Specifications

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Cisco 1751 Router Software Configuration Guide
OL-1070-01
How VoIP Processes a Typical Telephone Call
Before configuring VoIP on your router, it helps to understand what happens at an application level
when you place a call using VoIP. The general flow of a two-party voice call using VoIP is as follows:
1. The user picks up the handset; this signals an off-hook condition to the signaling application part
of VoIP in the router.
2. The session application part of VoIP issues a dial tone and waits for the user to dial a telephone
number.
3. The user dials the telephone number; those numbers are accumulated and stored by the session
application.
4. 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.
5. 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 Private
Branch Exchange (PBX), the PBX forwards the call to the destination telephone. If Resource
Reservation Protocol (RSVP) has been configured, the RSVP reservations are put into effect to
achieve the desired QoS over the IP network.
6. The coder-decoder compression schemes (CODECs) are enabled for both ends of the connection
and the conversation proceeds using Real-Time Transport Protocol/User Datagram
Protocol/Internet Protocol (RTP/UDP/IP) as the protocol stack.
7. Any call-progress indications (or other signals that can be carried inband) 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, inband dual-tone multifrequency (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 Real-Time Transport Control Protocol (RTCP) using the RTCP
application-defined (APP) extension mechanism.
8. 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.
Numbering Scheme
The standard PSTN is a large, circuit-switched network. It uses a specific numbering scheme, which
complies with the ITU-T international public telecommunications numbering plan (E.164)
recommendations. For example, in North America, the North American Numbering Plan (NANP) is
used, which consists of an area code, an office code, and a station code. Area codes are assigned
geographically, office codes are assigned to specific switches, and station codes identify a specific port
on that switch. The format in North America is 1Nxx-Nxx-xxxx, with N = digits 2 through 9 and x =
digits 0 through 9. Internationally, each country is assigned a one- to three-digit country code; the
country’s dialing plan follows the country code. In Cisco’s voice implementations, numbering schemes
are configured using the destination-pattern command.