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Introduction

 

This document covers the Procedure for Configuring the SIP Voice Gateway for IPV6 with examples.

 

Configuring a SIP Voice Gateway for IPv6

 

Users in a SIP network are identified by unique SIP addresses. A SIP address is similar to an e-mail address and is in the format of sip:userID@gateway.com. The user ID can be either a username or an E.164 address. The gateway can be either a domain (with or without a hostname) or a specific Internet IPv4 or IPv6 address.


A SIP trunk can operate in one of three modes: SIP trunk in IPv4-only mode, SIP trunk in IPv6-only mode, and SIP trunk in dual-stack mode, which supports both IPv4 and IPv6.

 

A Cisco Unified Border Element can interoperate between H.323/SIP IPv4 and SIP IPv6 networks in media flow-through mode. In media flow-through mode, both signaling and media flows through the Cisco Unified Border Element, and the Cisco Unified Border Element performs both signaling and media interoperation between H.323/SIP IPv4 and SIP IPv6 networks as shown in the below diagram.

 

sip ipv6.png

 

Procedure

 

 

Procedure to configure SIP Voice Gateway for IPv6

 

 

Step1: Enabling VoIPv6 Service on Cisco Gateway

 

Router> enable
Router # configure terminal
Router(config)# voice service voip
Router(config-voi-serv)# shutdown forced

 

Step2: Enabling VoIPv6 Submodes on Cisco Gateways

 

Router> enable
Router # configure terminal
Router(config)# voice service voip
Router(config-voi-serv)# sip
Router(config-serv-sip)# call service stop

 


Step3: Configuring the Protocol Mode of the SIP Stack

 

SIP service should be shut down before configuring the protocol mode. After configuring the protocol mode as IPv6, IPv4, or dual-stack, SIP service should be reenabled.

 

Router> enable
Router # configure terminal
Router(config)# sip-ua

 

Router(config-sip-ua)# protocol mode dual-stack

Syntax: protocol mode {ipv4 | ipv6 | dual-stack [preference {ipv4 | ipv6}]}

 

 

Step4: Disabling ANAT Mode


ANAT is automatically enabled on SIP trunks in dual-stack mode. This task describes how to disable ANAT in order to use a single-stack mode.

 

Router> enable
Router # configure terminal
Router(config)# voice service voip
Router(config-voi-serv)# sip
Router(conf-serv-sip)# no anat

 

 

Step5: Configuring the Source IPv6 Address of Signaling and Media Packets

 

Users can configure the source IPv4 or IPv6 address of signaling and media packets to a specific interface’s IPv4 or IPv6 address. Thus, the address that goes out on the packet is bound to the IPv4 or IPv6 address of the interface specified with the bind command. The bind command also can be configured with one IPv6 address to force the gateway to use the configured address when the bind interface has multiple IPv6 addresses. The bind interface should have both IPv4 and IPv6 addresses to send out ANAT.


Router> enable
Router # configure terminal
Router(config)# voice service voip
Router(config-voi-serv)# sip

 

Router(config-serv-sip)# bind control sourceinterface FastEthernet0/0

Syntax: bind {control | media | all} source-interface interface-id [ipv6-address ipv6-address]

 


Step6: Configuring the SIP Server

 

Router> enable
Router # configure terminal
Router(config)# sip-ua

 

Router(config-sip-ua)# sip-serveripv6:[2001:0DB8:0:0:8:800:200C:417A]

Syntax: sip-server {dns:[host-name] | ipv4:ipv4-address | ipv6:[ipv6-address][:port-num]}

 

Router(config-sip-ua)# keepalive target ipv6:[2001:0DB8:0:0:8:800:200C:417A]

Syntax: keepalive target {{ipv4:address | ipv6:address} | [:port] | dns:hostname} [tcp [tls]] | udp] [secondary]

 

Step7: Configuring the Session Target

 

Router> enable
Router # configure terminal

Router(config)# dial-peer voice 29 voip

Syntax: dial-peer voice tag {mmoip | pots | vofr | voip}

 

Router(config-dial-peer)# destination-pattern 7777

Syntax: destination-pattern [+]string[T]

 

Router(config-dial-peer)# session target [ipv6:2001:0DB8:0:0:8:800:200C:417A]

Syntax: session target {ipv4:destination-address | ipv6:[destination-address] | dns:[$s$. | $d$. | $e$. | $u$.]
host-name | enum:table-num | loopback:rtp | ras | sip-server} [:port]

 

Step8: Configuring SIP Register Support

 

Router> enable
Router # configure terminal
Router(config)# sip-ua

 

Router(config-sip-ua)# registrar ipv6:[3FFE:501:FFFF:5:20F:F7FF:FE0B:2972]

Syntax:registrar {dns:address | ipv4:destination-address [:port] | ipv6:destination-address[:port]} aor-domain expires seconds [tcp [tls]] type [secondary] [scheme string] expires 3600 secondary

 

Router(config-sip-ua)# retry register 10

Router(config-sip-ua)# timers register 500

 

 

Step9: Configuring Outbound Proxy Server Globally on a SIP Gateway

 

Router> enable
Router # configure terminal
Router(config)# voice service voip

Router(config-voi-serv)# sip

 

Router(config-serv-sip)# outbound-proxy ipv6[2001:0DB8:0:0:8:800:200C:417A]

Syntax: outbound-proxy {ipv4:ipv4-address |]ipv6:ipv6-address | dns:host:domain} [:port-number]

 

 

Step10: IPv4 to IPv6 Media Interoperating Through Cisco IOS MTP

 

Cisco Unified Border Element must be configured in IPv6-only or dual-stack mode to support IPv6 calls.

 

Router> enable
Router # configure terminal
Router(config)# voice service voip

 

Router(config-voi-serv)# allow-connections h323 to sip

Syntax: allow-connections from-type to to-type

 

 

Configuration Examples for Implementing VOIP over IPv6

 

1. Configuring the SIP Trunk


This example shows how to configure the SIP trunk to use dual-stack mode, with IPv6 as the preferred mode. The SIP service must be shut down before any changes are made to protocol mode configuration.


Router(config)# sip-ua
Router(config-sip-ua)# protocol mode dual-stack preference ipv6


2. Configuring the Source IPv6 Address of Signaling and Media Packets


This example shows how to configure the bind command:

 

Router(config)# voice service voip
Router(config-voi-serv)# sip
Router(config-serv-sip)# bind control source-interface FastEthernet 0/0


3. Configuring the SIP Server


This example shows how to configure the SIP server:


Router(config)# sip-ua
Router(config-sip-ua)# sip-server ipv6:[2001:0DB8:0:0:8:800:200C:417A]


4. Configuring the Session Target


This example shows how to configure the session target:


Router(config)# dial-peer voice 29 voip
Router(config-dial-peer)# destination-pattern 7777
Router(config-dial-peer)# session target ipv6:[2001:0DB8:0:0:8:800:200C:417A]

 

5. Configuring SIP Register Support


This example shows how to configure SIP register support:


Router(config)# sip-ua
Router(config-sip-ua)# registrar ipv6:[2001:0DB8:0:0:8:800:200C:417A] expires 3600 secondary
Router(config-sip-ua)# retry register 10
Router(config-sip-ua)# timers register 500


6. Configuring H.323 IPv4 to SIPv6 Connections in a Cisco Unified Border Element:

 

This example shows how to configure H.323 IPv4 to IPv6 connections in an Cisco Unified Border Element.


Router(config)# voice service voip
Router(config-voi-serv)# allow-connections h323 to sip


7. Configuring MTP for IPv4-to-IPv6 Translation


The following example shows how to configure MTP for IPv4-to-IPv6 translation and provides sample configuration output:


Router(global)# sccp ccm group 1
Router(config-sccp-ccm)# associate profile 5 register MTP3825
Router(config-sccp-ccm)# exit
Router(config)# dspfarm profile 5 mtp
Router(config-dspfarm-profile)# codec g711ulaw
Router(config-dspfarm-profile)# maximum sessions software 100
Router(config-dspfarm-profile)# associate application SCCP
Router# show sccp
sccp ccm group 1
associate profile 5 register MTP3825
!
dspfarm profile 5 mtp
codec g711ulaw
maximum sessions software 100
associate application SCCP

 

 


Verify

 

Verifying SIP Gateway Status


The show sip-ua calls command displays active user agent client (UAC) and user agent server (UAS) information on SIP calls:


Router# show sip-ua calls

 

Use the show sip-ua connections command to display SIP UA transport connection tables:
Router# show sip-ua connections udp brief

Router# show sip-ua connections udp detail


Use the show sip-ua status command to display the status of the SIP UA:
Router# show sip-ua status

 

 

 

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