A voice over IP (VoIP) enabled network digitizes speech into packets and transmits those packets across a data network. This allows voice, data, and even video to share the same medium. In a network with unified communications (UC) such as voice, video and data, specialized UC servers, controllers, devices, and voice gateways are also likely to be used. In a cloud computing environment, they may be virtualized as well. Figure 9-22 shows a sample VoIP network topology. Not only can a VoIP network provide significant cost savings over a traditional PBX solution, may VoIP networks offer enhanced services (for example, integration with video conferencing applications and calendaring software to determine availability) not found in traditional corporate telephony environments.
An IP phone is a telephone with an integrated Ethernet connection. Although users speak into a traditional analog handset (or headset) on the IP phone, the IP phone digitizes the user's speech, packetizes it, and sends it out over a data network (via the IP phone's Ethernet port).While an IP phone is a common example of a VoIP endpoint, an alternative is software running on a computer.
A call agent is a repository for a VoIP network's dial plan. For example, when a user dials a number from an IP phone, the call agent analyzes the dialed digits and determines how to route the call toward the destination.
A gateway in a VoIP network acts as a translator between two different telephony signaling environments. In figure 9-22, both gateways interconnect a VoIP network with the PSTN. Also, the gateway on the right interconnects a traditional PBX with a VoIP network.
A private branch exchange (PBX) is a privately owned telephone switch traditionally used in corporate telephone systems. Although a PBX is not typically considered a VoIP device, it connects into a VoIP network through a gateway, as shown in figure 9-22.
An analog phone is a traditional telephone, like the ones individuals used to have in their homes. Even though an analog phone is not typically considered a VoIP device, it can connect to a VoIP network via a VoIP adapter or, as shown in figure 9-22, via a PBX, which is connected to a VoIP network.
Session Initiation Protocol (SIP) is a signaling, setup, and management protocol used with voice and video sessions over IP networks. SIP, in conjunction with other protocols, specifies the encoder/decoder (codec) that will be used for voice and video connections over the network.
Real-Time Transport Protocol (RTP) is a protocol that carries voice and interactive video. Notice in figure 9-22 that the bidirectional RTP stream does not flow through the call agent.
About The Author
Anthony Sequeira, CCIE No. 15626, is a Cisco Certified Systems Instructor (CCSI) and author regarding all levels and tracks of Cisco Certification. Anthony formally began his career in the information technology industry in 1994 with IBM in Tampa, Florida. He quickly formed his own computer consultancy, Computer Solutions, and then discovered his true passion-teaching and writing about Microsoft and Cisco technologies. Anthony joined Mastering Computers in 1996 and lectured to massive audiences around the world about the latest in computer technologies. Mastering Computers became the revolutionary online training company, KnowledgeNet, and Anthony trained there for many years. Anthony is currently pursuing his second CCIE in the area of Security and is a full-time instructor for the next-generation of KnowledgeNet, StormWind.com. Anthony is also a VMware Certified Professional.
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