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.
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.
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.
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.configureterminal
3.voiceservicevoip
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
configureterminal
Example:
Device# configure terminal
Enters global configuration mode.
Step 3
voiceservicevoip
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.configureterminal
3.voiceservicevoip
4.sip
5.callservicestop [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
configureterminal
Example:
Device# configure terminal
Enters global configuration mode.
Step 3
voiceservicevoip
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
callservicestop [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.
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.
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.configureterminal
3.voiceservicevoip
4.sip
5.noanat
DETAILED STEPS
Command or Action
Purpose
Step 1
enable
Example:
Device> enable
Enables privileged EXEC mode.
Enter your password if prompted.
Step 2
configureterminal
Example:
Device# configure terminal
Enters global configuration mode.
Step 3
voiceservicevoip
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
noanat
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.configureterminal
3.voiceservicevoip
4.sip
5.bind {control |
media |
all}
sourceinterfaceinterface-id [ipv6-addressipv6-address
DETAILED STEPS
Command or Action
Purpose
Step 1
enable
Example:
Device> enable
Enables privileged EXEC mode.
Enter your password if prompted.
Step 2
configureterminal
Example:
Device# configure terminal
Enters global configuration mode.
Step 3
voiceservicevoip
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}
sourceinterfaceinterface-id [ipv6-addressipv6-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
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
retryregisterretries
Example:
Device(config-sip-ua)# retry register 10
Configures the total number of SIP register messages that the gateway should send.
Step 6
timersregistermilliseconds
Example:
Device(config-sip-ua)# timers register 500
Configures how long the SIP UA waits before sending register requests.
Specifies the SIP outbound proxy globally for a Cisco IOS voice gateway using an IPv6 address.
Verifying SIP Gateway Status
SUMMARY STEPS
1.showsip-uacalls
2.showsip-uaconnections
3.showsip-uastatus
DETAILED STEPS
Step 1
showsip-uacalls
The
showsip-uacalls 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
showsip-uaconnections
Use the
showsip-uaconnections 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
showsip-uastatus
Use the
showsip-uastatus 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.configureterminal
3.voiceservicevoip
4.allow-connectionsfromtypetototype
DETAILED STEPS
Command or Action
Purpose
Step 1
enable
Example:
Device> enable
Enables privileged EXEC mode.
Enter your password if prompted.
Step 2
configureterminal
Example:
Device# configure terminal
Enters global configuration mode.
Step 3
voiceservicevoip
Example:
Device(config)# voice service voip
Enters voice service VoIP configuration mode.
Step 4
allow-connectionsfromtypetototype
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.
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.
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
sccpccmgroupgroup-number
Example:
Device(config)# sccp ccm group 1
Creates a Cisco CallManager group and enters SCCP Cisco CallManager configuration mode
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
Cisco IOS XE Release 3.8S
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
Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S. and other countries. To view a list of Cisco trademarks, go to this URL:
www.cisco.com/go/trademarks. Third-party trademarks mentioned are the property of their respective owners. The use of the word partner does not imply a partnership relationship between Cisco and any other company. (1110R)
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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