Troubleshooting guide

4-10
Cisco Broadband Local Integrated Services Solution Troubleshooting Guide
OL-5169-01
Chapter 4 Troubleshooting with Call Flows
Understanding SS7
Understanding SS7
Connections to the PSTN are done through ISDN user part (ISUP) trunks with a trunk gateway providing
the bearer connections, and an SS7 gateway providing the signaling connections into the SS7 network.
SS7 provides call setup and teardown, network management, fault resolution, and traffic management
services. The SS7 network is used solely for network control, and the only data sent over it is signaling
messages.
The Call Agent maps the PSTN and trunk gateway bearer circuits to the SS7 connection. This allows the
Call Agent to determine which bearer circuit to use for the outgoing part of a call when a call originates
from the PSTN or trunk gateway. Each SS7 connection is identified using the following three codes:
• Originating Point Code (OPC)
• Destination Point Code (DPC)
• Circuit Identification Code (CIC)
The signaling controller uses the SS7 circuit identification information to uniquely identify each bearer
circuit, which is identified by:
• Span ID (the trunk ID)
• Timeslot within the trunk
When troubleshooting a call flow, you need to may need to access the Call Agent mapping table to
identify a call.
Processing a Telephone Call
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:
Step 1 The user picks up the handset. This signals an off-hook condition that is generated by the end-office
Service Switching Point (SSP) in the PSTN.
Step 2 The SSP provides digit analysis and route determination, then generates dial tone and sends an IAM to
the signaling application part of VoIP in the gateway.
Step 3 The VoIP issues an alerting/ACM that indicates to the SSP to start the dial tone or busy tone (based on
the state of the receiving end).
Step 4 Upon dial tone, the user dials the telephone number. Those numbers are accumulated and stored by the
session application.
Step 5 After enough digits are accumulated to match a configured destination pattern, the telephone number is
mapped to an IP host through the dial plan mapper. The IP host has a direct connection to the destination
telephone number.
Step 6 The session application uses the MGCP 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.
Step 7 The codecs are enabled for both ends of the connection and the conversation proceeds using
RTP/UDP/IP as the protocol stack.