SIP Configuration Guide, Cisco IOS Release 15M&T
SIP Binding
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SIP Binding

SIP Binding

The SIP Binding feature enables you to configure a source IP address for signaling packets and media packets.

Feature Information for SIP Binding

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 SIP Binding

Feature Name

Releases

Feature Information

SIP Gateway Support for the bind Command

Cisco IOS 12.2(2)XB, 12.2(2)XB2, 12.2(8)T, 12.2(11)T, and 12.3(4)T

Cisco IOS XE 3.1.0S

The SIP Gateway Support for the bind Command feature allows you to configure the source IP address of signaling packets and media packets.

In 12.2(2)XB, this feature was introduced.

In 12.3(4)T, this feature was expanded to provide the flexibility to specify different source interfaces for signaling and media, and allow network administrators a finer granularity of control on the network interfaces used for voice traffic.

The following commands were introduced or modified: bind, show dial-peer voice, show ip sockets, show sip-ua connections, and show sip-ua status.

Support for Ability to Configure Source IP Address for Signaling and Media per SIP Trunk

15.1(2)T

This feature allows you to configure a separate source IP address per SIP trunk. This source IP address is embedded in all SIP signaling and media packets that traverse the SIP trunk. This feature enables service providers for better profiling and billing policies. It also enables greater security for enterprises by the use of distinct IP addresses within and outside the enterprise domain.

The following command was introduced or modified: voice-class sip bind.

Information About SIP Binding

When you configure SIP on a router, the ports on all its interfaces are open by default. This makes the router vulnerable to malicious attackers who can execute toll fraud across the gateway if the router has a public IP address and a public switched telephone network (PSTN) connection. To eliminate the threat, you should bind an interface to an IP address so that only those ports are open to the outside world. In addition, you should protect any public or untrusted interface by configuring a firewall or an access control list (ACL) to prevent unwanted traffic from traversing the router.

Benefits of SIP Binding

  • SIP signaling and media paths can advertise the same source IP address on the gateway for certain applications, even if the paths used different addresses to reach the source. This eliminates confusion for firewall applications that may have taken action on source address packets before the use of binding.

  • Firewalls filter messages based on variables such as the message source, the target address, and available ports. Normally a firewall opens only certain addresses or port combination to the outside world and those addresses can change dynamically. Because VoIP technology requires the use of more than one address or port combination, the bind command adds flexibility by assigning a gateway to a specific interface (and therefore the associated address) for the signaling or media application.

  • You can obtain a predefined and separate interface for both signaling and media traffic. After a bind command is in effect, the interface it limits is bound solely to that purpose. Administrators can therefore dictate the use of one network to transport the signaling and another network to transport the media. The benefits of administrator control are:
    • Administrators know the traffic that runs on specific networks, thereby making debugging easier.
    • Administrators know the capacity of the network and the target traffic, thereby making engineering and planning easier.
    • Traffic is controlled, allowing Qualtiy of Service (QoS) to be monitored.
  • The bind media command relaxes the constraints imposed by the bind control and bind all commands, which cannot be set during an active call. The bind media command works with active calls.

Source Address

In early releases of Cisco IOS software with SIP functionality, the source address of a packet going out of the gateway was never deterministic. That is, the session protocols and VoIP layers always depended on the IP layer to give the best local address . The best local address was then used as the source address (the address showing where the SIP request came from) for signaling and media packets. Using this non-deterministic address occasionally caused confusion for firewall applications, because a firewall could not be configured with an exact address and would take action on several different source address packets.

However, the bind command enables you to configure the source IP address of signaling and media packets to a specific interface’s IP address. Thus, the address that goes out on the packet is bound to the IP address of the interface specified with the bind command. Packets that are not destined to the bound address are discarded.

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

The Support Ability to Configure Source IP Address for Signaling and Media per SIP Trunk feature extends the global bind functionality to support the SIP signaling Transport Layer Socket (TLS) with UDP and TCP. The source address at the dial peer is the source address in all the signaling and media packets between the gateway and the remote SIP entity for calls using the dial-peer. Multiple SIP listen sockets with specific source address handle the incoming SIP traffic from each selected SIP entity. The order of preference for retrieving the SIP signalling and media source address for inbound and outbound calls is as follows:

  • Bind configuration at dial peer level

  • Bind configuration at global level

  • Best local IP address to reach the destination

The table below describes the state of the system when the bind command is applied in the global or dial peer level:

Table 2 State of the System for the bind Address

Bind State

System Status

No global bind

The best local address is used in all outbound SIP messages.

Only one SIP listen socket with a wildcard source address.

Global bind

Global bind address used in all outbound SIP messages.

Only one SIP listen socket with global bind address.

No global bind

Dial peer bind

Dial peer bind address is used in outbound SIP messages of this dial peer. The remaining SIP messages use the best local address.

One SIP listen socket with a wildcard source address.

Additional SIP listen socket for each different dial peer bind listening on the specific dial peer bind address.

Global bind

Dial peer bind

Dial peer bind address is used in outbound SIP messages of this dial peer. The remaining SIP messages use the global bind address.

One SIP listen socket with global bind address.

Additional SIP listen socket for each different dial peer bind command listening on the specific dial peer bind address.

The bind command performs different functions based on the state of the interface (see the table below).

Table 3 State of the Interface for the bind Command

Interface State

Result Using Bind Command

Shut down

With or without active calls

TCP, TLS, and User Datagram Protocol (UDP) socket listeners are initially closed. (Socket listeners receive datagrams addressed to the socket.)

Then the sockets are opened to listen to any IP address.

If the outgoing gateway has the bind command enabled and has an active call, the call becomes a one-way call with media flowing from the outgoing gateway to the terminating gateway.

The dial peer bind socket listeners of the interface are closed and the configuration turns inactive for all subsequent SIP messages.

No shut down

No active calls

TCP, TLS, and UDP socket listeners are initially closed. (Socket listeners receive datagrams addressed to the socket.)

Then the sockets are opened and bound to the IP address set by the bind command.

The sockets accept packets destined for the bound address only.

The dial peer bind socket listeners of the interface are reopened and the configuration turns active for all subsequent SIP messages.

No shut down

Active calls

TCP, TLS, and UDP socket listeners are initially closed.

Then the sockets are opened to listen to any IP address.

The dial peer bind socket listeners of the interface are reopened and the configuration turns active for all subsequent SIP messages.

Bound-interface IP address is removed.

TCP, TLS, and UDP socket listeners are initially closed.

Then the sockets are opened to listen to any address, because the IP address has been removed. This happens even when SIP was never bound to an IP address.

A message stating that the IP address has been deleted from the SIP bound interface is printed.

If the outgoing gateway has the bind command enabled and has an active call, the call becomes a one-way call with media flowing from the outgoing gateway to the terminating gateway.

The dial peer bind socket listeners of the interface are closed and the configuration turns inactive for all subsequent SIP messages.

The physical cable is pulled on the bound port or the interface layer is down.

TCP, TLS, and UDP socket listeners are initially closed.

Then the sockets are opened and bound to listen to any address.

When the pulled cable is replaced, the result is as documented for no shutdown interfaces.

The dial peer bind socket listeners of the interface are closed and the configuration turns inactive for all subsequent SIP messages.

A bind interface is shut down or its IP address is changed or the physical cable is pulled while SIP calls are active.

The call becomes a one-way call with media flowing in only one direction. It flows from the gateway where the change or shutdown took place, to the gateway where no change occurred. Thus, the gateway with the status change no longer receives media.

The call is then disconnected, but the disconnected message is not understood by the gateway with the status change, and the call is still assumed to be active.

If the bind interface is shutdown, the dial peer bind socket listeners of the interface are closed. If the IP address of the interface is changed, the socket listeners representing the bind command is opened with the available IP address of the interface and the configuration turns active for all subsequent SIP messages.

Note   

If there are active calls, the bind command does not take effect if it is issued for the first time or if another bind command is in effect. A message reminds you that there are active calls and that the change cannot take effect.

The bind command applied at the dial peer level can be modified only in the following situations:

  • Dial peer bind is disabled in the supported IOS configuration options.

  • Dial peer bind is removed when the bound interface is removed.

  • Dial peer bind is removed when the dial peer is removed.

Voice Media Stream Processing

The SIP Gateway Support Enhancements to the bind Command feature extends the capabilities of the bind command by supporting a deterministic network interface for the voice media stream. Before the voice media stream addition, the bind command supported a deterministic network interface for control (signaling) traffic or all traffic. With the SIP Gateway Support Enhancements to the bind Command feature, a finer granularity of control is achieved on the network interfaces used for voice traffic.

If multiple bind commands are issued in sequence—that is, if one bind command is configured and then another bind command is configured—a set interaction happens between the commands. The table below describes the expected command behavior.

Table 4 Interaction Between Previously Set and New bind Commands

Interface State

bind Command

Result Using bind Command

Without active calls

bind all

Generated bind control and bind media commands to override existing bind control and bind media commands.

bind control

Overrides existing bind control command.

bind media

Overrides existing bind media command.

With active calls

bind all or bind control

Blocks the command, and the following messages are displayed:

  • 00:16:39: There are active calls

  • 00:16:39: configure_sip_bind_command: The bind command change will not take effect

bind media

Succeeds and overrides any existing bind media command.

The bind all and bind control commands perform different functions based on the state of the interface.


Note


The bind all command only applies to global level, whereas the bind control and bind media command apply to global and dial peer. The table below applies to bind media only if the media interface is the same as the bind control interface. If the two interfaces are different, media behavior is independent of the interface state.


Table 5 bind all and bind control Functions, Based on Interface State

Interface State

Result Using bind all or bind control Commands

Shut down

With or without active calls

TCP, TLS, and UDP socket listeners are initially closed. (Socket listeners receive datagrams addressed to the socket.)

Then the sockets are opened to listen to any IP address.

If the outgoing gateway has the bind command enabled and has an active call, the call becomes a one-way call with media flowing from the outgoing gateway to the terminating gateway.

The dial peer bind socket listeners of the interface are closed and the configuration turns inactive for all subsequent SIP messages.

Not shut down

Without active calls

TCP, TLS, and UDP socket listeners are initially closed. (Socket listeners receive datagrams addressed to the socket.)

Then the sockets are opened and bound to the IP address set by the bind command.

The sockets accept packets destined for the bound address only.

The dial peer bind socket listeners of the interface are reopened and the configuration turns active for all subsequent SIP messages.

Not shut down

With active calls

TCP, TLS, and UDP socket listeners are initially closed.

Then the sockets are opened to listen to any IP address.

The dial peer bind socket listeners of the interface are reopened and the configuration turns active for all subsequent SIP messages.

Bound interface’s IP address is removed.

TCP, TLS, and UDP socket listeners are initially closed.

Then the sockets are opened to listen to any address because the IP address has been removed.

A message is printed that states the IP address has been deleted from the bound SIP interface.

If the outgoing gateway has the bind command enabled and has an active call, the call becomes a one-way call with media flowing from the outgoing gateway to the terminating gateway.

The dial peer bind socket listeners of the interface are closed and the configuration turns inactive for all subsequent SIP messages.

The physical cable is pulled on the bound port, or the interface layer goes down.

TCP, TLS, and UDP socket listeners are initially closed.

Then the sockets are opened and bound to listen to any address.

When the pulled cable is replaced, the result is as documented for interfaces that are not shut down.

The dial peer bind socket listeners of the interface are closed and the configuration turns inactive for all subsequent SIP messages.

A bind interface is shut down, or its IP address is changed, or the physical cable is pulled while SIP calls are active.

The call becomes a one-way call with media flowing in only one direction. The media flows from the gateway where the change or shutdown took place to the gateway where no change occurred. Thus, the gateway with the status change no longer receives media.

The call is then disconnected, but the disconnected message is not understood by the gateway with the status change, and the call is still assumed to be active.

If the bind interface is shutdown, the dial peer bind socket listeners of the interface are closed. If the IP address of the interface is changed, the socket listeners representing the bind command is opened with the available IP address of the interface and the configuration turns active for all subsequent SIP messages.

Configuring SIP Binding

SUMMARY STEPS

    1.    enable

    2.    configure terminal

    3.    interface type number

    4.    ip address ip-address mask [secondary]

    5.    exit

    6.    Use one of the following commands to configure SIP binding:

    • bind {control | media | all} source-interface interface-id [ipv6-address ipv6-address] in SIP configuration mode.
    • voice-class sip bind {control | media} source interface interface-id [ipv6-address ipv6-address] in dial-peer configuration mode.

    7.    end


DETAILED STEPS
     Command or ActionPurpose
    Step 1 enable


    Example:
    Router> enable
     

    Enables privileged EXEC mode.

    • Enter your password if prompted.

     
    Step 2 configure terminal


    Example:
    Router# configure terminal
     

    Enters global configuration mode.

     
    Step 3 interface type number


    Example:
    Router(config)# interface fastethernet0/0
     

    Configures an interface type and enters the interface configuration mode.

    • type number —Type of interface to be configured and the port, connector, or interface card number.

     
    Step 4 ip address ip-address mask [secondary]


    Example:
    Router(config-if)# ip address 192.168.200.33 255.255.255.0 
     

    Configures a primary or secondary IP address for an interface.

     
    Step 5 exit


    Example:
    Router(config-if)# exit
     

    Exits the current mode.

     
    Step 6Use one of the following commands to configure SIP binding:
    • bind {control | media | all} source-interface interface-id [ipv6-address ipv6-address] in SIP configuration mode.
    • voice-class sip bind {control | media} source interface interface-id [ipv6-address ipv6-address] in dial-peer configuration mode.


    Example:SIP binding in SIP configuration mode:
    Device(config)# voice service voip
    Device(conf-voi-serv)# sip
    Device(conf-serv-sip)# bind control source-interface FastEthernet0/0
    Device(conf-serv-sip)# exit
    


    Example:SIP binding in dial-peer configuration mode:
    Device(config)# dial-peer voice 100 voip
    Device(config-dial-peer)# session protocol sipv2
    Device(config-dial-peer)# voice-class sip bind control source-interface fastethernet0/0 
    Device(config-dial-peer)# exit
     

    Sets a source interface for signaling and media packets. The binding applies to the specified interfaces only. SIP must be configured globally or at a dial peer level.

    • control —Binds signaling packets.

    • media —Binds media packets.

    • all —Binds signaling and media packets.

    • source interface interface-id —Type of interface and its ID.

    • ipv6-address ipv6-address —Configures the IPv6 address. Ensure that the IPv6 address has been applied to an interface.

     
    Step 7 end
     

    Exits to privileged EXEC mode.

     

    Verifying SIP Binding

    SUMMARY STEPS

      1.    show ip sockets

      2.    show sip-ua status

      3.    show sip-ua connections {tcp [tls] | udp} {brief | detail}

      4.    show dial-peer voice

      5.    show running-config


    DETAILED STEPS
      Step 1   show ip sockets

      Use this command to display IP socket information and indicate whether the bind address of the receiving gateway is set.

      The following sample output indicates that the bind address of the receiving gateway is set:



      Example:
      Device# show ip sockets
      
      Proto Remote Port Local Port In Out Stat TTY OutputIF 
      17 0.0.0.0 0--any-- 2517 0 0 9 0 
      17 --listen--	172.18.192.204 1698 0 0 1 0
      17 0.0.0.0 0 172.18.192.204 67 0 0 489 0
      17 0.0.0.0 0 172.18.192.204 5060 0 0 A1 0  			 
      


      Example:
      
       
      			 
      Step 2   show sip-ua status

      Use this command to display SIP user-agent status and indicate whether bind is enabled.

      The following sample output indicates that signaling is disabled and media on 172.18.192.204 is enabled:



      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): ENABLED 172.18.192.204
      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 ipv4 
      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
      
      Step 3   show sip-ua connections {tcp [tls] | udp} {brief | detail}

      Use this command to display the connection details for the UDP transport protocol. The command output looks identical for TCP and TLS.



      Example:
      Device# show sip-ua connections udp detail
       
      Total active connections      : 0
      No. of send failures          : 0
      No. of remote closures        : 0
      No. of conn. failures         : 0
      No. of inactive conn. ageouts : 10
      ---------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
       No Active Connections Found
      -------------- SIP Transport Layer Listen Sockets ---------------
        Conn-Id               Local-Address             
       ===========    ============================= 
         2            [9.42.28.29]:5060

      Step 4   show dial-peer voice

      Use this command, for each dial peer configured, to verify that the dial-peer configuration is correct. The following is sample output from this command for a VoIP dial peer:



      Example:
      Device# show dial-peer voice 101
      
      VoiceOverIpPeer1234
              peer type = voice, system default peer = FALSE, information type = voice,
              description = `',
              tag = 1234, destination-pattern = `',
              voice reg type = 0, corresponding tag = 0,
              allow watch = FALSE
              answer-address = `', preference=0,
              CLID Restriction = None
              CLID Network Number = `'
              CLID Second Number sent 
              CLID Override RDNIS = disabled,
              rtp-ssrc mux = system
              source carrier-id = `', target carrier-id = `',
              source trunk-group-label = `',  target trunk-group-label = `',
              numbering Type = `unknown'
              group = 1234, Admin state is up, Operation state is down,
              incoming called-number = `', connections/maximum = 0/unlimited,
              DTMF Relay = disabled,
              modem transport = system,
              URI classes:
                  Incoming (Request) = 
                  Incoming (Via) = 
                  Incoming (To) = 
                  Incoming (From) = 
                  Destination = 
              huntstop = disabled,
              in bound application associated: 'DEFAULT'
              out bound application associated: ''
              dnis-map = 
              permission :both
              incoming COR list:maximum capability
              outgoing COR list:minimum requirement
              outgoing LPCOR: 
              Translation profile (Incoming):
              Translation profile (Outgoing):
              incoming call blocking:
              translation-profile = `'
              disconnect-cause = `no-service'
              advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
              mailbox selection policy: none
              type = voip, session-target = `',
              technology prefix: 
              settle-call = disabled
              ip media DSCP = ef, ip media rsvp-pass DSCP = ef
              ip media rsvp-fail DSCP = ef, ip signaling DSCP = af31,
              ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41
              ip video rsvp-fail DSCP = af41,
              ip defending Priority = 0, ip preemption priority = 0
              ip policy locator voice: 
              ip policy locator video: 
              UDP checksum = disabled,
              session-protocol = sipv2, session-transport = system,
              req-qos = best-effort, acc-qos = best-effort,
              req-qos video = best-effort, acc-qos video = best-effort,
              req-qos audio def bandwidth = 64, req-qos audio max bandwidth = 0,
              req-qos video def bandwidth = 384, req-qos video max bandwidth = 0, 
              RTP dynamic payload type values: NTE = 101
              Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122
                     CAS=123, TTY=119, ClearChan=125, PCM switch over u-law=0,
                     A-law=8, GSMAMR-NB=117 iLBC=116, AAC-ld=114, iSAC=124
                     lmr_tone=0, nte_tone=0
                     h263+=118, h264=119
                     G726r16 using static payload
                     G726r24 using static payload
              RTP comfort noise payload type = 19
              fax rate = voice,   payload size =  20 bytes
              fax protocol = system
              fax-relay ecm enable
              Fax Relay ans enabled
              Fax Relay SG3-to-G3 Enabled (by system configuration)
              fax NSF = 0xAD0051 (default)
              codec = g729r8,   payload size =  20 bytes,
              video codec = None
              voice class codec = `'
              voice class sip session refresh system
              voice class sip rsvp-fail-policy voice post-alert mandatory keep-alive interval 30
              voice class sip rsvp-fail-policy voice post-alert optional keep-alive interval 30
              voice class sip rsvp-fail-policy video post-alert mandatory keep-alive interval 30
              voice class sip rsvp-fail-policy video post-alert optional keep-alive interval 30
              text relay = disabled
              Media Setting = forking (disabled) flow-through (global)
              Expect factor = 10, Icpif = 20,
              Playout Mode is set to adaptive,
              Initial 60 ms, Max 1000 ms
              Playout-delay Minimum mode is set to default, value 40 ms 
              Fax nominal 300 ms
              Max Redirects = 1, signaling-type = cas,
              VAD = enabled, Poor QOV Trap = disabled, 
              Source Interface = NONE
              voice class sip url = system,
              voice class sip tel-config url = system,
              voice class sip rel1xx = system,
              voice class sip anat = system,
              voice class sip outbound-proxy = "system",
              voice class sip associate registered-number =
                               system,
              voice class sip asserted-id system,
              voice class sip privacy system
              voice class sip e911 = system,
              voice class sip history-info = system,
              voice class sip reset timer expires 183 = system,
              voice class sip pass-thru headers = system,
              voice class sip pass-thru content unsupp = system,
              voice class sip pass-thru content sdp = system,
              voice class sip copy-list = system,
              voice class sip g729 annexb-all = system,
              voice class sip early-offer forced = system,
              voice class sip negotiate cisco = system,
              voice class sip block 180 = system,
              voice class sip block 183 = system,
              voice class sip block 181 = system,
              voice class sip preloaded-route = system,
              voice class sip random-contact = system,
              voice class sip random-request-uri validate = system,
              voice class sip call-route p-called-party-id = system,
              voice class sip call-route history-info = system,
              voice class sip privacy-policy send-always = system,
              voice class sip privacy-policy passthru = system,
              voice class sip privacy-policy strip history-info = system,
              voice class sip privacy-policy strip diversion = system,
              voice class sip map resp-code 181 = system,
              voice class sip bind control = enabled, 9.42.28.29,
              voice class sip bind media = enabled, 9.42.28.29,
              voice class sip bandwidth audio = system,
              voice class sip bandwidth video = system,
              voice class sip encap clear-channel = system,
              voice class sip error-code-override options-keepalive failure = system,
              voice class sip calltype-video = false
              voice class sip registration passthrough = System
              voice class sip authenticate redirecting-number  = system,
              redirect ip2ip = disabled
              local peer = false
              probe disabled,
              Secure RTP: system (use the global setting)
              voice class perm tag = `'
              Time elapsed since last clearing of voice call statistics never
              Connect Time = 0, Charged Units = 0,
              Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
              Accepted Calls = 0, Refused Calls = 0,
              Last Disconnect Cause is "",
              Last Disconnect Text is "",
              Last Setup Time = 0.
              Last Disconnect Time = 0.
      Note   

      If the bind address is not configured at the dial-peer, the output of the show dial-peer voice command remains the same except for the values of the voice class sip bind control and voice class sip bind media, which display “system,” indicating that the bind is configured at the global level.

      Step 5   show running-config

      Although the bind all command is an accepted configuration, it does not appear in show running-config command output. Because the bind all command is equivalent to issuing the commands bind source and bind media, those are the commands that appear in the show running-config command output.



      Example:The following sample output shows that bind is enabled on router 172.18.192.204:
      Building configuration...
      Current configuration : 2791 bytes
      !
      version 12.2
      service config
      no service single-slot-reload-enable
      no service pad
      service timestamps debug uptime
      service timestamps log uptime
      no service password-encryption
      service internal
      service udp-small-servers
      !
      ip subnet-zero
      ip ftp source-interface Ethernet0
      !
      voice service voip
       sip
        bind control source-interface FastEthernet0
      !
      interface FastEthernet0
       ip address 172.18.192.204 255.255.255.0
       duplex auto
       speed auto
       fair-queue 64 256 1000
       ip rsvp bandwidth 75000 100
      !
      voice-port 1/1/1
      no supervisory disconnect lcfo
      !
      dial-peer voice 1 pots
      application session
      destination-pattern 5550111
      port 1/1/1
      !
      dial-peer voice 29 voip
      application session
      destination-pattern 5550133
      session protocol sipv2
      session target ipv4:172.18.200.33
      codec g711ulaw
      !
      gateway
      !
      line con 0
      line aux 0
      line vty 0 4
      login
      !
      end