by voice-enabled routers.
Cisco Unified Communications Manager is the Cisco enterprise IP telephony solution’s software-based call-processing component. It is scalable, distributable, and highly available, acting similarly to a traditional telephone network’s PBX, including call setup and processing. It can be installed on Cisco MCS 7800 Series server platforms and selected third-party servers.
Applications enhance the IP telephony infrastructure by adding new functionalities. Cisco Unity unified messaging (combining e-mail and voice mail), Cisco Unified MeetingPlace (multimedia conferencing), Cisco Unified IP IVR, and Cisco Unified Contact Center products (intelligent contact routing, call treatment, network-to-desktop computer telephony integration, and multichannel automatic call distribution) are just a few examples. Third-party organisations can create software that works with Cisco Unified Communications Manager thanks to the open-source application layer.
Client devices are IP phones and software programmes that allow users to communicate over an IP network. The IP telephones are managed centrally by Cisco Unified Communications Manager via Ethernet connections in the Building Access Layer switches.
End-to-end IP telephony: Using end-to-end IP telephony between sites that have previously established IP connectivity. IP telephony can be implemented as an add-on service to existing infrastructure.
Cisco Unified Communications Manager, IP telephones, LAN switches with inline power (PoE), applications such as voice mail, and a voice-enabled router are all part of a single-site IP telephony solution. The IP telephones are powered by the LAN switch via their Ethernet interface. Users can make external calls using gateway trunks that are connected to the PSTN.
Each site can be totally self-contained with single-site deployment. The PSTN is used to make all calls to the outside world and to remote areas. If an IP WAN is used in the single-site model, it is exclusively for data traffic and does not provide telephony services — so if an IP WAN fails or the WAN’s capacity is insufficient, call processing is not affected. The only external requirements are a PSTN carrier and PSTN network route diversity.
Best practice: use this strategy for a single campus or a facility with fewer than 30,000 lines.
Cisco Unified Communications Manager at the central site links to outlying locations over the IP WAN in a multisite WAN design model with centralised call processing. The centralised Cisco Unified Communications Manager handles call processing for remote IP telephones. The IP WAN transfers voice traffic between locations and calls control signalling between the central and remote sites. Voice mail and IVR systems are likewise consolidated, lowering the total cost of ownership and allowing for centralised management and maintenance.
The Enterprise Campus requires IP connectivity from the remote locations. Voice is converted into IP packets and sent to the local LAN through IP telephones, which are powered by a local LAN switch. The packets are forwarded to the right destination by the local router using its routing table. With Cisco Unified Survivable Remote Site Telephony (SRST) services, voice-enabled routers at remote sites can provide backup call processing functionality in the case of a WAN outage, extending high-availability IP telephony to branch offices.
The service provider must integrate QoS measures if a company demands high-quality voice communication over the WAN. A service level agreement (SLA) is a contract between a company and a service provider that ensures bandwidth and latency standards adequate for voice transfer.
Note: The routers have speech capabilities, allowing voice communication with the rest of the world via the PSTN.
Best practice: use this architecture for a main site with numerous smaller remote sites connected via a QoS-enabled WAN but not requiring full capabilities during a WAN outage.
A multisite WAN design architecture with distributed call processing consists of a number of separate sites, each with its own call processing agent, connected to an IP WAN that transports voice traffic between them. Because each site has its own call processing agent, the IP WAN in this approach doesn’t transmit call control signalling between the sites. In most cases, the PSTN acts as a backup connection between the sites in the event that the IP WAN connection breaks or has inadequate bandwidth for incremental calls.
Note: A standalone site is one that is solely connected to the PSTN and is not covered by the distributed call processing architecture.
Best practice: use this architecture for a big central site with more than 30,000 lines, or a deployment with more than six large sites (totaling more than 30,000 lines) interconnected via a QoS-enabled WAN.
Note: Cisco Unified Communications Manager Express, embedded in Cisco IOS software, enables call processing for up to 240 Cisco Unified IP phones — extending IP telephony to a small site or branch office with a low-cost, dependable, feature-rich solution.
TechBee designs and deploys Cisco Unified Communications Manager environments for businesses across Dubai, Abu Dhabi, Sharjah, Ajman, Ras Al Khaimah, Fujairah, and Umm Al Quwain. Whether you need a Cisco IP telephony partner in Dubai, a Cisco Unified Communications Manager Express installer in Abu Dhabi, or a multisite WAN voice design consultant in Sharjah, our engineers handle site surveys, QoS planning, and SRST configuration for branch resilience.
We also support UAE system integrators and enterprises migrating from legacy PBX systems to Cisco IP telephony, including PSTN gateway integration and Cisco MCS server deployment.
Unified Communications Manager is the full enterprise call-processing server for larger deployments, while Communications Manager Express runs inside Cisco IOS software on a router, supporting up to 240 IP phones — a lower-cost option for small sites and branch offices.
In a centralized design, Cisco Unified SRST on the remote-site router takes over local call processing during a WAN outage. In a distributed design, each site already has its own call processing agent, so only inter-site calling is affected, with the PSTN available as backup.
Voice-to-IP conversion happens in the IP phone itself, so a voice-enabled router isn’t required for internal calling. You’ll still need one (or another gateway) if calls need to reach the PSTN.
Distributed designs are typically used for deployments exceeding 30,000 lines or more than six large sites interconnected via a QoS-enabled WAN, since each site manages its own call processing independently.
Yes, TechBee handles legacy PBX-to-Cisco migrations across the UAE, including number porting, PSTN gateway configuration, and staff training on the new system.
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R-12, France Cluster, International City, Dubai, UAE.
📞 Phone: 04-2434882
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✉️ Email: info@techbee.ae
🌐 Website: techbee.ae
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