Configure RTP Source Port Validation

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Updated:September 14, 2017
Document ID:212071

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    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




    TAC Authored

    Contributed by Cisco Engineers

    • Andres Salgado
      Cisco TAC Engineer
    • Henry Murray
      Cisco TAC Engineer

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