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

VoIP for IPv6

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.

For more information on implementing VoIP for IPv6 support, refer to the Implementing VoIP for IPv6 chapter in the IPv6 Implementation Guide.

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest feature information and caveats, see 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 document.

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.

Restrictions for VoIP for IPv6

Media Flow-Through

  • Video call flows with Alternative Network Address Types (ANAT) are not supported.
  • WebEx call flow with ANAT are not supported (Cisco UBE does not support ANAT on Video and Application media types).

Media Flow-Around

  • Media Anti-Trombone will not start if the initial call before tromboning is in Flow-Around (FA) mode. For media anti-tromboning, the initial call should be in Flow-Through (FT) and post tromboning it will move to FA.
  • Call transfer/forward flows when media moves from FT to FA or vise versa (SNR Feature Callflows).
  • When a transcoder is inserted, the call moves from FA to FT.
  • For Dual-Tone Multi-Frequency Signaling (DTMF) interworking, the call moves from FA to FT.

SDP Pass-Through

  • SDP pass-through is supported only for Early Offer (EO)-Early Offer (EO) and Delayed Offer (DO)-Delayed Offer (DO) call flows.
  • DO-EO call flow falls back to DO-DO call flow.
  • Supplementary services are not supported.
  • Transcoding, DTMF interworking are not be supported.
  • Dual-stack configuration is a no-op as the SDP received on the peer leg is passed to the other leg; CUBE just replaces the SDP source IP address and port on the out leg.
  • Media interworking is not supported for ANAT call flows; media will end up as IPv4<->IPv4 or IPv6<->IPv6 (IPv4<->IPv6 and IPv6<->IPv4 media interworking not possible).

UDP Checksum for Media

  • “cef” and “process” options not applicable for ASR1000 (no-op).
  • “none” option does not work well on ISR-G2.

Media Anti-Trombone

  • Supports only symmetric media address type interworking (IPv4-IPv4 or IPv6-IPv6 media) with or without ANAT.
  • Does not provide support for IPv4-IPv6 interworking cases with or without ANAT because Cisco UBE cannot operate in FA mode post tromboning.

Information About VoIP for IPv6

Cisco Unified Border Element Features Supported on IPv6

Cisco Unified Border Element (CUBE) being a signaling proxy, it also processes all signaling messages regarding the setup of media channels. This enables CUBE to affect the flow of media traffic using the media flow-through and the media flow-around options.

CUBE features support the following Basic Audio Calls:

  • Media Flow-Through (FT) Audio Calls: When using media flow-through, Cisco UBE replaces the source IP address used for media connections with its own IP address. This makes Cisco UBE with media flow-through ideal for interworking with external VoIP networks and enforcing a tighter security policy. The signaling between the two Cisco Unified Communications Manager clusters is processed by Cisco UBE, and the source IP addresses of the endpoints are replaced by the Cisco UBE IP address. Both endpoints have the same IP address, but because Cisco UBE is involved, no interworking issues arise. SIP-SIP flow-through audio calls are supported in three modes of operation—IPv4-only, IPv6-only and dual-stack with or without ANAT. IPv4 to IPv6 interworking calls can negotiate IPv4 on one leg and IPv6 on the other leg. Media flow-through supports the following call flows:
    • Early Offer (EO) <-> Early Offer (EO): In an EO-EO FT ANAT call flow scenario, when you configure CUBE with dual-stack prefix IPv4, the outing ANAT offer has 1st mline as IPv4 and 2nd mline as IPv6. However, when you configure CUBE with dual-stack preference IPv6, the outoing offer towards UAS will have the 1st mline as IPv6 and the 2nd mline as IPv4.
    • Delayed Offer (DO) <-> Delayed Offer (DO): In a DO-DO FT ANAT call scenario, when you configure Cube with dual stack prefix ipv4, CUBE sends the outgoing ANAT offer in 200OK with IPv4 in the 1st mline and IPv6 in the 2nd mline. Additionally, if CUBE adds supported header sdp-anat and if the UAS is unable to interprete ANAT, the outoing offer towards UAS becomes a single stream or a mline, and then, the DO-DO FT ANAT call becomes ANAT to Non–ANAT call.
    • Delayed Offer (DO) <-> Early Offer (EO): In a DO-EO FT ANAT call scenario, when a CUBE configured with ANAT and an early-offer-forced Command-Line-Interface (CLI) receives a DO INVITE from UAC, the call creates a ANAT offer towards UAS with IPv4 on the 1st mline and IPv6 on the 2nd mline. As a result, CUBE receives an ANSWER from UAS and immediately sends an acknowledgement (ACK) so that the OFFER or ANSWER is complete on the CUBE to UAS side. Similarly, CUBE generates an offer towards UAC and sends it in 200OK response. Once ACK with ANSWER is received from the UAC, CUBE bridges the call legs and offer or answer is complete on CUBE to UAC side.
  • Media Flow-Around (FA): When using media flow-around, Cisco UBE leaves the IP addresses used for the media connections untouched. Call signaling is still processed by Cisco UBE, but after the call is set up, Cisco UBE is no longer involved with the traffic flow. This enables media packets to be passed directly between endpoints without the intervention of Cisco UBE. Media flow-around improves scalability and performance of audio calls when network-topology hiding and bearer-level interworking features are not required. In a media flow-around call flow, CUBE receives ANAT offer with a codec payload-type and DTMF payload-type so that the same information is passed to the peer leg. On the other leg, CUBE keeps all SDP parameters intact except that it changes the media ip address, media port and inserts its own source ip address and source port. Similarly, for an ANSWER negotiation, Non-ANAT DO-EO FA calls can still work as FA.SIP-SIP flow-around audio calls are supported in three modes of operation—IPv4-only, IPv6-only and dual-stack with or without ANAT: Media flow-around supports the following call flows:
    • Early Offer (EO) <-> Early Offer (EO): In an EO-EO Flow Around ANAT call flow, CUBE, which is in a dual-stack mode with prefix ipv4, receives a v4 mline with a port, for example a and another port b for a v6 mline. Now, since CUBE is in an FA mode, it does not give any preference to the dual-stack configuration, but copies the configuration from a peer leg and builds an Offer with the same IP address and ports the configuration without any changes on another peer leg. As a result, after an ANSWER is received from an UAS, CUBE copies the media address, ports and passes the information to a peer leg. Subsequently, the peer leg copies the same information and builds the ANSWER and sends it towards the UAC.
    • Delayed Offer (DO) <-> Delayed Offer (DO): In an EO-EO Flow Around ANAT call, CUBE, which is in a dual-stack mode with prefix IPv4, receives a DO INVITE with a supported: sdp-anat header. With the help of this header, CUBE sends a DO INVITE to UAS, and the UAS then sends an Offer with v4 and v6 mlines. CUBE copies the media address and media port information from the peer leg and builds an OFFER sdp towards UAC in 200OK response. After getting an ANSWER as an Acknowledgement (ACK) from the UAS, CUBE copies the media address and the port, and then passes the information to the peer leg. Then, the peer leg copies the same information and builds the ANSWER sdp and sends it to the UAC.
    • Delayed Offer (DO) <-> Early Offer (EO): In an DO-EO Flow Around ANAT call, CUBE, which is in the FA mode with dual-stack and ANAT enabled, receives the DO with a SUPPORTED:SDP-ANAT and then CUBE automatically switches over the call from Flow-Around to Flow-Through. As a result, CUBE generates a local ANAT offer with v4 and v6 address in the outgoing INVITE. After the call moves from Flow-Around to Flow-Through, the offer-answer negotiation is from one leg to another and there is no dependency of peer address on either side of the CUBE.
  • Assisted RTCP (RTCP Keepalive): Assisted Real-time Transport Control Protocol (RTCP) enables CUBE to generate RTCP keepalive reports on behalf of endpoints; however, endpoints, such as second generation Cisco IP phones (7940/7960) and Nortel Media Gateways (MG 1000T) do not generate any RTCP keepalive reports. Assisted RTCPs enable customers to use CUBE to interoperate between endpoints and call control agents, such as Microsoft OCS/Lync so that RTCP reports are generated to indicate session liveliness during periods of prolonged silence, such as call hold or on mute. The assisted RTCP feature helps Cisco Unified Border Element (Cisco UBE) to generate standard RTCP keepalive reports on behalf of endpoints. RTCP reports determine the liveliness of a media session during prolonged periods of silence, such as a call on hold or a call on mute.
  • SDP Pass-Through: The SDP Pass-Through feature introduces the ability to configure the Cisco UBE to pass through end to end headers at a global or dial-peer level that are not processed or understood in a SIP trunk to SIP trunk scenario. The pass through functionality includes all or only a few unsupported or nonmandatory SIP headers, and all unsupported content/MIME types. The feature supports all three modes of operation—IPv4-only, IPv6-only and dual-stack with/without ANAT. SDP Pass-Through supports the following call flows:
    • EO-EO FT SDP Pass-through ANAT Call: In an EO-EO FT SDP Pass-through ANAT call, CUBE receives ANAT offer with codec payload-type, for example 18 and dtmf payload-type—101, so the same information is passed to the peer leg, on the other leg CUBE keeps all the SDP parameters intact except it will change the media ip address, media port and insert its own source ip address and source port.
    • EO-EO FA SDP Pass-through ANAT Call: In an EO-EO FA SDP Pass-through ANAT call, CUBE receives an ANAT offer with codec payload-type, for example 18 and dtmf payload-type-101, so that the same information passes to the peer leg. As the CUBE is in FA with SDP pass-thru mode, it does not touch the SDP, but copies the information from the peer leg received SDP and builds its source SDP and Offers an outing INVITE towards the UAS. Similarly, for an ANSWER negotiation, there is no media negotiation in CUBE, signals flow through CUBE and media flows between UAC and UAS directly without any CUBE intervention.
  • UDP Checksum for Media: When a User Datagram Protocol (UDP) packet originates from an IPv6 node, UDP checksum is mandatory.
  • IP Toll Fraud: If the source IP address does not match an explicit configuration entry as a trusted VoIP source, the call is rejected. In such a scenario, a 403 forbidden HTTP status code notification is sent from the web server.
  • RTP Port Range: Provides the capability where the port range is managed per IP address range. This features solves the problem of limited number of rtp ports for more than 4000 calls. It enables combination of an IP address and a port as a unique identification for each call.

CUBE features support the following Supplementary Services:

  • Hold/Resume: Digital Signal Processors (DSPs) generate and transmit Real-time Transport Protocol (RTP) media packets from a source to a destination address during a SIP call session. However, when a SIP call is put on hold, DSPs stop generating the RTP media packets and resumes generating and transmitting RTP media packets after the SIP call has resumed. This ensures that the RTP sequence number is continuous from the time of origin until the end of the SIP call.
  • Call Transfer (re-INVITE, REFER, 302 based)
  • Media Anti-Trombone: Anti-tromboning is a media signaling service in SIP entity to overcome the media loops. Antitrombone service has to be enabled only when no media interworking is required in both the out-legs.
  • RE-INVITE Consumption
  • Supplementary Services with Audio Transcoding using Local Transcoding Interface (LTI)

CUBE features support the following generic features:

  • Address Hiding
  • Header Passing
  • Refer-To Passing
  • Error Pass-through
  • SIP UPDATE Interworking
  • SIP Session timer (RFC 4028)
  • SIP OPTIONS Ping
  • Configurable Error Response Code in OPTIONS Ping
  • Limiting the Rate of Incoming SIP Calls per Dial-Peer (aka Call Spike)
  • SIP Profiles
  • SIP Media Inactivity Detection
  • Dynamic Payload Type Interworking (DTMF and Codec Packets)
  • Audio Transcoding using Local Transcoding Interface (LTI)
  • Voice Class Codec (VCC) with/without Transcoding
  • PPI/PAI/Privacy and RPID Passing

Session Initiation Protocol Gateway Features Supported on IPv6

Session Initiation Protocol (SIP) Gateway feature supports visible message waiting indication (VMWI) on FXS phones. This feature provides users with the option to enable one message waiting indication (MWI): audible, visible, or both.

This feature is supported for analog endpoints that are connected to Foreign Exchange Station (FXS) ports or a Cisco VG224 Analog Phone Gateway and controlled by a Cisco call-control system, such as a Cisco Unified Communications Manager (Cisco Unified CM) or a Cisco Unified Communications Manager Express (Cisco Unified CME). The VMWI mechanism uses SIP Subscribe or Notify to get MWI updates from a VM system, and then forwards the updates to the FXS phone on the port.

  • Some of the features supported on SIP VMWI are:
  • SIP 302 Message: The SIP 302 message is used to redirect the SIP call.
  • 181(call is being forwarded)/183 Messages(Session Progress): 181/183 are provisional responses. Shows request that are received and those being processed.
  • PPI/PAI: Provides support for RFC 3323 and RFC 3325 that allow you to enable either P-Asserted-Identity (PAI) or P-Preferred-Identity (PPI) privacy headers in outgoing SIP request or response messages to assert the identity of authenticated users in trusted domains.
  • Media Inactivity Timer: The Media Inactivity Timer is used to indicate that RTP packets have stopped flowing for the configured amount of time. An event is generated to the signaling layers and the signals releases the channel.

Session Initiation Protocol Features Supported on IPv6

The Session Initiation Protocol (SIP) is an alternative protocol developed by the Internet Engineering Task Force (IETF) for multimedia conferencing over IP. SIP features are compliant with IETF RFC 2543, SIP: Session Initiation Protocol, published in March 1999.

The Cisco SIP functionality enables Cisco access platforms to signal the setup of voice and multimedia calls over IP networks. The SIP feature also provides nonproprietary advantages in the following areas:
  • Protocol extensibility
  • System scalability
  • System scalability
  • Personal mobility services
  • Interoperability with different vendors

A Session Initiation Protocol (SIP) User Agent (UA) can operate in one of the three modes:

  • IPv4-only: Communication with only IPv6 UA is unavailable.
  • IPv6-only: Communication with only IPv4 UA is unavailable.
  • Dual-stack: Communication with only IPv4, only IPv6 and dual-stack UAs are available.

Dual-stack SIP UAs use Alternative Network Address Transport (ANAT) grouping semantics:

  • Includes both IPv4 and IPv6 addresses in the Session Description Protocol (SDP).
  • Is automatically enabled in dual-stack mode (can be disabled if required).
  • Requires media to be bound to an interface having both IPv4 and IPv6 addresses.
  • Is described in RFC 4091 and RFC 4092 (RFC 5888 describes general SDP grouping framework).

SIP UAs use “sdp-anat” option tag in the Required and Supported SIP header fields:

  • Early Offer (EO) INVITE using ANAT semantics places “sdp-anat” in the Require header.
  • Delayed Offer (DO) INVITE places “sdp-anat” in the Supported header.

Source address for SIP signaling is selected based on the destination signaling address type configured in the session-target of the outbound dial-peer:

  • If signaling bind is configured, source SIP signaling address is chosen from the bound interface.
  • If signaling bind is not configured, source SIP signaling address is chosen based on the best address in the UA to reach the destination signaling address.

SDP may or may not use ANAT semantics:

  • When ANAT is used, media addresses in SDP are chosen from the interface media that is configured. When ANAT is not used, media addresses in SDP are chosen from the interface media that is configured OR based on the best address to reach the destination signaling address (when no media bind is configured).

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.

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 ActionPurpose
    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 ActionPurpose
      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 ActionPurpose
        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 ActionPurpose
          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 ActionPurpose
            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 ActionPurpose
              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 ActionPurpose
                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 ActionPurpose
                  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 ActionPurpose
                    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 ActionPurpose
                        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

                        Troubleshooting Tips for VoIP for IPv6

                        Media Flow-Through

                        To enable all Session Initiation Protocol (SIP)-related debugging, use the debug ccsip all command in privileged EXEC mode.

                        To trace the execution path through the call control application programming interface (CCAPI), use the debug voip ccapi inout command.

                        Media Flow-Around

                        To enable all Session Initiation Protocol (SIP)-related debugging, use the debug ccsip all command.

                        To trace the execution path through the call control application programming interface (CCAPI), use the debug voip ccapi inout command.

                        SDP Pass-Through

                        To enable all Session Initiation Protocol (SIP)-related debugging (when the call is active in Pass through mode), use the debug ccsip all command.

                        RTP Port Range

                        To enable all Session Initiation Protocol (SIP)-related debugging, use the debug ccsip all command.

                        To enable debugging for Real-Time Transport Protocol (RTP) named event packets, use the debug voip rtp command.

                        VMWI SIP

                        To collect debug information only for signaling events, use the debug vpm signal command.

                        To show all Session Initiation Protocol (SIP) Service Provider Interface (SPI) message tracing, use the debug ccsip messages command.

                        Verification of Basic Audio Calls and Supplementary Services (CUBE and SIP Gateway)

                        To verify that media setting is enabled in the Media FT and Media FA feature; and CoderTypeRate, CodecBytes, media settings are enabled in the SDP Pass-through feature, use the following commands:

                        SUMMARY STEPS

                          1.    show call active voice

                          2.    show call active voice brief

                          3.    show call active voice compact

                          4.    show voip rtp connection

                          5.    show sip-ua mwi


                        DETAILED STEPS
                          Step 1   show call active voice


                          Example: 
                          Device# show call active voice | include Media Setting
                          Step 2   show call active voice brief


                          Example:
                          Device# show call active voice brief
                          Step 3   show call active voice compact


                          Example:
                          Device# show call active voice compact
                          Step 4   show voip rtp connection


                          Example:
                          Device# show voip rtp connection

VoIP RTP Port Usage Information:
Max Ports Available: 24273, Ports Reserved: 303, Ports in Use: 2   
Port range not configured, Min: 16384, Max: 32767   
                                                Ports       Ports       Ports
Media-Address Range                             Available   Reserved    In-use

Default Address-Range                           8091        101         0
2001::
2002::                                          8091        101         1
9.0.0.0              10.0.0.0                   8091        101         1
Found 2 active RTP connections  
....
                          Step 5   show sip-ua mwi


                          Example:
                          Device# show sip-ua mwi

 MWI type: 2
 MWI server: 2001:DB8:12:1::2002   
 MWI expires: 3600
 MWI port: 5060
 MWI dial peer tag: 0    
 MWI transport type: UDP
 MWI solicited                           
 MWI ipaddr cnt 1:
 MWI ipaddr idx 0:
 MWI server: 2001:DB8:12:1::2002, port 5060, transport 1   
 MWI server dns lookup retry cnt: 0

                          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

                          15.3(2)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

                          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

                          15.3(2)T

                          IPv6 supports this feature.

                          • Session Initiation Protocol Features Supported on IPv6
                          • Cisco UBE features Supported on IPv6
                          • SIP Gateway Generic Features