Typical voice call handling in a VoIP application
It is useful to understand what happens at an application level when a call is placed using
VoIP. The diagram below describes the general flow of a two-party voice call using VoIP.
Step 1 - Off Hook
Step 2 - Dial Tone
Step 3 - Dialed Digits
Step 4 - IP Address resolution
Step 5 - Call Request using H.323/SIP
Step 5 - Call Request using H.323/SIP
Step 5 - PSTN Call signaling to phone
Step 5 - Call signaling to PBX
Step 5 - Off Hook IP Network PSTN Handset VoIP Application Gateway PBX Phone
Step 7 - DTMF CODEC
Step 8 - Bye
Talk-Audio
Ringing
Typical VoIP Call Handling
Step Action
Step1. The user picks up the handset; this signals an off-hook condition to the signaling
application part of VoIP.
Step2. The session application part of VoIP issues a dial tone and waits for the user to dial a
telephone number.
Step3. The user dials the telephone number; those numbers are accumulated and stored by
the session application.
Step4. 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.
Step5. The session application then runs the session protocol (H.323 or SIP/MGCP) 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.
Step6. 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.
Step7. Any call-progress indications (or other signals that can be carried inband) are cut
through the voice path as soon as an 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. It is carried over the IP network, encapsulated in the
Real-time Transport Control Protocol (RTCP) using the RTCP application-defined (APP)
extension mechanism.
Step8. 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.