How Voice over IP Processes a Telephone Call
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How Voice over IP Processes a Telephone Call The user picks up the handset; this signals an off-hook condition to the signaling application part of Voice over IP in the Cisco 3600 series router The session application part of Voice over IP issues a dial tone and waits for the user to dial a telephone number The user dials the telephone number; those numbers are accumulated and stored by the session application. After enough digits are accumulated to match a configured destination pattern, the telephone number is mapped to an IP host via the dial plan mapper. The IP host has a direct connection to either the destination telephone number or a PBX that is responsible for completing the call to the configured destination pattern. The session application then runs the H.323 session protocol to establish a transmission and a reception channel for each direction over the IP network. If the call is being handled by a PBX, the PBX forwards the call to the destination telephone. If RSVP has been configured, the RSVP reservations are put into effect to achieve the desired quality of service over the IP network. Any call-progress indications (or other signals that can be carried in-band) are cut through the voice path as soon as end-to-end audio channel is established. Signaling that can be detected by the voice ports (for example, in-band DTMF digits after the call setup is complete) is also trapped by the session application at either end of the connection and carried over the IP network encapsulated in RTCP using the RTCP APP extension mechanism When either end of the call hangs up, the RSVP reservations are torn down (if RSVP is used) and the session ends. Each end becomes idle, waiting for the next off-hook condition to trigger another call setup
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Voice over IP Configuration Task List Cisco 36001. Configure IP Networks for Real-Time Voice Traffic Configure your IP network to support real-time voice traffic. To configure your IP network for real-time voice traffic, you need to take into consideration the entire scope of your network, then select and configure the appropriate QoS tool or tools: (a) Multilink PPP with Interleaving (b) RTP Header Compression (c) Custom Queuing (d) Weighted Fair Queuing 2. Configure Frame Relay for Voice over IP configuring Voice over IP for it to run smoothly over Frame Relay. For example, a public Frame Relay cloud provides no guarantees for QoS. Use the num-exp command to configure number expansion if your telephone network is configured so that you can reach a destination by dialing only a portion (an extension number) of the full E.164 telephone number Use the dial-peer voice command to define dial peers and switch to the dial-peer configuration mode. Each dial peer defines the characteristics associated with a call leg There are two different kinds of dial peers: (a) POTS---Dial peer describing the characteristics of a traditional telephony network connection. POTS peers point to a particular voice port on a voice network device (b) VoIP---Dial peer describing the characteristics of a packet network connection; in the case of Voice over IP, this is an IP network 5. Optimize Dial Peer and Network Interface Configurations You can use VoIP peers to define characteristics such as IP precedence, additional QoS parameters (when RSVP is configured), CODEC, and VAD. Use the ip precedence command 6. Configure Voice Ports voice-port commands define the characteristics associated with a particular voice-port signaling type. voice ports on the Cisco 3600 series support three basic voice signaling types (a) FXO---Foreign Exchange Office interface (b) FXS---The Foreign Exchange Station interface (c) E&M---The "Ear and Mouth" interface (or "RecEive and TransMit" interface Under most circumstances, the default voice-port command values are adequate to configure FXO and FXS ports to transport voice data over your existing IP network. Because of the inherent complexities involved with PBX networks 7. Configure Voice over IP for Microsoft NetMeeting (Optional) Voice over IP can be used with Microsoft NetMeeting (Version 2.x) when the Cisco 3600 series router is used as the voice gateway.
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