Cisco Unified Border Element (Enterprise) Protocol-Independent Features and Setup Configuration Guide, Cisco IOS XE Release 3S (Cisco ASR 1000)
VoIP for IPv6
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VoIP for IPv6

VoIP for IPv6

Last Updated: July 30, 2012

This document describes VoIP in IPv6 (VoIPv6), a feature that adds IPv6 capability to existing VoIP features. This feature adds dual-stack (IPv4 and IPv6) support on voice gateways and media termination points (MTPs), IPv6 support for Session Initiation Protocol (SIP) trunks, and support for Skinny Client Control Protocol (SCCP)-controlled analog voice gateways. In addition, the Session Border Controller (SBC) functionality of connecting a SIP IPv4 or H.323 IPv4 network to a SIP IPv6 network is implemented on a Cisco Unified Border Element to facilitate migration from VoIPv4 to VoIPv6.

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and feature information, see Bug Search Tool and the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the feature information table at the end of this module.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Prerequisites for VoIP for IPv6

  • Cisco Express Forwarding for IPv6 must be enabled.
  • Virtual routing and forwarding (VRF) is not supported in IPv6 calls.
Cisco Unified Border Element
  • Cisco IOS Release 12.4(22)T or a later release must be installed and running on your Cisco Unified Border Element.
Cisco Unified Border Element (Enterprise)
  • Cisco IOS XE Release 3.3S or a later release must be installed and running on your Cisco ASR 1000 Series Router.

Information About VoIP for IPv6

SIP Voice Gateways in VoIPv6

SIP is a simple, ASCII-based protocol that uses requests and responses to establish communication among the various components in the network and to ultimately establish a conference between two or more endpoints.

For further information about this feature and information about configuring the SIP voice gateway for VoIPv6, see the Configuring a SIP Voice Gateway for IPv6.

MTP Used with Voice Gateways in VoIPv6

Cisco IOS MTP trusted relay point (TRP) supports media interoperation between IPv4 and IPv6 networks.

How to Configure VoIP for IPv6

Configuring a SIP Voice Gateway for IPv6

SIP is a simple, ASCII-based protocol that uses requests and responses to establish communication among the various components in the network and to ultimately establish a conference between two or more endpoints.

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 SIP trunk uses the Alternative Network Address Transport (ANAT) mechanism to exchange multiple IPv4 and IPv6 media addresses for the endpoints in a session. ANAT is automatically enabled on SIP trunks in dual-stack mode. The ANAT Session Description Protocol (SDP) grouping framework allows user agents (UAs) to include both IPv4 and IPv6 addresses in their SDP session descriptions. The UA is then able to use any of its media addresses to establish a media session with a remote UA.

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 (see the figure below).

Figure 1 H.323/SIP IPv4--SIP IPv6 Interoperating in Media Flow-Through Mode


Shutting Down or Enabling VoIPv6 Service on Cisco Gateways

SUMMARY STEPS

1.    enable

2.    configure terminal

3.    voice service voip

4.    shutdown [ forced ]


DETAILED STEPS
  Command or Action Purpose
Step 1
enable


Example:

Device> enable

 

Enables privileged EXEC mode.

  • Enter your password if prompted.
 
Step 2
configure terminal


Example:

Device# configure terminal

 

Enters global configuration mode.

 
Step 3
voice service voip


Example:

Device(config)# voice service voip

 

Enters voice service VoIP configuration mode.

 
Step 4
shutdown [ forced ]


Example:

Device(config-voi-serv)# shutdown forced

 

Shuts down or enables VoIP call services.

 

Shutting Down or Enabling VoIPv6 Submodes on Cisco Gateways

SUMMARY STEPS

1.    enable

2.    configure terminal

3.    voice service voip

4.    sip

5.    call service stop [forced] [maintain-registration


DETAILED STEPS
  Command or Action Purpose
Step 1
enable


Example:

Device> enable

 

Enables privileged EXEC mode.

  • Enter your password if prompted.
 
Step 2
configure terminal


Example:

Device# configure terminal

 

Enters global configuration mode.

 
Step 3
voice service voip


Example:

Device(config)# voice service voip

 

Enters voice service VoIP configuration mode.

 
Step 4
sip


Example:

Device(config-voi-serv)# sip

 

Enters SIP configuration mode.

 
Step 5
call service stop [forced] [maintain-registration


Example:

Device(config-serv-sip)# call service stop

 

Shuts down or enables VoIPv6 for the selected submode.

 

Configuring the Protocol Mode of the SIP Stack

Before You Begin

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.


SUMMARY STEPS

1.    enable

2.    configure terminal

3.    sip-ua

4.    protocol mode ipv4 | ipv6 | dual-stack [preference {ipv4 | ipv6}]}


DETAILED STEPS
  Command or Action Purpose
Step 1
enable


Example:

Device> enable

 

Enables privileged EXEC mode.

  • Enter your password if prompted.
 
Step 2
configure terminal


Example:

Device# configure terminal

 

Enters global configuration mode.

 
Step 3
sip-ua


Example:

Device(config)# sip-ua

 

Enters SIP user agent configuration mode.

 
Step 4
protocol mode ipv4 | ipv6 | dual-stack [preference {ipv4 | ipv6}]}


Example:

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

 

Configures the Cisco IOS SIP stack in dual-stack mode.

 
Example: 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.

Device(config)# sip-ua
Device(config-sip-ua)# protocol mode dual-stack preference ipv6
Disabling ANAT Mode

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

SUMMARY STEPS

1.    enable

2.    configure terminal

3.    voice service voip

4.    sip

5.    no anat


DETAILED STEPS
  Command or Action Purpose
Step 1
enable


Example:

Device> enable

 

Enables privileged EXEC mode.

  • Enter your password if prompted.
 
Step 2
configure terminal


Example:

Device# configure terminal

 

Enters global configuration mode.

 
Step 3
voice service voip


Example:

Device(config)# voice service voip

 

Enters voice service VoIP configuration mode.

 
Step 4
sip


Example:

Device(config-voi-serv)# sip

 

Enters SIP configuration mode.

 
Step 5
no anat


Example:

Device(conf-serv-sip)# no anat

 

Disables ANAT on a SIP trunk.

 

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.

When you do not specify a bind address or if the interface is down, the IP layer still provides the best local address.

SUMMARY STEPS

1.    enable

2.    configure terminal

3.    voice service voip

4.    sip

5.    bind {control | media | all} source interface interface-id [ipv6-address ipv6-address


DETAILED STEPS
  Command or Action Purpose
Step 1
enable


Example:

Device> enable

 

Enables privileged EXEC mode.

  • Enter your password if prompted.
 
Step 2
configure terminal


Example:

Device# configure terminal

 

Enters global configuration mode.

 
Step 3
voice service voip


Example:

Device(config)# voice service voip

 

Enters voice service VoIP configuration mode.

 
Step 4
sip


Example:

Device(config-voi-serv)# sip

 

Enters SIP configuration mode.

 
Step 5
bind {control | media | all} source interface interface-id [ipv6-address ipv6-address


Example:

Device(config-serv-sip)# bind control source- interface FastEthernet 0/0

 

Binds the source address for signaling and media packets to the IPv6 address of a specific interface.

 
Example: Configuring the Source IPv6 Address of Signaling and Media Packets
Device(config)# voice service voip
Device(config-voi-serv)# sip
Device(config-serv-sip)# bind control source-interface fastEthernet 0/0

Configuring the SIP Server

SUMMARY STEPS

1.    enable

2.    configure terminal

3.    sip-ua

4.    sip-server {dns: host-name] | ipv4: ipv4-address | ipv6: [ipv6-address] :[port-nums]}

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


DETAILED STEPS
  Command or Action Purpose
Step 1
enable


Example:

Device> enable

 

Enables privileged EXEC mode.

  • Enter your password if prompted.
 
Step 2
configure terminal


Example:

Device# configure terminal

 

Enters global configuration mode.

 
Step 3
sip-ua


Example:

Device(config)# sip-ua

 

Enters SIP user agent configuration mode.

 
Step 4
sip-server {dns: host-name] | ipv4: ipv4-address | ipv6: [ipv6-address] :[port-nums]}

Example:

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

 

Configures a network address for the SIP server interface.

 
Step 5
keepalive target {{ipv4 : address | ipv6 : address}[: port] | dns : hostname } [ tcp [ tls ]] | udp] [secondary]


Example:

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

 

Identifies SIP servers that will receive keepalive packets from the SIP gateway.

 
Example: Configuring the SIP Server
Device(config)# sip-ua
Device(config-sip-ua)# sip-server ipv6:[2001:DB8:0:0:8:800:200C:417A]

Configuring the Session Target

Perform this task to configure the session target.

SUMMARY STEPS

1.    enable

2.    configure terminal

3.    dial-peer voice tag {mmoip | pots | vofr | voip}

4.    destination pattern [+ string T

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


DETAILED STEPS
  Command or Action Purpose
Step 1
enable


Example:

Device> enable

 

Enables privileged EXEC mode.

  • Enter your password if prompted.
 
Step 2
configure terminal


Example:

Device# configure terminal

 

Enters global configuration mode.

 
Step 3
dial-peer voice tag {mmoip | pots | vofr | voip}


Example:

Device(config)# dial-peer voice 29 voip

 

Defines a particular dial peer, specifies the method of voice encapsulation, and enters dial peer configuration mode.

 
Step 4
destination pattern [+ string T


Example:

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

 

Specifies either the prefix or the full E.164 telephone number to be used for a dial peer.

 
Step 5
session target {ipv4: destination-address| ipv6: [ destination-address ]| dns : $s$. | $d$. | $e$. | $u$.] host-name | enum:table -num | loopback:rtp | ras| sip-server} [: port


Example:

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

 

Designates a network-specific address to receive calls from a VoIP or VoIPv6 dial peer.

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

Configuring SIP Register Support

SUMMARY STEPS

1.    enable

2.    configure terminal

3.    sip-ua

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

5.    retry register retries

6.    timers register milliseconds


DETAILED STEPS
  Command or Action Purpose
Step 1
enable


Example:

Device> enable

 

Enables privileged EXEC mode.

  • Enter your password if prompted.
 
Step 2
configure terminal


Example:

Device# configure terminal

 

Enters global configuration mode.

 
Step 3
sip-ua


Example:

Device(config)# sip-ua

 

Enters SIP user agent configuration mode.

 
Step 4
registrar {dns: address | ipv4: destination-address [: port] | ipv6: destination-address : port] } aor-domain expires seconds [tcp tls] ] type [secondary] [scheme string]


Example:

Device(config-sip-ua)# registrar ipv6:[2001:DB8::1:20F:F7FF:FE0B:2972] expires 3600 secondary

 

Enables SIP gateways to register E.164 numbers on behalf of analog telephone voice ports, IP phone virtual voice ports, and SCCP phones with an external SIP proxy or SIP registrar.

 
Step 5
retry register retries


Example:

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

 

Configures the total number of SIP register messages that the gateway should send.

 
Step 6
timers register milliseconds


Example:

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

 

Configures how long the SIP UA waits before sending register requests.

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

Configuring Outbound Proxy Server Globally on a SIP Gateway

SUMMARY STEPS

1.    enable

2.    configure terminal

3.    voice service voip

4.    sip

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


DETAILED STEPS
  Command or Action Purpose
Step 1
enable


Example:

Device> enable

 

Enables privileged EXEC mode.

  • Enter your password if prompted.
 
Step 2
configure terminal


Example:

Device# configure terminal

 

Enters global configuration mode.

 
Step 3
voice service voip


Example:

Device(config)# voice service voip

 

Enters voice service VoIP configuration mode.

 
Step 4
sip


Example:

Device(config-voi-serv)# sip

 

Enters sip configuration mode.

 
Step 5
outbound-proxy {ipv4: ipv4-address | ipv6: ipv6-address | dns: host : domain} [: port-number]


Example:

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

 

Specifies the SIP outbound proxy globally for a Cisco IOS voice gateway using an IPv6 address.

 

Verifying SIP Gateway Status

SUMMARY STEPS

1.    show sip-ua calls

2.    show sip-ua connections

3.    show sip-ua status


DETAILED STEPS
Step 1   show sip-ua calls

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

Device# show sip-ua calls 
SIP UAC CALL INFO
	Call 1
	SIP Call ID : 8368ED08-1C2A11DD-80078908-BA2972D0@2001::21B:D4FF:FED7:B000
		State of the call       : STATE_ACTIVE (7)
		Substate of the call    : SUBSTATE_NONE (0)
		Calling Number          : 2000
		Called Number           : 1000
		Bit Flags               : 0xC04018 0x100 0x0
CC Call ID              : 2
   Source IP Address (Sig ): 2001:DB8:0:ABCD::1
   Destn SIP Req Addr:Port : 2001:DB8:0:0:FFFF:5060
   Destn SIP Resp Addr:Port: 2001:DB8:0:1:FFFF:5060
   Destination Name        : 2001::21B:D5FF:FE1D:6C00
   Number of Media Streams : 1
   Number of Active Streams: 1
   RTP Fork Object         : 0x0
   Media Mode              : flow-through
   Media Stream 1
     State of the stream      : STREAM_ACTIVE
     Stream Call ID           : 2
     Stream Type              : voice-only (0)
     Stream Media Addr Type   : 1709707780
     Negotiated Codec         :  (20 bytes)
     Codec Payload Type       : 18 
     Negotiated Dtmf-relay    : inband-voice
     Dtmf-relay Payload Type  : 0
     Media Source IP Addr:Port: [2001::21B:D4FF:FED7:B000]:16504
     Media Dest IP Addr:Port  : [2001::21B:D5FF:FE1D:6C00]:19548
Options-Ping    ENABLED:NO    ACTIVE:NO
   Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
   Number of SIP User Agent Server(UAS) calls: 0
Step 2   show sip-ua connections

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



Example:
Device# show sip-ua connections udp brief 
Total active connections      : 1
No. of send failures          : 0
No. of remote closures        : 0
No. of conn. failures         : 0
No. of inactive conn. ageouts : 0
Router# show sip-ua connections udp detail
 
Total active connections      : 1
No. of send failures          : 0
No. of remote closures        : 0
No. of conn. failures         : 0
No. of inactive conn. ageouts : 0
---------Printing Detailed Connection Report---------
Note:
 ** Tuples with no matching socket entry
    - Do 'clear sip <tcp[tls]/udp> conn t ipv4:<addr>:<port>'
      to overcome this error condition
 ++ Tuples with mismatched address/port entry
    - Do 'clear sip <tcp[tls]/udp> conn t ipv4:<addr>:<port> id <connid>'
      to overcome this error condition
Remote-Agent:2001::21B:D5FF:FE1D:6C00, Connections-Count:1
  Remote-Port Conn-Id Conn-State  WriteQ-Size
  =========== ======= =========== ===========
         5060       2 Established           0
Step 3   show sip-ua status

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



Example:
Device# show sip-ua status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED
SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): DISABLED 
SIP User Agent bind status(media): DISABLED 
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED
protocol mode is ipv6
SDP application configuration:
 Version line (v=) required
 Owner line (o=) required
 Timespec line (t=) required
 Media supported: audio video image 
 Network types supported: IN 
 Address types supported: IP4 IP6 
 Transport types supported: RTP/AVP udptl 

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

An organization with an IPv4 network can deploy a Cisco Unified Border Element on the boundary to connect with the service provider's IPv6 network (see the figure below).

Figure 2 Cisco Unified Border Element Interoperating IPv4 Networks with IPv6 Service Provider


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 (see the figure below).

Figure 3 IPv4 to IPv6 Media Interoperating Through Cisco IOS MTP


The Cisco Unified Border Element feature adds IPv6 capability to existing VoIP features. This feature adds dual-stack support on voice gateways and MTP, IPv6 support for SIP trunks, and SCCP-controlled analog voice gateways. In addition, the SBC functionality of connecting SIP IPv4 or H.323 IPv4 network to a SIP IPv6 network is implemented on an Cisco Unified Border Element to facilitate migration from VoIPv4 to VoIPv6.

Before You Begin

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


Note


A Cisco Unified Border Element interoperates between H.323/SIP IPv4 and SIP IPv6 networks only in media flow-through mode.



SUMMARY STEPS

1.    enable

2.    configure terminal

3.    voice service voip

4.    allow-connections from type to to type


DETAILED STEPS
  Command or Action Purpose
Step 1
enable


Example:

Device> enable

 

Enables privileged EXEC mode.

  • Enter your password if prompted.
 
Step 2
configure terminal


Example:

Device# configure terminal

 

Enters global configuration mode.

 
Step 3
voice service voip


Example:

Device(config)# voice service voip

 

Enters voice service VoIP configuration mode.

 
Step 4
allow-connections from type to to type


Example:

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

 

Allows connections between specific types of endpoints in a VoIPv6 network.

Arguments are as follows:

  • from-type --Type of connection. Valid values: h323, sip.
  • to-type --Type of connection. Valid values: h323, sip.
 

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

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

Configuring MTP Used with Voice Gateways

Cisco IOS MTP trusted relay point (TRP) supports media interoperation between IPv4 and IPv6 networks (see the figure below). This functionality is used when an IPv4 phone (registered to Cisco Unified Communications Manager, formerly known as Cisco Unified Call Manager) communicates with an IPv6 phone (registered to another Cisco Unified Communications Manager). In this case, one of the Cisco Unified Communications Managers inserts a Cisco IOS MTP to perform the IPv4-to-IPv6 media translation between the phones.

MTP for IPv4-to-IPv6 media translation operates only in dual-stack mode. Communication between Cisco IOS MTP and Cisco Unified Communications Manager occurs over SCCP for IPv4 only.

Figure 4 IPv4 to IPv6 Media Interoperating Through Cisco IOS MTP


The VoIPv6 feature includes IPv4 and IPv6 dual-stack support on voice gateways and MTP, IPv6 support for SIP trunks, and SCCP-controlled analog phones. In addition, connecting a SIP IPv4 or H.323 IPv4 network to a SIP IPv6 network is implemented on Cisco Unified Border Element.

Configuring MTP for IPv4-to-IPv6 Translation

MTP for IPv4-to-IPv6 media translation operates in dual-stack mode only. A SIP trunk can be configured over IPv4 only, over IPv6 only, or in dual-stack mode. In dual-stack mode, ANAT is used to describe both IPv4 and IPv6 media capabilities.

SUMMARY STEPS

1.    enable

2.    configure terminal

3.    sccp ccm {ipv4-address | ipv6-address | dns} identifier identifier-number [priority priority] [port port-number] [version version-number]

4.    sccp ccm group group -number

5.    associate profile profile-identifier register device -name

6.    exit

7.    dspfarm profile profile -identifier {conference | mtp | transcode} [security]

8.    codec {codec-type | pass-through}

9.    maximum sessions {hardware | software} number

10.    associate application sccp


DETAILED STEPS
  Command or Action Purpose
Step 1
enable


Example:

Device> enable

 

Enables privileged EXEC mode.

  • Enter your password if prompted.
 
Step 2
configure terminal


Example:

Device# configure terminal

 

Enters global configuration mode.

 
Step 3
sccp ccm {ipv4-address | ipv6-address | dns} identifier identifier-number [priority priority] [port port-number] [version version-number]


Example:

Device(config)# sccp ccm 2001:DB8:C18:1::102 identifier 2 version 7.0

 

Adds a Cisco Unified CallManager server to the list of available servers and set various parameters--including IP address, IPv6 address, or Domain Name System (DNS) name, port number, and version number.

Note    SCCP communication between Cisco IOS MTP and Cisco Unified Border Element is supported only for an IPv4-only network. Do not use the ipv6-address argument with this command if you are configuring for the Cisco Unified Border Element.
 
Step 4
sccp ccm group group -number


Example:

Device(config)# sccp ccm group 1

 

Creates a Cisco CallManager group and enters SCCP Cisco CallManager configuration mode

 
Step 5
associate profile profile-identifier register device -name


Example:

Device(conif-sccp-ccm)# associate profile 5 register MTP3825

 

Associates a digital signal processor (DSP) farm profile with a Cisco CallManager group.

 
Step 6
exit


Example:

Device(config-sip-ua)# exit

 

Exits the current configuration mode.

 
Step 7
dspfarm profile profile -identifier {conference | mtp | transcode} [security]


Example:

Device(config)# dspfarm profile 5 mtp

 

Enters DSP farm profile configuration mode and defines a profile for DSP farm services.

 
Step 8
codec {codec-type | pass-through}


Example:

Device(config-dspfarm-profile)# codec g711ulaw

 

Specifies the codecs that are supported by a DSP farm profile.

 
Step 9
maximum sessions {hardware | software} number


Example:

Device(config-dspfarm-profile)# maximum sessions software 100

 

Specifies the maximum number of sessions that are supported by the profile.

 
Step 10
associate application sccp


Example:

Device(config-dspfarm-profile)# associate application sccp

 

Associates SCCP to the DSP farm profile.

 
Example: Configuring MTP for IPv4-to-IPv6 Translation
Device(config)# sccp ccm group 1
Device(config-sccp-ccm)#associate profile 5 register MTP3825
Device(config-sccp-ccm)# exit
Device(config)# dspfarm profile 5 mtp
Device(config-dspfarm-profile)# codec g711ulaw
Device(config-dspfarm-profile)# maximum sessions software 100
Device(config-dspfarm-profile)# associate application sccp

Device# show sccp
sccp ccm group 1
associate profile 5 register MTP3825
!
dspfarm profile 5 mtp
 codec g711ulaw
 maximum sessions software 100	
 associate application SCCP

Feature Information for VoIP for IPv6

The following table provides release information about the feature or features described in this module. This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Table 1 Feature Information for VoIP for IPv6
Feature Name Releases Feature Information

Cisco UBE support for IPv6

12.4(22)T

Cisco UBE support for SIP IPv4-IPv6 dual stack and IPv4 and IPv6 capability provides the following functionality:

  • Translation of SIP IPv4 to IPv6 addresses
  • Administration and enforcement of policies for the IPv4/IPv6 mode of operation of each component.
  • Support the following scenarios: H.323 IPv4 to SIP IPv6; SIP IPv4 to SIP IPv6, SIP IPv6 to SIP IPv6
  • DTMF: Interworking capability on Cisco UBE (H.245 Signal, RFC 2833, SIP Notify, Key Press Markup Language,H.323 to SIP, RFC 2833 to G.711 Inband)
  • IPv6 topology hiding and demarcation
  • SIP Options-ping

DSCP-Based QoS Support

12.4(22)T

IPv6 supports this feature.

IPv6 Dual Stack

12.4(22)T

Adds IPv6 capability to existing VoIP features on the Cisco Unified Border Element. Additionally, the SBC functionality of connecting SIP IPv4 or H.323 IPv4 network to SIP IPv6 network is implemented on a Cisco Unified Border Element to facilitate migration from VoIPv4 to VoIPv6.

The following commands were introduced or modified: None

IPv6 Dual Stack

Cisco IOS XE Release 3.3S

Adds IPv6 capability to existing VoIP features on the Cisco Unified Border Element (Enterprise). Additionally, the SBC functionality of connecting SIP IPv4 or H.323 IPv4 network to SIP IPv6 network is implemented on a Cisco Unified Border Element to facilitate migration from VoIPv4 to VoIPv6.

The following commands were introduced or modified: None

RTP/RTCP over IPv6

12.4(22)T

RTP stack supports the ability to create IPv6 connections using IPv6 unicast and multicast addresses as well as IPV4 connections.

TDM-SIP GW for IPv6

12.4(24)T

IPv6 supports this feature.

Voice Gateway/MTP

12.4(22)T

Support for If an MTP (Media Translation Point) is used for SIP IPv4/IPv6 media translation.

The following commands were introduced or modified: None

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Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be actual addresses and phone numbers. Any examples, command display output, network topology diagrams, and other figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses or phone numbers in illustrative content is unintentional and coincidental.

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