Contents

Introduction

Prerequisites

Requirements

Components Used

Background Information

Configuration Check:

Configuration Steps

Verify

Troubleshoot

Additional Reference

Introduction

This document describes how to enable Real Time Protocol (RTP) source port validation in order to avoid voice quality problem like crosstalk.

Prerequisites

Requirements

IOS 12.4(6)T or newer versions.

Components Used

This document is not restricted to specific software and hardware versions.

Caution: The information in this document was created from the devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If your network is live, ensure that you understand the potential impact of any command.

Background Information

Public Switch Telephone Network (PSTN) callers experience crosstalk or mixed voice streams in this call flow:

IP Phones -- Cisco Unified Communications Manager (CUCM) --- Session Initiation Protocol (SIP) IOS Gateway -- PSTN

This could happen when the gateway receives an invalid RTP stream destined to the same IP address and port of an active call.

The invalid stream has different source IP address and port than the one negotiated via SIP Session Description Protocol (SDP).

Configuration Check:

Check whether:

a) Hoot n Holler is being used:

dial-peer voice x voip

session protocol multicast

CLI definition

This command is used for voice conference configurations in a hoot and holler networking implementation. This command allows more than two ports to join the same session simultaneously.

b)  If SIP is configured:

dial-peer voice x voip

session protocol sipv2

CLI definition

Configuration Steps

Configure these commands:

voice service voip

sip

source filter

This command eliminates the risk for crosstalk since the gateway blocks all rogue audio from an unknown source.

Note: The above command works for SIP only, so H323, Media Gateway Control Protocol (MGCP), and Skinny Client Control Protocol (SCCP) are still affected.

Configuration example:

voice service voip

allow-connections sip to sip

sip

bind control source-interface loopback0

bind media source-interface loopback0

source filter

dial-peer voice 2001 voip

destination-pattern 79...

session protocool sipv2

session target ipv4:172.16.32.21

incoming called-number .

voice-class codec 1

dtmf-relay rtp-nte

Verify

There is currently no verification procedure available for this configuration.

Troubleshoot

There is currently no specific troubleshooting information available for this configuration.

Additional Reference