5The 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.
6The CODECs are enabled for both ends of the connection and the conversation proceeds using RTP/UDP/IP as the protocol stack.
7Any 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.
8When 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.
List of Terms
ACOM—Term used in G.165, “General Characteristics of International Telephone Connections and International Telephone Circuits: Echo Cancellers.” ACOM is the combined loss achieved by the echo canceller, which is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call.
Call leg—A logical connection between the router and either a telephony endpoint over a bearer channel or another endpoint using a session protocol.
CIR—Committed information Rate. The average rate of information transfer a subscriber (for example, the network administrator) has stipulated for a Frame Relay PVC.
CODEC—coder-decoder. Device that typically uses pulse code modulation to transform analog signals into a digital bit stream and digital signals back into analog signals. In Voice over IP, it specifies the voice coder rate of speech for a dial peer.
Dial peer—An addressable call endpoint. In Voice over IP, there are two kinds of dial peers: POTS and VoIP.
DTMF—Dual tone multifrequency. Use of two simultaneous voice-band tones for dial (such as touch tone).
E&M—E&M stands for recEive and transMit (or Ear and Mouth). E&M is a trunking arrangement generally used for two-way switch-to-switch or switch-to-network connections. Cisco’s E&M interface is an RJ-48 connector that allows connections to PBX trunk lines (tie lines).
FIFO—First-in, first-out. In data communication, FIFO refers to a buffering scheme where the first byte of data entering the buffer is the first byte retrieved by the CPU. In telephony, FIFO refers to a queuing scheme where the first calls received are the first calls processed.
FXO—Foreign Exchange Office. An FXO interface connects to the PSTN’s central office and is the interface offered on a standard telephone. Cisco’s FXO interface is an RJ-11 connector that allows an analog connection to be directed at the PSTN’s central office. This interface is of value for off-premise extension applications.
FXS—Foreign Exchange Station. An FXS interface connects directly to a standard telephone and supplies ring, voltage, and dial tone. Cisco’s FXS interface is an RJ-11 connector that allows connections to basic telephone service equipment, keysets, and PBXs.
Multilink PPP—Multilink Point-to-Point Protocol. This protocol is a method of splitting, recombining, and sequencing datagrams across multiple logical data links.
