How To Implement A Cisco Vip From Scratch

Transcription

1 Overview of Cisco VoIP Infrastructure Solution for SIP The Cisco VoIP Infrastructure Solution for SIP implements a voice-over-packet network design using SIP to provide telephony services. It lays the foundation for building SIPbased VoIP solution, which is build using Cisco products. Components used implement toll by-pass, effect Dedicated Access Line (DAL) replacement and provide enhanced IP telephony services such as a scaleable private number plan, and to provide desktop services such as call forwarding, call hold, and call transfer. The solution includes: a SIP telephone a SIP gateway a SIP proxy server a unified messaging server a firewall a VoIP solution for service providers These components work together to provide a SIP-based VIP solution that can be integrated with existing telephone systems There is a phased implementation procedure of the Cisco infrastructure solution to VoIP and the solution is implemented from an intranetwork and an internetwork approach. The intranetwork phased implementation of the Cisco VoIp involves first replacing the traditional Dedicated Access Line (DAL) and by-passing carrier toll lines by introducing Cisco SIP gateways and an IP network between the private branch exchanges (PBXs) as follows:

2 The next phase involves introducing SIP proxy servers to provide support for a scalable private number plan to the IP network as follows: The next phase is the addition of Cisco telephones which connect directly to the IP network. They provide features such as Call Waiting, Call Holding, Calltransfer and Call forwarding. A Radius Server is integrated into the IP network with the SIP proxy servers to provide application services. This enables the SIP proxy servers to perform Authentication. It also provides end customers with enhanced services, such as find me and call screening. The Sip gateways interface with the application services using the Radius server for billing purposes.

3 A unified messaging server is added to provide voice mail To summarize the final intranetwork phase: A Quality of Service IP Network is central to the network using Cisco internetworking equipment with a set of Cisco Gateways and one or two proxy servers. The Cisco SIP gateways are connected to the PBXs using T1 or E1 lines with channel associated signaling (CAS) or primary rate (PRI) signaling. Telephones and Fax machines are connected to the PBXs SIP IP phones are connected directly to the IP network A server running a unified messaging service is connected to the IP network Sip is used for signaling between Sip clients, the Cisco SIP IP phones, the Cisco SIP gateways, and the SIP proxy servers RTP/RTCP is used to transmit voice data between the SIP endpoints after sessions are established.

4 The Internet work phased approach is as follows, this is the solution to integrating a SIP enabled VoIP network with the public network (PSDN) structure PIX firewalls are added to provide inside security Cisco SS7 Interconnect for Voice Gateways Solution components are added so as to integrate the SIP enabled VoIP network with a public infrastructure

5 When calls are made within a single SIP IP telephone network, the process involves the origination and destination phones and a single proxy server 1. Phone A initiates the call by sending an INVITE message to the SIP proxy server 2. The Sip proxy server interacts with the location server to determine user addressing, location and features. It may also interact with application services. 3. The SIP proxy server then proxies the INVITE message to the destination phone. 4. After a response and acknowledgement are exchange with each phone, an RTP session is established between Phone A and B To process calls between Sip IP telephony networks, the process involves the origination and destination phones as well as two or more proxy servers

6 1. Sip Phone A initiates a call by sending an INVITE to the SIP proxy server 2. The SIP phone can obtain additional information from the Radius server to interact with application services. 3. The Sip proxy server in phone A s network contacts the SIP proxy server in phone B s network 4. The local proxy uses the domain name system (DNS) to determine if it should handle the call or route to another proxy server. 5. The SIP proxy server in phone B s network may interact with application services to obtain additional information. 6. The SIP proxy server in phone B s network then contacts the destination phone. 7. After responses and acknowledgments are exchanged, an RTP session is established between the Cisco SIP phones A and B. When calls are made between a SIP P telephony network and a traditional telephone network, the process involves the originating phone, one or more proxy servers, a gateway, and a PBX or PSDN device 1. The Cisco Phone sends an INVITE to the SIP proxy server 2. The SIP proxy server might interact with RADIUS for application services 3. The SIP proxy server proxies the INVITE to the Cisco gateway 4. The Cisco gateway establishes communication with the traditional telephony network 5. An RTP session is established after responses and acknowledgements re exchanged.

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