SIP Configuration Guide, Cisco IOS Release 15M&T - Configuring SIP Bind Features [Support]

Finding Feature Information

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Prerequisites for SIP Bind Features

The following are the prerequisites for this feature:

Ensure the gateway has voice functionality that is configurable for Session Initiation Protocol (SIP).
Establish a working IP network. For more information about configuring IP, refer to the Cisco IOS IP Addressing Configuration Guide.
Configure VoIP. For more information about configuring VoIP, refer to the Cisco IOS Voice Command Reference.

Restrictions for SIP Bind Features

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.

Information About SIP Bind Features

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 Bind Features
Source Address
Voice Media Stream Processing

Benefits of SIP Bind Features

The benefits of SIP Bind feature is as follows:

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. Once 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.

To configure SIP Gateway Support for the bind Command, you should understand the following concepts:

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 nondeterministic 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 allows 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 1

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 2

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 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 3

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
With active calls	bind media	Succeeds and overrides any existing bind mediacommand.

The bind all and bind controlcommands perform different functions based on the state of the interface. The table below describes the actions performed based on the interface state.

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 4

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.

How to Configure SIP Bind Features

Setting the Bind Command at the Global Level
Setting the Bind Command at the Dial-peer Level
Monitoring the Bind Command

Setting the Bind Command at the Global Level

To configure the bind command to an interface at the global level, perform the following steps.

Note

The bind media command applies to specific interfaces.

SUMMARY STEPS

1. enable

2. configure terminal

3. interface type / number

4. ip address ip-address mask [secondary]

5. exit

6. voice service voip

7. sip

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

9. exit

DETAILED STEPS

	Command or Action	Purpose
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. ip-address mask --IP address and mask for the associated IP subnet. secondary --Makes the configured address a secondary IP address. If this keyword is omitted, the configured address is the primary IP address.
Step 5	exit Example: Router(config-if)# exit	Exits the current mode.
Step 6	voice service voip Example: Router(config)# voice service voip	Enters voice service configuration mode.
Step 7	sip Example: Router(conf-voi-serv)# sip	Enters SIP configuration mode.
Step 8	bind {control \| media \| all} source-interface interface-id[ipv6-address ipv6-address] Example: Router(conf-serv-sip)# bind control source-interface FastEthernet0/0	Sets a source interface for signaling and media packets. 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.
Step 9	exit Example: Router(conf-serv-sip)# exit	Exits the current mode.

Setting the Bind Command at the Dial-peer Level

To configure the bind command on SIP for a VoIP dial-peer, perform the following steps.

SUMMARY STEPS

1. enable

2. configure terminal

3. interface type / number

4. ip address ip-address mask [secondary]

5. exit

6. dial-peer voice tag voip

7. session protocol sipv2

8. voice-class sip bind {control | media} source interface interface-id[ipv6-address ipv6-address]

9. exit

DETAILED STEPS

Command or Action

Purpose

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.

Note

You can only bind Loopback, Ethernet, FastEthernet, GigabitEthernet and Serial interfaces for dial peer.

Step 4

ip address ip-address mask [secondary]

Example:

Router(config-if)# ip address 2001:0DB8:0:1::1

Configures a primary or secondary IP address for an interface.

ip-address mask --IP address and mask for the associated IP subnet.
secondary --Makes the configured address a secondary IP address. If this keyword is omitted, the configured address is the primary IP address.

Step 5

exit

Example:

Router(config-if)# exit

Exits the current mode.

Step 6

dial-peer voice tag voip

Example:

Router(config)# dial-peer voice 100 voip

Enters dial peer voice configuration mode for the specified VoIP dial peer.

Step 7

session protocol sipv2

Example:

Router(config-dial-peer)# session protocol sipv2

Specifies use of IETF SIP.

Step 8

voice-class sip bind {control | media} source interface interface-id[ipv6-address ipv6-address]

Example:

Router(config-dial-peer)# voice-class sip bind control source-interface fastethernet0/0 ipv6-address 2001:0DB8:0:1::1

Sets a source interface for signaling and media packets.

control --Binds signaling packets.
media --Binds media packets.
source interface interface-id --Type of interface and its ID.
ipv6-address ipv6-address --(Optional) Configures the IPv6 address to the source interface.

Step 9

exit

Example:

Router(config-dial-peer)# exit

Exits the current mode.

Troubleshooting Tips

Troubleshooting Tips

For troubleshooting tips and a list of important debug commands, see "Verifying and Troubleshooting SIP Features".

Monitoring the Bind Command

To monitor the bind command, perform the following steps.

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

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:

Router# 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:

Router# 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:

Router# 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:

Router# 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.

Troubleshooting Tips

Troubleshooting Tips

For troubleshooting tips and a list of important debug commands, see "Verifying and Troubleshooting SIP Features".

Configuration Examples for SIP Bind Features

Example Verifying the bind Command

Example Verifying the bind Command

This sample output shows that bind is enabled on router 172.18.192.204:

Router# show running-config
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

Additional References

Standards

Standard	Title
No new or modified standards are supported by this feature, and support for existing standards has not been modified by this feature.	--

MIBs

MIB

MIBs Link

CISCO-SIP-UA-MIB

To locate and download MIBs for selected platforms, Cisco software releases, and feature sets, use Cisco MIB Locator found at the following URL:

http://www.cisco.com/go/mibs

RFCs

RFC	Title
RFC 2543	SIP: Session Initiation Protocol
RFC 2806	URLs for Telephone Calls

Technical Assistance

Description

Link

The Cisco Support website provides extensive online resources, including documentation and tools for troubleshooting and resolving technical issues with Cisco products and technologies.

To receive security and technical information about your products, you can subscribe to various services, such as the Product Alert Tool (accessed from Field Notices), the Cisco Technical Services Newsletter, and Really Simple Syndication (RSS) Feeds.

Access to most tools on the Cisco Support website requires a Cisco.com user ID and password.

http://www.cisco.com/cisco/web/support/index.html

Feature Information for SIP Bind Features

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 5

Feature Information for SIP Bind Features

Feature Name

Releases

Feature Information

SIP Gateway Support for the bind Command

12.2(2)XB 12.2(2)XB2 12.2(8)T 12.2(11)T 12.3(4)T Cisco IOS XE Release 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 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.

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.

Configuring SIP Bind Features

Finding Feature Information

Prerequisites for SIP Bind Features

Restrictions for SIP Bind Features

Information About SIP Bind Features

Benefits of SIP Bind Features

Source Address

Voice Media Stream Processing

How to Configure SIP Bind Features

Setting the Bind Command at the Global Level

Setting the Bind Command at the Dial-peer Level

Troubleshooting Tips

Monitoring the Bind Command

Troubleshooting Tips

Configuration Examples for SIP Bind Features

Example Verifying the bind Command

Additional References

Related Documents

Standards

MIBs

RFCs

Technical Assistance

Feature Information for SIP Bind Features