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.
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