13-Voice Command Reference

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04-SIP commands
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SIP commands

The following compatibility matrix shows the support of hardware platforms for SIP:

 

Hardware

SIP compatibility

MSR810, MSR810-W, MSR810-W-DB, MSR810-LM, MSR810-W-LM, MSR810-10-PoE, MSR810-LM-HK, MSR810-W-LM-HK, MSR810-LMS-EA

MSR810, MSR810-W, MSR810-W-DB, MSR810-LM, MSR810-W-LM, MSR810-10-PoE, MSR810-LM-HK, MSR810-W-LM-HK: Yes

MSR810-LMS-EA: No

MSR810-LMS, MSR810-LUS

No

MSR2600-6-X1

No

MSR2600-10-X1

Yes

MSR 2630

Yes

MSR3600-28, MSR3600-51

Yes

MSR3600-28-SI, MSR3600-51-SI

No

MSR3600-28-X1, MSR3600-28-X1-DP, MSR3600-51-X1, MSR3600-51-X1-DP

No

MSR3610-I-DP, MSR3610-IE-DP

No

MSR3610-X1, MSR3610-X1-DP, MSR3610-X1-DC, MSR3610-X1-DP-DC

Yes

MSR 3610, MSR 3620, MSR 3620-DP, MSR 3640, MSR 3660

Yes

MSR3610-G, MSR3620-G

No

To support SIP, some device models require the Voice Software License. For more information, see license management in Fundamentals Configuration Guide.

address sip

Use address sip to configure a call destination IP address for a VoIP entity.

Use undo address sip to restore the default.

Syntax

address sip { dns domain-name port port-number | ip ip-address [ port port-number ] | proxy }

undo address sip { dns | ip | proxy }

Default

No call destination IP address exists.

Views

VoIP entity view

Predefined user roles

network-admin

Parameters

dns domain-name: Specifies a destination domain name, which consists of case-insensitive strings separated by a dot (for example, aabbcc.com). Each separated string contains no more than 63 characters. A domain name can include letters, digits, hyphens (-), and underscores (_), and has a maximum length of 253 characters.

port port-number: Specifies a destination port by its port number in the range of 1 to 65535. If the ip keyword is specified, the default port number is 5060 for TCP and UDP and 5061 for TLS. If the dns keyword is specified, the destination port number must be configured.

ip ip-address: Specifies a destination IP address.

proxy: Contacts the SIP proxy server to obtain the destination IP address.

Examples

# Configure the call destination IP address as 3.3.3.3 for VoIP entity 10.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 voip

[Sysname-voice-dial-entity10] address sip ip 3.3.3.3

asserted-id

Use asserted-id to add the P-Asserted-Identity or P-Preferred-Identity header field to outgoing SIP messages.

Use undo asserted-id to restore the default.

Syntax

asserted-id { pai | ppi }

undo asserted-id

Default

Outgoing SIP messages do not carry the P-Asserted-Identity or P-Preferred-Identity header field in outgoing SIP messages.

Views

SIP view

Predefined user roles

network-admin

Parameters

pai: Adds the P-Asserted-Identity header field to outgoing SIP messages.

ppi: Adds the P-Preferred-Identity header field to outgoing SIP messages.

Examples

# Add the P-Asserted-Identity header field to outgoing SIP messages.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] asserted-id pai

bind

Use bind to configure global source interface binding for outbound SIP messages or media packets.

Use undo bind to restore the default.

Syntax

bind { control | media } source-interface interface-type interface-number

undo bind { control | media }

Default

The egress interface is used as the source interface of outbound SIP messages or media packets.

Views

SIP view

Predefined user roles

network-admin

Parameters

control: Specifies outbound SIP messages.

media: Specifies outbound media packets.

source-interface interface-type interface-number: Uses the specified interface as the source interface of outbound SIP messages or media packets. The specified interface must be a Layer 3 Ethernet interface, VLAN interface, loopback interface, or dialer interface.

Usage guidelines

The following table describes how source interface binding works in different conditions:

 

Condition

Result

Configure a new source interface when ongoing calls exist.

·     The new source interface takes effect for new SIP media sessions but does not take effect for existing SIP media sessions.

·     The new source interface immediately takes effect for all SIP signaling sessions.

The bound source interface is shut down.

The source interface binding does not take effect, and the default setting is restored.

The IP address of the bound source interface or the bound source interface is removed.

The source interface binding does not take effect, and the default setting is restored.

The physical or link layer state of the bound interface is down.

The source interface binding does not take effect, and the default setting is restored.

The bound source interface obtains a new IP address from the DHCP or PPPoE server.

The new IP address is used as the source IP address.

Configure a new source interface during SIP registration.

The new source interface takes effect for new registrations.

 

You can configure source interface binding both globally (by using the bind command in SIP view) and for a specific VoIP entity (by using the voice-class sip bind command in VoIP entity view). The configuration in VoIP entity view takes precedence over the global configuration. A VoIP entity uses the global configuration only when source interface binding is not configured in VoIP entity view.

If the specified source interface does not have an IP address or its IP address is invalid, the device uses the default configuration.

Examples

# Specify Dialer 0 as the source interface for outbound SIP messages.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] bind control source-interface dialer 0

Related commands

voice-class sip bind

crypto

Use crypto to configure an SSL client or server policy to be used by TLS.

Use undo crypto to delete the SSL client or server policy used by TLS.

Syntax

crypto { ssl-client-policy client-policy-name | ssl-server-policy server-policy-name }

undo crypto { server-policy | client-policy }

Default

No SSL client or server policy is configured for TLS.

Views

SIP view

Predefined user roles

network-admin

Parameters

ssl-client-policy client-policy-name: Specifies an SSL client policy by its name, a case-insensitive string of 1 to 31 characters.

ssl-server-policy server-policy-name: Specifies an SSL server policy by its name, a case-insensitive string of 1 to 31 characters.

Usage guidelines

To enable the device to receive TLS requests, specify the SSL client and server policies to be used by TLS and then enable the TLS listening port by using the transport command.

You must disable the TLS listening port before you can use a new SSL server policy or modify the configuration of the existing SSL server policy.

If you use a new SSL client policy or modify the configuration of the existing SSL client policy, the new policy applies only to new TLS connections. Existing TLS connections still use the original SSL client policy.

Examples

# Configure TLS to use the SSL server policy named Server1 and the SSL client policy named Server2.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] crypto ssl-server-policy Server1

[Sysname-voice-sip] crypto ssl-client-policy Server2

Related commands

transport

display voice ip address trusted list

Use display voice ip address trusted list to display the trusted node list.

Syntax

display voice ip address trusted list

Views

Any view

Predefined user roles

network-admin

network-operator

Usage guidelines

This command displays trusted nodes configured by using the ip command and call destination IP addresses configured by using the address sip command.

Examples

# Display the trusted node list.

<Sysname> display voice ip address trusted list

IP address trusted authentication: Enabled

 

VoIP entity IP addresses:

Entity tag      State    SIP IP address

----------      -----    --------------

20              Up       192.168.4.110

53232           Down     192.168.4.210

55555           Up       192.168.4.210

9613            Up       192.168.4.125

 

IP address trusted list:

 192.168.4.0 255.255.255.0

 192.168.5.120 255.255.255.255

Table 1 Command output

Field

Description

IP address trusted authentication

Whether IP address trusted authentication is enabled:

·     Enabled.

·     Disabled.

VoIP entity IP addresses

Trusted IP addresses for VoIP entities.

Entity tag

Tag of a VoIP entity.

State

Status of a VoIP entity:

·     Up.

·     Down.

SIP IP address

Call destination IP address of a VoIP entity.

IP address trusted list

List of trusted nodes.

 

Related commands

address sip

ip

ip address trusted authenticate

display voice sip call

Use display voice sip call to display information about SIP calls.

Syntax

display voice sip call

Views

Any view

Predefined user roles

network-admin

network-operator

Examples

# Display information about SIP calls.

<Sysname> display voice sip call

SIP UAC Call Information

                                                                               

Call 1

   Call ID: 2856599de8c8824524de623ac7b1755e@200.1.1.36

   Call status: Connected

   Calling number: 77201

   Called number: 30

   Control block ID: 8

   Local IP address: 200.1.1.36: 5060

   Remote IP address: 200.1.1.30: 5060

Media stream

   Media status: Send and receive

   Negotiated codec: g729r8

   Codec payload type: 18

   Codec payload size: 30

   Codec transparent: Disabled

   Media mode: Flow-through

   Negotiated DTMF-relay: Inband-voice

   Local IP address: 200.1.1.36: 16316

   Remote IP address: 200.1.1.30: 16642

                                                                               

Number of SIP UAC calls: 1

Table 2 Command output

Field

Description

Call status

Call status:

·     Originating.

·     Answering.

·     Connected.

·     Releasing.

Local IP address

Source IP address and port number for SIP messages.

Remote IP address

Destination IP address and port number for SIP messages.

Negotiated codec

Negotiated codec:

·     g711alaw.

·     g711ulaw.

·     g723r53.

·     g723r63.

·     g726r16.

·     g726r24.

·     g726r32.

·     g726r40.

·     g729a.

·     g729br8.

·     g729r8.

This field displays N/A if codec negotiation fails or no codec is used.

Media status

Media status:

·     Send and receive.

·     Send only.

·     Receive only.

·     Inactive.

·     None.

Media mode

Whether media flow around is enabled:

·     Flow-around—Enabled.

·     Flow-through—Disabled.

Negotiated DTMF-relay

Negotiated type of DTMF signaling:

·     Inband-voice—Inband DTMF signaling.

·     Outband-SIP—SIP mode for out-of-band DTMF signaling.

·     Outband-NTE—NTE mode for out-of-band DTMF signaling.

Number of SIP UAC calls

Number of SIP calls initiated by the device that acts as a UAC.

Number of SIP UAS calls

Number of SIP calls initiated by the device that acts as a UAS.

 

display voice sip connection

Use display voice sip connection to display information about SIP connections.

Syntax

display voice sip connection { tcp | tls }

Views

Any view

Predefined user roles

network-admin

network-operator

Parameters

tcp: Specifies TCP connections.

tls: Specifies TLS connections.

Usage guidelines

This command displays information about established SIP connections and SIP connections that are being established.

Examples

# Display information about TCP connections.

<Sysname> display voice sip connection tcp

Conn-Id  Local-IP         Local-Port  Remote-IP        Remote-Port      Conn-State

----------------------------------------------------------------------------------

 569      100.1.1.84       1593       100.1.1.100       5060            Established

 570      100.1.1.84       1594       100.1.1.101       5060            Established

 571      100.1.1.84       1595       100.1.1.81        5060            Established

 572      192.168.0.82     1596       192.168.0.81      5060            Established

# Display information about TLS connections.

<Sysname> display voice sip connection tls

Conn-Id  Local-IP         Local-Port  Remote-IP        Remote-Port      Conn-State

----------------------------------------------------------------------------------

 73       192.168.0.202    1086       192.168.0.132     5061            Established

Table 3 Command output

Field

Description

Conn-Id

Connection ID.

Conn-State

Connection state:

·     Connecting.

·     Established.

 

Related commands

reset voice sip connection

display voice sip map

Use display voice sip map to display mappings between PSTN causes and SIP status.

Syntax

display voice sip map { pstn-sip | sip-pstn }

Views

Any view

Predefined user roles

network-admin

network-operator

Parameters

pstn-sip: Displays PSTN cause-to-SIP status mappings.

sip-pstn: Displays SIP status-to-PSTN cause mappings.

Examples

# Display PSTN cause-to-SIP status mappings.

<Sysname> display voice sip map pstn-sip

 The PSTN Cause to SIP Status code mapping table:

 Index       PSTN-Cause     SIP-Status     Default

---------------------------------------------------

  1              1            400*           404

  2              2            400*           404

  3              3            404            404

  4             16            ---            ---

  5             17            486            486

  6             18            408            408

  7             19            480            480

  8             20            480            480

  9             21            403            403

 10             22            410            410

 11             23            410            410

 12             25            500            500

 13             26            404            404

 14             27            502            502

 15             28            484            484

 16             29            501            501

 17             31            480            480

 18             34            503            503

 19             38            503            503

 20             41            503            503

 21             42            503            503

 22             47            503            503

 23             55            403            403

 24             57            403            403

 25             58            503            503

 26             63            500            500

 27             65            488            488

 28             70            488            488

 29             79            501            501

 30             87            403            403

 31             88            503            503

 32            102            504            504

 33            111            500            500

 34            127            500            500

Table 4 Command output

Field

Description

PSTN-Cause

PSTN cause code.

SIP-Status

SIP status code.

If the configured SIP status code is different from the default, it is highlighted with an asterisk.

Default

Default SIP status code.

 

# Display SIP status-to-PSTN cause mappings.

<Sysname> display voice sip map sip-pstn

 The SIP Status code to PSTN Cause mapping table:

 Index       SIP-Status     PSTN-Cause     Default

---------------------------------------------------

  1            400             41             41

  2            401             21             21

  3            402             21             21

  4            403             21             21

  5            404              1              1

  6            405             63             63

  7            406             79             79

  8            407             21             21

  9            408            102            102

 10            410             22             22

 11            413            127            127

 12            414            127            127

 13            415             79             79

 14            416            127            127

 15            420            127            127

 16            421            127            127

 17            423            127            127

 18            480             18             18

 19            481             41             41

 20            482             25             25

 21            483             25             25

 22            484             28             28

 23            485              1              1

 24            486             17             17

 25            487            127            127

 26            488            127            127

 27            500             41             41

 28            501             79             79

 29            502             38             38

 30            503             41             41

 31            504            102            102

 32            505            127            127

 33            513            127            127

 34            600             17             17

 35            603             21             21

 36            604              1              1

 37            606             58             58

Table 5 Command output

Field

Description

SIP-Status

SIP status code.

PSTN-Cause

PSTN cause code.

If the configured PSTN cause code is different from the default, it is highlighted with an asterisk.

Default

Default PSTN cause code.

 

display voice sip register-status

Use display voice sip register-status to display SIP UA registration status information.

Syntax

display voice sip register-status

Views

Any view

Predefined user roles

network-admin

network-operator

Examples

# Display SIP UA registration status information.

<Sysname> display voice sip register-status

Number                          Entity     Registrar Server      Expires Status

--------------------------------------------------------------------------------

98                              98         192.168.4.240:5060    N/A     Offline

1000                            0          192.168.4.240:5060    2877    Online

Table 6 Command output

Field

Description

Number

Phone number.

Entity

Entity number. This field displays 0 if a SIP trunk account is configured by using the credentials command.

Registrar Server

Address of the registrar.

Expires

Aging time for the phone number, in seconds. This field displays N/A if the phone number is registering or fails to register.

Status

State of the phone number:

·     Offline.

·     Online.

·     Login.

·     Logout.

·     DNS-in—DNS query is being performed before the number is registered.

·     DNS-out—DNS query is being performed before the number is deregistered.

 

early-media enable

Use early-media enable to enable early media negotiation.

Use undo early-media enable to disable early media negotiation.

Syntax

early-media enable

undo early-media enable

Default

Early media negotiation is enabled.

Views

SIP view

Predefined user roles

network-admin

Usage guidelines

If this feature is enabled, the terminating device sends a 183 session progress response with media information to the originating device after receiving the request to establish a call. If this feature is disabled, the device sends a 180 ringing response without media information to the originating device.

Examples

# Enable early media negotiation.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] early-media enable

ip

Use ip to specify a trusted node.

Use undo ip to delete trusted nodes.

Syntax

ip ipv4-address [ mask ]

undo ip ipv4-address [ mask ]

Default

No trusted nodes exist.

Views

Trusted node list view

Predefined user roles

network-admin

Parameters

ipv4-address: Specifies a trusted node by its IPv4 address.

mask: Specifies the mask of the IPv4 address. If you do not specify a mask, the 32-bit mask is used.

Examples

# Specify the devices on subnet 1.1.1.0/24 as trusted nodes.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip-server

[Sysname-voice-server] trusted-point ipv4 100.1.1.125

ip address trusted authenticate

Use ip address trusted authenticate to enable IP address trusted authentication.

Use undo ip address trusted authenticate to disable IP address trusted authentication.

Syntax

ip address trusted authenticate

undo ip address trusted authenticate

Default

IP address trusted authentication is disabled. All nodes are regarded as trusted, and the device accepts calls from any nodes.

View

SIP view

Predefined user roles

network-admin

Usage guidelines

After you enable this feature, the device accepts calls only from trusted nodes.

For calls to be successfully established, configure the proxy server, registrars, the DNS server, and the MWI server as trusted nodes.

Examples

# Enable IP address trusted authentication.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] ip address trusted authenticate

ip address trusted list

Use ip address trusted list to enter trusted node list view.

Use undo ip address trusted list to restore the default.

Syntax

ip address trusted list

undo ip address trusted list

Default

No trusted node list exists.

Views

SIP view

Predefined user roles

network-admin

Examples

# Enter trusted node list view

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] ip address trusted list

[Sysname-voice-sip-iptrust-list]

ip qos dscp

Use ip qos dscp to set the global DSCP value for IP packets carrying media streams or signaling.

Use undo ip qos dscp to restore the default.

Syntax

ip qos dscp { dscp-value | dscp-value-set } { media | signaling }

undo ip qos dscp { dscp-value | dscp-value-set } { media | signaling }

Default

The DSCP value for IP packets is ef (101110).

Views

SIP view

Predefined user roles

network-admin

Parameters

dscp-value: Specifies a DSCP value in the range of 0 to 63.

dscp-value-set: Specifies a DSCP value, which can be the keyword af11, af12, af13, af21, af22, af23, af31, af32, af33, af41, af42, af43, cs1, cs2, cs3, cs4, cs5, cs6, cs7, or ef.

media: Applies the DSCP value to IP packets carrying media streams.

signaling: Applies the DSCP value to IP packets carrying signaling.

Table 7 DSCP values

Keyword

DSCP value in binary

DSCP value in decimal

af11

001010

10

af12

001100

12

af13

001110

14

af21

010010

18

af22

010100

20

af23

010110

22

af31

011010

26

af32

011100

28

af33

011110

30

af41

100010

34

af42

100100

36

af43

100110

38

cs1

001000

8

cs2

010000

16

cs3

011000

24

cs4

100000

32

cs5

101000

40

cs6

110000

48

cs7

111000

56

ef

101110

46

 

Usage guidelines

You can configure the ip qos dscp command both globally (in SIP view) and for a specific POTS/VoIP entity (in POTS/VoIP entity view). The configuration in POTS/VoIP entity view takes precedence over the global configuration. A POTS/VoIP entity uses the global configuration only when the ip qos dscp command is not configured in POTS/VoIP entity view.

Examples

# Set the global DSCP value to af41 for IP packets carrying signaling.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] ip qos dscp af41 signaling

Related commands

ip qos dscp (VoIP/POTS entity view)

min-se

Use min-se to set the maximum and minimum session expiration timers.

Use undo min-se to restore the default.

Syntax

min-se time [ session-expires interval ]

undo min-se

Default

Both the maximum and minimum session expiration timers are 1800 seconds.

Views

SIP view

Predefined user roles

network-admin

Parameters

time: Specifies the minimum session expiration time in the range of 90 to 65535 seconds. The minimum session expiration time cannot be greater than the maximum session expiration time.

session-expires interval: Specifies the maximum session expiration time in the range of 90 to 65535 seconds. The default maximum session expiration time equals the minimum session expiration time.

Usage guidelines

You can use this command to set the values of the Min-SE and Session-Expires header fields on a UAC. If you execute this command on a UAS, only the Min-SE header field is set because the UAS copies the Session-Expires header field from the request to the response.

Examples

# Set the minimum session expiration time to 1000 seconds and the maximum session expiration time to 2000 seconds.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] min-se 1000 session-expires 2000

Related commands

session refresh

mki

Use mki to enable support for the Master Key Identifier (MKI) field and set the length of the MKI field.

Use undo mki to restore the default.

Syntax

mki mki-length

undo mki

Default

The MKI field is not supported.

Views

SIP view

Predefined user roles

network-admin

Parameters

mki-length: Specifies the length of the MKI field, in the range of 1 to 128 bytes.

Usage guidelines

This command enables the device to add the MKI field to outgoing SRTP and SRTCP packets and to identify the MKI field in incoming SRTP and SRTCP packets.

This command takes effect only when SRTP is the media stream protocol for SIP calls. To specify SRTP as the medial stream protocol for SIP calls, use the srtp command.

For SIP calls to be established, configure this command on both the originating and terminating devices.

Examples

# Enable support for the MKI field and set the length of the MKI field to 1 byte.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] srtp

[Sysname-voice-sip] mki 1

Related commands

srtp

options-ping

Use options-ping to globally enable in-dialog keepalive.

Use undo options-ping to globally disable in-dialog keepalive.

Syntax

options-ping seconds

undo options-ping

Default

In-dialog keepalive is disabled globally.

View

SIP view

Predefined use roles

network-admin

Parameters

seconds: Specifies the global interval for sending OPTIONS messages during a session, in the range of 60 to 1200 seconds.

Usage guidelines

This command enables the device to periodically send OPTIONS messages at the specified interval to monitor the status of the remote SIP UA during a session. It does not take effect when the session refresh negotiation succeeds before a call is established.

If you disable this feature, the device does not send OPTIONS messages after a call is established.

For a VoIP entity, the entity-specific in-dialog keepalive interval takes priority over the global in-dialog keepalive interval set in SIP view.

Example

# Globally enable in-dialog keepalive and set the interval to 60 seconds for sending OPTIONS messages during a session.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] options-ping 60

Related commands

voice-class sip options-ping

outband sip

Use outband sip to enable out-of-band DTMF signaling.

Use undo outband sip to restore the default.

Syntax

outband sip

undo outband

Default

Inband DTMF signaling is enabled.

Views

POTS entity view

VoIP entity view

Predefined user roles

network-admin

Usage guidelines

If you use out-of-band DTMF signaling, configure the outband sip command on both the calling and called devices.

Examples

# Enable out-of-band DTMF signaling for VoIP entity 10.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 pots

[Sysname-voice-dial-entity10] outband sip

privacy

Use privacy to add the Privacy header field to outgoing INVITE requests.

Use undo privacy to restore the default.

Syntax

privacy

undo privacy

Default

Outgoing INVITE requests do not include the Privacy header field.

Views

SIP view

Predefined user roles

network-admin

Examples

# Add the Privacy header field to outgoing INVITE requests.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] privacy

proxy

Use proxy to specify a proxy server.

Use undo proxy to restore the default.

Syntax

proxy { dns domain-name port port-number | ip ip-address [ port port-number ] }

undo proxy { dns | ip }

Default

No proxy servers are specified.

Views

SIP view

Predefined user roles

network-admin

Parameters

dns domain-name: Specifies the domain name of the proxy server, which consists of case-insensitive strings separated by a dot (for example, aabbcc.com). Each separated string contains no more than 63 characters. A domain name can include letters, digits, hyphens (-), and underscores (_), and has a maximum length of 253 characters.

ip ip-address: Specifies the IPv4 address of the proxy server.

port port-number: Specifies the port number of the proxy server, in the range of 1 to 65535. If the ip keyword is specified, the default port number is 5060. If the dns keyword is specified, the destination port number must be configured.

Examples

# Specify the proxy server with IP address 169.54.5.10 and port number 1120.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] proxy ipv4 169.54.5.10 port 1120

# Specify the proxy server with the domain name abc.com and port number 1100.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] proxy dns abc.com port 1100

register-number

Use register-number to configure a POTS entity to register the phone number with the registrar.

Use undo register-number to configure a POTS entity to deregister the phone number with the registrar.

Syntax

register-number

undo register-number

Default

After you complete the SIP registration configuration, a POTS entity registers the phone number with the registrar.

Views

POTS entity view

Predefined user roles

network-admin

Usage guidelines

A registrar cannot store multiple entries for one phone number. If multiple POTS entities on a device have the same phone number, only one POTS entity can register that number with the registrar.

Examples

# Configure POTS entity 10 to deregister the phone number with the registrar.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 pots

[Sysname-voice-dial-entity10] match-template 1000

[Sysname-voice-dial-entity10] line 2/1/1

[Sysname-voice-dial-entity10] undo register-number

Related commands

match-template

registrar

Use registrar to specify a registrar.

Use undo registrar to remove the configuration of a registrar and to notify the registrar to deregister the SIP UA.

Syntax

registrar registrar-index { dns domain-name port port-number | ip ip-address [ port port-number ] } [ expires seconds ] [ refresh-ratio ratio-percentage ] [ scheme { sip | sips } ] [ tcp [ tls ] ]

undo registrar registrar-index

Default

No registrars are specified.

Views

SIP view

Predefined user roles

network-admin

Parameters

registrar-index: Specifies the index for a registrar, in the range of 1 to 6.

dns domain-name: Specifies a registrar by its domain name, which consists of case-insensitive strings separated by dots (for example, aabbcc.com). Each separated string contains no more than 63 characters. A domain name can include letters, digits, hyphens (-), and underscores (_), and has a maximum length of 253 characters.

ip ip-address: Specifies a registrar by its IP address.

port port-number: Specifies the port number of a registrar, in the range of 1 to 65535. If the ip keyword is specified, the default port number is 5060 for TCP and UDP and 5061 for TLS. If the dns keyword is specified, the port number must be configured.

expires seconds: Specifies the registration expiration time in the range of 60 to 65535 seconds. The default registration expiration time is 3600 seconds.

refresh-ratio ratio-percentage: Specifies the refresh percentage in the range of 50 to 100. The default refresh percentage is 80%.

tcp: Uses TCP as the transport protocol. By default, UDP is used.

tls: Uses TLS as the transport protocol.

scheme: Specifies a URL scheme.

sip: Specifies SIP as the URL scheme. By default, SIP is used.

sips: Specifies SIPS as the URL scheme.

Usage guidelines

When the registration time reaches the registration expiration time multiplied by the refresh percentage, a voice entity or SIP trunk re-registers the number with the registrar to avoid expiration.

If you use TLS as the transport protocol for registration, the port number specified in this command must be the same as the one configured on the registrar.

Examples

# Configure a registrar with the IP address 169.54.5.10 and port number 1120, and set the registration expiration time to 120 seconds.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] registrar 1 ip 169.54.5.10 port 1120 expires 120

# Configure a registrar with domain name cc.news.com and port number 1100, and set the registration expiration time to 520 seconds.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] registrar 2 dns cc.news.com port 1100 expires 520

Related commands

credentials

display voice sip register-status

transport

rel1xx

Use rel1xx to configure reliable provisional responses.

Use undo rel1xx to restore the default.

Syntax

rel1xx { disable | require value | supported value }

undo rel1xx

Default

The UAC sends INVITE requests with the Supported: 100rel header field. The UAS sends 18x responses with the Require: 100rel header field.

Views

SIP view

Predefined user roles

network-admin

Parameters

disable: Disables reliable provisional responses.

require value: Enables reliable provisional responses by adding the Required: value header field to outgoing INVITE requests or outgoing 18x responses. The value argument is a case-sensitive string of 1 to 49 characters.

supported value: Enables reliable provisional responses by adding the Supported: value header field to outgoing INVITE requests or the Require: value header field to outgoing 18x responses. The value argument is a case-sensitive string of 1 to 49 characters.

Usage guidelines

To implement reliable provisional responses, enable this feature and configure the same value for the value argument on both the UAC and UAS.

Examples

# Enable the device to send INVITE requests with the Supported: 100rel header field or 18x responses with the Require: rel100 header field.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] rel1xx require 100rel

remote-party-id

Use remote-party-id to add the Remote-Party-ID header field to outgoing INVITE requests.

Use remote-party-id to not add the Remote-Party-ID header field to outgoing INVITE requests.

Syntax

remote-party-id

undo remote-party-id

Default

Outgoing INVITE requests include the Remote-Party-ID header field.

Views

SIP view

Predefined user roles

network-admin

Examples

# Add the Remote-Party-ID header field to outgoing INVITE requests.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] remote-party-id

reset voice sip connection

Use reset voice sip connection to disconnect a SIP connection.

Syntax

reset voice sip connection { tcp | tls } id conn-id

Views

User view

Predefined user roles

network-admin

Parameters

tcp: Specifies TCP connections.

tls: Specifies TLS connections.

id conn-id: Specifies a connection by its connection ID in the range of 0 to 2147483647. You can use the display voice sip connection command to determine connection IDs.

Usage guidelines

This command disconnects either an established SIP connection or a SIP connection that is being established.

Examples

# Disconnect the TCP connection with a connection ID of 1.

<Sysname> reset voice sip connection tcp id 1

Related commands

display voice sip connection

retry invite

Use retry invite to set the maximum number of INVITE request retries.

Use undo retry invite to restore the default.

Syntax

retry invite times

undo retry invite

Default

The maximum number of INVITE request retries is 6.

Views

SIP view

Predefined user roles

network-admin

Parameters

times: Specifies the maximum number of INVITE request retries, in the range of 1 to 10.

Usage guidelines

The originating device starts an INVITE retry timer when sending an INVITE request. If no 100 response arrives when the timer expires, the originating device retransmits the INVITE request. If no 100 response arrives when the maximum number of INVITE retries is reached, the originating device clears the call.

Examples

# Set the maximum number of INVITE request retries to 5.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] retry invite 5

Related commands

timers trying

session refresh

Use session refresh to enable SIP session refresh globally.

Use undo session refresh to disable SIP session refresh globally.

Syntax

session refresh

undo session refresh

Default

SIP session refresh is globally disabled if the device acts as a UAC, and is globally enabled if the device acts as a UAS.

Views

SIP view

Predefined user roles

network-admin

Usage guidelines

Use this command on a UAC.

Examples

# Globally enable SIP session refresh.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] session refresh

Related commands

min-se

session transport

Use session transport to specify a transport protocol for outgoing SIP calls.

Use undo session transport to restore the default.

Syntax

session transport { tcp [ tls ] | udp }

undo session transport

Default

The global transport protocol for outgoing SIP calls is UDP. The transport protocol for outgoing SIP calls on a VoIP entity is the same as the global default.

Views

SIP view

VoIP entity view

Predefined user roles

network-admin

Parameters

udp: Specifies UDP as the transport protocol.

tcp: Specifies TCP as the transport protocol.

tls: Specifies TLS as the transport protocol.

Usage guidelines

You can configure the transport protocol both globally (in SIP view) and for a specific VoIP entity (in VoIP entity view). The configuration in VoIP entity view takes precedence over the global configuration. A VoIP entity uses the global configuration only when no transport protocol is configured in VoIP entity view.

Configure the same transport protocol on the called and calling devices. For example, if you configure the session transport tcp command on the calling device, you must configure the transport tcp command on the called device.

You must configure the SSL client and server policies by using the crypto command before you can use TLS to initiate SIP calls.

Examples

# Specify TLS as the global transport protocol for outgoing SIP calls.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] session transport tcp tls

Related commands

crypto

transport

set pstn-cause

Use set pstn-cause to configure a PSTN cause-to-SIP status mapping.

Use undo set pstn-cause to restore the default.

Syntax

set pstn-cause pstn-cause sip-status sip-status

undo set pstn-cause pstn-cause

Default

Table 8 Default PSTN cause-to-SIP status mappings

PSTN cause code

PSTN cause description

SIP status code

SIP status description

1

Unallocated (unassigned) number!

404

Not Found.

2

No route to specified transit network!

404

Not Found.

3

No route to destination!

404

Not Found.

16

Normal clearing!

N/A

BYE or CANCEL.

17

User busy!

486

Busy here.

18

No user responding!

408

Request Timeout.

19

No answer from user!

480

Temporarily unavailable.

20

Subscriber absent!

480

Temporarily unavailable.

21

Call rejected!

403

Forbidden.

22

Number changed!

410

Gone.

23

Redirection to new destination!

410

Gone.

25

Exchange routing error!

500

Server internal error.

26

Non-selected user clearing!

404

Not Found.

27

Destination out of order!

502

Bad Gateway.

28

Invalid number format (address incomplete)!

484

Address incomplete.

29

Facility rejected!

501

Not implemented.

31

Normal, unspecified!

480

Temporarily unavailable.

34

No circuit/channel available!

503

Service unavailable.

38

Network out of order!

503

Service unavailable.

41

Temporary failure!

503

Service unavailable.

42

Switching equipment congestion!

503

Service unavailable.

47

Resource unavailable, unspecified!

503

Service unavailable.

55

Incoming class barred within Closed User Group (CUG)!

403

Forbidden.

57

Bearer capability not authorized!

403

Forbidden.

58

Bearer capability not presently available!

503

Service unavailable.

63

Service or option not available, unspecified!

500

Server internal error.

65

Bearer capability not implemented!

488

Not Acceptable Here.

70

Only restricted digital information bearer capability is available!

488

Not Acceptable Here.

79

Service or option not implemented, unspecified!

501

Not implemented.

87

User not member of Closed User Group (CUG)!

403

Forbidden.

88

Incompatible destination!

503

Service unavailable.

102

Recovery on timer expiry!

504

Gateway timeout.

111

Protocol error, unspecified!

500

Server internal error.

127

Interworking, unspecified!

500

Server internal error.

 

Views

SIP view

Predefined user roles

network-admin

Parameters

pstn-code: Specifies a PSTN cause code in Table 8. The PSTN cause code 16 is invalid.

sip-code: Specifies a SIP status code in Table 8.

Examples

# Map the PSTN cause code 17 to the SIP status code 408.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] set pstn-cause 17 sip-status 408

set sip-status

Use set sip-status to configure a SIP status-to-PSTN cause mapping.

Use undo set sip-status to restore the default.

Syntax

set sip-status sip-status pstn-cause pstn-cause

undo set sip-status sip-status

Default

Table 9 Default SIP status-to-PSTN cause mappings

SIP status code

SIP status description

PSTN cause code

PSTN cause description

400

Bad Request.

41

Temporary failure!

401

Unauthorized.

21

Call rejected!

402

Payment required.

21

Call rejected!

403

Forbidden.

21

Call rejected!

404

Not found.

1

Unallocated (unassigned) number!

405

Method not allowed.

63

Service or option not available, unspecified!

406

Not acceptable.

79

Service or option not implemented, unspecified!

407

Proxy authentication required.

21

Call rejected!

408

Request timeout.

102

Recovery on timer expiry!

410

Gone.

22

Number changed!

413

Request Entity too long.

127

Interworking, unspecified!

414

Request-URI too long.

127

Interworking, unspecified!

415

Unsupported media type.

79

Service or option not implemented, unspecified!

416

Unsupported URI Scheme.

127

Interworking, unspecified!

420

Bad extension.

127

Interworking, unspecified!

421

Extension Required.

127

Interworking, unspecified!

423

Interval Too Brief.

127

Interworking, unspecified!

480

Temporarily unavailable.

18

No user responding!

481

Call/Transaction Does not Exist.

41

Temporary failure!

482

Loop Detected.

25

Exchange routing error!

483

Too many hops.

25

Exchange routing error!

484

Address incomplete.

28

Invalid number format (address incomplete)!

485

Ambiguous.

1

Unallocated (unassigned) number!

486

Busy here.

17

User busy!

487

Request Terminated.

127

Interworking, unspecified!

488

Not Acceptable here.

127

Interworking, unspecified!

500

Server internal error.

41

Temporary failure!

501

Not implemented.

79

Service or option not implemented, unspecified!

502

Bad gateway.

38

Network out of order!

503

Service unavailable.

41

Temporary failure!

504

Server time-out.

102

Recovery on timer expiry!

505

Version Not Supported.

127

Interworking, unspecified!

513

Message Too Large.

127

Interworking, unspecified!

600

Busy everywhere.

17

User busy!

603

Decline.

21

Call rejected!

604

Does not exist anywhere.

1

Unallocated (unassigned) number!

606

Not acceptable.

58

Bearer capability not presently available!

 

Views

SIP view

Predefined user roles

network-admin

Parameters

sip-code: Specifies a SIP status code in Table 9.

pstn-code: Specifies a PSTN cause code in Table 9.

Examples

# Map the SIP status code 486 to the PSTN cause code 18.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] set sip-status 486 pstn-cause 18

signaling forward rawmsg

Use signaling forward rawmsg to enable QSIG tunneling over SIP-T.

Use undo signaling forward rawmsg to disable QSIG tunneling over SIP-T.

Syntax

signaling forward rawmsg

undo signaling forward rawmsg

Default

QSIG tunneling over SIP-T is disabled.

Views

VoIP entity view

Predefined user roles

network-admin

Usage guidelines

This command enables sending QSIG signaling in SIP messages. In the SIP messages, the Content-type field is application/qsig, and the message body is the QSIG signaling received from the ISDN side.

The device does not support QSIG tunneling over SIP-T when the ISDN network uses overlap sending.

The SIP server might fail to interpret SIP messages carrying QSIG signaling. As a best practice, do not enable QSIG tunneling over SIP-T on a network where the device communicates with the SIP server.

Examples

# Enable QSIG tunneling over SIP-T.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 voip

[Sysname-voice-dial-entity10] signaling forward rawmsg

sip

Use sip to enter SIP view.

Use undo sip to remove the settings from SIP view.

Syntax

sip

undo sip

Views

Voice view

Predefined user roles

network-admin

Examples

# Enter SIP view.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip]

sip-compatible

Use sip-compatible to configure SIP compatibility with a third-party device.

Use undo sip-compatible to restore the default.

Syntax

sip-compatible { cause-code | early-media | t38 | x-param }

undo sip-compatible { cause-code | early-media | t38 | x-param }

Default

SIP compatibility is not configured.

Views

SIP view

Predefined user roles

network-admin

Parameters

cause-code: Configures SIP compatibility for SIP cause code interaction. With this keyword specified, the device uses Cause 27 instead of Cause 38 as the cause code for disconnecting a call.

early-media: Configures SIP compatibility for early media channels. With this keyword specified, the device does not disconnect the established early media channel upon receiving an 18x message without SDP from the terminating side.

t38: Configures SIP compatibility for standard T.38 fax. With this keyword specified, the device excludes :0 from the following SDP parameters in the originated re-INVITE messages:

·     T38FaxTranscodingJBIG.

·     T38FaxTranscodingMMR.

·     T38FaxFillBitRemoval.

This keyword is required when the device interoperates with a third-party softswitch device to exchange T.38 fax messages.

x-param: Configures SIP compatibility for fax pass-through and modem pass-through. With this keyword specified, the device adds SDP description information for fax pass-through and modem pass-through to outgoing re-INVITE messages. This keyword is required when the device interoperates with a third-party softswitch device to perform fax pass-through and modem pass-through.

Usage guidelines

If SIP implementations of a third-party device are special, you can configure SIP compatibility for the device to interoperate with the third-party device.

You can execute this command multiple times to specify multiple parameters.

Examples

# Configure SIP compatibility for standard T.38 fax.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] sip-compatible t38

sip-domain

Use sip-domain to configure a SIP domain name for the device.

Use undo sip-domain to restore the default.

Syntax

sip-domain domain-name

undo sip-domain

Default

No SIP domain name is configured. The device inserts the IP address of the outgoing interface in the Contact header field of an outgoing SIP packet.

Views

SIP view

Predefined user roles

network-admin

Parameters

domain-name: Specifies the SIP domain name, a case-insensitive string of 1 to 31 characters. Valid characters are letters, digits, underscore (_), hyphen (-), and dot (.).

Usage guidelines

Use this command to enable the device to insert the SIP domain name in the Contact header field of outgoing SIP packets.

Examples

# Configure the SIP domain name as abc.com.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] sip-domain abc.com

srtp

Use srtp to configure SRTP as the media stream protocol for SIP calls.

Use undo srtp to restore the default.

Syntax

srtp [ fallback ]

undo srtp

Default

The global media stream protocol for SIP calls is RTP. The media stream protocol for SIP calls on a VoIP entity is the same as the global default.

Views

SIP view

VoIP entity view

Predefined user roles

network-admin

Parameters

fallback: Supports fallback to RTP if the peer does not support SRTP.

Usage guidelines

The differences between the srtp and srtp fallback commands are as follows:

·     If the srtp command is configured, the following conditions exist:

¡     The device includes crypto and RTP/SAVP parameters in outgoing INVITE requests and disconnects the call after receiving a 488 response.

¡     The device can accept only calls using SRTP.

·     If the srtp fallback command is configured, the following conditions exist:

¡     The device includes crypto and RTP/SAVP parameters in outgoing INVITE requests and retransmits INVITE requests with RTP/AVP parameters after receiving a 488 response.

¡     The device can accept calls using SRTP or RTP. SRTP is preferred for media stream protocol negotiation. If the negotiation fails, RTP is used.

You can configure the srtp command globally (in SIP view) and for a specific VoIP entity (in VoIP entity view). The configuration in VoIP entity view takes precedence over the global configuration. A VoIP entity uses the global configuration only when the srtp command is not configured in VoIP entity view.

Examples

# Configure SRTP as the media stream protocol for SIP calls.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] srtp

timers connection aging

Use timers connection aging to set the aging time for TCP or TLS connections.

Use undo timers connection aging to restore the default.

Syntax

timers connection aging { tcp tcp-age-time | tls tls-age-time }

undo timers connection aging { tcp | tls }

Default

The aging time for TCP connections is 5 minutes. The aging time for TLS connections is 30 minutes.

Views

SIP view

Predefined user roles

network-admin

Parameters

tcp tcp-age-time: Specifies the amount of idle time that elapses before a TCP connection is removed. The value range for the tcp-age-time argument is 5 to 30 minutes.

tls tls-age-time: Specifies the amount of idle time that elapses before a TLS connection is removed. The value range for the tls-age-time argument is 30 to 180 minutes.

Examples

# Set the aging time to 6 minutes for TCP connections and 60 minutes for TLS connections.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] timers connection aging tcp 6

[Sysname-voice-sip] timers connection aging tls 60

timers options

Use timers options to set the interval for sending out-of-dialog OPTIONS messages.

Use undo timers options to restore the default.

Syntax

timers options value

undo timers options

Default

The interval for sending out-of-dialog OPTIONS messages is 500 milliseconds.

Views

SIP view

Predefined user roles

network-admin

Parameters

value: Specifies the interval for sending out-of-dialog OPTIONS messages, in the range of 100 to 1000 milliseconds.

Usage guidelines

This command takes effect only when the out-of-dialog keepalive feature has been enabled by using the voice-class sip options-keepalive command. For more information about the interval, see the usage guidelines for the voice-class sip options-keepalive command.

Examples

# Set the interval to 600 milliseconds for sending out-of-dialog OPTIONS messages.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] timer registration expires 600

Related commands

voice-class sip options-keepalive

timers trying

Use timers trying to set the INVITE retry timer.

Use undo timers trying to restore the default.

Syntax

timers trying timer-length

undo timers trying

Default

The INVITE retry timer is 500 milliseconds.

Views

SIP view

Predefined user roles

network-admin

Parameters

timer-length: Specifies the INVITE retry timer value in the range of 100 to 1000 milliseconds.

Usage guidelines

The INVITE retry timer defines the amount of time to wait for a 100 response to an INVITE request. The originating device starts an INVITE retry timer when sending an INVITE request. If it does not receive a 100 response when the timer expires, it retransmits the INVITE request.

Examples

# Set the INVITE retry timer to 600 milliseconds.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] timers trying 600

Related commands

retry invite

transport

Use transport to enable the listening port for a transport protocol.

Use undo transport to disable the listening port for a transport protocol.

Syntax

transport { tcp [ tls ] | udp }

undo transport { tcp [ tls ] | udp }

Default

The UDP and TCP listening ports are enabled. The TLS listening port is disabled.

Views

SIP view

Predefined user roles

network-admin

Parameters

udp: Enables the UDP listening port (port 5060).

tcp: Enables the TCP listening port (port 5060).

tls: Enables the TLS listening port (port 5061).

Usage guidelines

You can use this command multiple times to enable multiple listening ports for different protocols. The protocols do not interfere with each other.

For the device to receive calls and initiate registrations or subscriptions, configure this command to enable the corresponding listening port.

You must configure the SSL client and server policies by using the crypto command before you can enable the TLS listening port.

The undo transport command removes established connections.

Examples

# Enable the TLS listening port.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] transport tcp tls

Related commands

crypto

mwi

registrar

url

Use url to configure a global URL scheme for outgoing SIP calls.

Use undo url to restore the default.

Syntax

url { sip | sips }

undo url

Default

The SIP scheme is used.

Views

SIP view

Predefined user roles

network-admin

Parameters

sip: Specifies the SIP scheme.

sips: Specifies the SIPS scheme.

Usage guidelines

You can configure the URL scheme both globally (by using the url command in SIP view) and for a specific VoIP entity (by using the voice-class sip url command in VoIP entity view). The configuration in VoIP entity view takes precedence over the global configuration. A VoIP entity uses the global configuration only when no URL scheme is configured in VoIP entity view.

Examples

# Specify SIPS as the global URL scheme.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] url sips

Related commands

voice-class sip url (in VoIP entity view)

user

Use user to configure SIP credentials.

Use undo user to delete SIP credentials.

Syntax

user username password { cipher | simple } string [ realm realm ]

undo user [ username password { cipher | simple } string [ realm realm ] ]

Default

No SIP credentials exist.

Views

SIP view

POTS entity view

Predefined user roles

network-admin

Parameters

username: Specifies a username, a case-sensitive string of 1 to 63 characters.

cipher: Specifies a password in encrypted form.

simple: Specifies a password in plaintext form. For security purposes, the password specified in plaintext form will be stored in encrypted form.

string: Specifies the password. Its plaintext form is a case-sensitive string of 1 to 16 characters. Its encrypted form is a case-sensitive string of 1 to 53 characters.

realm realm: Specifies a realm, a case-sensitive string of 1 to 50 characters. If you do not specify a realm, the credentials can be used to respond to any registrars.

Usage guidelines

A SIP UA can register with a maximum of six registrars, and it uses the domain name in the 401/407 response from a registrar to identify the credentials to be sent to the registrar.

You can configure only one username with the user command in SIP view or voice entity view. The username can contain 12 credentials bindings. A binding that does not include a domain name can be used to respond to a 401/407 response that does not match any domain name-included binding. The following example configures four credentials bindings:

[Sysname-voice-dial-entity100] user 1000 password simple 1000 realm server1

[Sysname-voice-dial-entity100] user 1000 password simple 1000 realm server2

[Sysname-voice-dial-entity100] user 1000 password simple 2000 realm server3

[Sysname-voice-dial-entity100] user 1000 password simple 3000

The first three bindings each contain a domain name, and the last binding contains no domain name. If the SIP UA receives a 401/407 response that includes a domain name server2, the SIP UA responds with the username 1000 and password 1000. If the SIP UA receives a 401/407 response that includes a domain name server4, the SIP UA responds with the username 1000 and password 3000 because no credentials binding contains the domain name server4.

Examples

# Configure global SIP credentials that include username abcd and plaintext password 1234.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] user abcd password simple 1234

# Configure SIP credentials that include username abcd, plaintext password 1234, and realm abc for POTS entity 100.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 100 pots

[Sysname-voice-dial-entity100] user abcd password simple 1234 realm abc

Related commands

registrar

voice-class sip bind

Use voice-class sip bind to configure source interface binding for a VoIP entity.

Use undo voice-class sip bind to restore the default.

Syntax

voice-class sip bind { control | media } source-interface interface-type interface-number

undo voice-class sip bind { control | media }

Default

The default global source interface is used.

Views

VoIP entity view

Predefined user roles

network-admin

Parameters

control: Specifies outbound SIP messages.

media: Specifies outbound media packets.

source-interface interface-type interface-number: Uses the specified interface as the source interface of outbound SIP messages or media packets. The specified interface must be a Layer 3 Ethernet interface or dialer interface.

Usage guidelines

This command uses the IP address of the specified interface as the source address for outgoing SIP or media packets.

For information about how source interface binding works in different conditions, see the usage guidelines for the bind command.

You can configure source interface binding both globally (by using the bind command in SIP view) and for a specific VoIP entity (by using the voice-class sip bind command in VoIP entity view). The configuration in VoIP entity view takes precedence over the global configuration. A VoIP entity uses the global configuration only when source interface binding is not configured in VoIP entity view.

Examples

# Specify Dialer 0 as the source interface for outbound SIP messages on VoIP entity 10.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 voip

[Sysname-voice-dial-entity10] voice-class sip bind control source-interface dialer 0

Related commands

bind

voice-class sip options-keepalive

Use voice-class sip options-keepalive to enable out-of-dialog keepalive for a VoIP entity and optionally set the out-of-dialog keepalive parameters.

Use undo voice-class sip options-keepalive to disable out-of-dialog keepalive for a VoIP entity.

Syntax

voice-class sip options-keepalive [ up-interval interval ] [ down-interval interval ] [ retry retries ]

undo voice-class sip options-keepalive

Default

Out-of-dialog keepalive is disabled for a VoIP entity.

Views

VoIP entity view

Predefined user roles

network-admin

Parameters

up-interval interval: Specifies the interval for sending out-of-dialog OPTIONS packets when the VoIP entity is available, in the range of 5 to 1200 seconds. The default value is 60 seconds.

down-interval interval: Specifies the interval for sending out-of-dialog OPTIONS packets when the VoIP entity is not available, in the range of 5 to 1200 seconds. The default value is 30 seconds.

retry retries: Specifies the number of retries to change the state for the VoIP entity.

Usage guidelines

After you enable the out-of-dialog keepalive feature for a VoIP entity, the UA sends OPTIONS packets at the up-interval. If the UA receives a response within the up-interval, it considers the VoIP entity to be available. If the UA receives no response within the up-interval, or receives an error response, it sends OPTIONS packets at the timers options interval. (Error responses include 408, 499, and 5XX responses except for 500, 501, 502, 503, 504, and 513 responses.) If the UA still receives no responses after the maximum number of retries is reached, it considers the VoIP entity to be unavailable.

Then, the UA sends OPTIONS packets at the down-interval. If the UA receives a response within the down-interval, it sends OPTIONS packets at the timers options interval. If the UA still can receive responses after the number of retries is reached, it considers the VoIP entity to be available.

The keepalive feature does not take effect for a VoIP entity that has been shut down by using the shutdown command.

Examples

# Enable out-of-dialog keepalive for VoIP entity 10, and set the up-interval to 50 seconds, the down-interval to 20 seconds, and the number of retries to 2.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 voip

[Sysname-voice-dial-entity10] voice-class sip options-keepalive up-interval 50 down-interval 20 retry 2

Related commands

timers options

voice-class sip options-ping

Use voice-class sip options-ping to enable in-dialog keepalive for a VoIP entity.

Use voice-class sip options-ping to disable in-dialog keepalive for a VoIP entity.

Syntax

voice-class sip options-ping { global | seconds }

undo voice-class sip options-ping

Default

A VoIP entity uses the global configuration for in-dialog keepalive.

Views

VoIP entity view

Predefined user roles

network-admin

Parameters

global: Applies the global configuration for in-dialog keepalive to the VoIP entity.

seconds: Specifies the interval for sending OPTIONS messages during a session, in the range of 60 to 1200 seconds.

Usage guidelines

For a VoIP entity, the entity-specific in-dialog keepalive configuration takes priority over the global in-dialog keepalive configuration set in SIP view.

Examples

# Enable in-dialog keepalive for VoIP entity 1 and set the interval to 60 seconds for sending OPTIONS messages during a session.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 1 voip

[Sysname-voice-dial-entity1] voice-class sip options-ping 60

# Apply the global configuration for in-dialog keepalive to VoIP entity 1.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 1 voip

[Sysname-voice-dial-entity1] voice-class sip options-ping global

Related commands

options-ping

voice-class sip session refresh

Use voice-class sip session refresh to enable SIP session refresh for a VoIP entity.

Use undo voice-class sip session refresh to disable SIP session refresh for a VoIP entity.

Syntax

voice-class sip session refresh [ global ]

undo voice-class sip session refresh

Default

A VoIP entity uses the global configuration for SIP session refresh.

Views

VoIP entity view

Predefined user roles

network-admin

Parameters

global: Applies the global configuration for SIP session refresh to the VoIP entity.

Usage guidelines

The configuration for SIP session refresh in VoIP entity view takes priority over that in SIP view.

Examples

# Enable SIP session refresh for VoIP entity 1.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 1 voip

[Sysname-voice-dial-entity1] voice-class sip session refresh

# Apply the global configuration for SIP session refresh to VoIP entity 1.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 1 voip

[Sysname-voice-dial-entity1] voice-class sip session refresh global

Related commands

min-se

session refresh

voice-class sip url

Use voice-class sip url to configure a URL scheme for outgoing SIP calls on a VoIP entity.

Use undo voice-class sip url to restore the default.

Syntax

voice-class sip url { sip | sips }

undo voice-class sip url

Default

The default global URL scheme (SIP scheme) is used.

Views

VoIP entity view

Predefined user roles

network-admin

Parameters

sip: Specifies the SIP scheme.

sips: Specifies the SIPS scheme.

Usage guidelines

You can configure the URL scheme both globally (by using the url command in SIP view) and for a specific VoIP entity (by using the voice-class sip url command in VoIP entity view). The configuration in VoIP entity view takes precedence over the global configuration. A VoIP entity uses the global configuration only when no URL scheme is configured in VoIP entity view.

Examples

# Specify the SIP scheme for outgoing SIP calls on VoIP entity 1000.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 1000 voip

[Sysname-voice-dial-entity1000] voice-class sip url sips

Related commands

url

vpn-instance

Use vpn-instance to associate a VPN instance with SIP.

Use undo vpn-instance to remove the association.

Syntax

vpn-instance vpn-instance-name

undo vpn-instance

Default

No VPN instance is associated with SIP.

Views

SIP view

Predefined user roles

network-admin

Parameters

vpn-instance-name: Specifies an MPLS L3VPN instance by its name, a case-sensitive string of 1 to 31 characters.

Usage guidelines

The VPN instance to be associated with SIP must be already created.

You cannot associate a VPN instance or remove the association when a SIP service is being used.

Examples

# Associate VPN instance vpn-voice with SIP.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] vpn-instance vpn-voice

Related commands

ip binding vpn-instance (MPLS Command Reference)

ip vpn-instance (MPLS Command Reference)

 


SIP trunk commands

The following compatibility matrix shows the support of hardware platforms for SIP trunk:

 

Hardware

SIP trunk compatibility

MSR810, MSR810-W, MSR810-W-DB, MSR810-LM, MSR810-W-LM, MSR810-10-PoE, MSR810-LM-HK, MSR810-W-LM-HK, MSR810-LMS-EA

MSR810, MSR810-W, MSR810-W-DB, MSR810-LM, MSR810-W-LM, MSR810-10-PoE, MSR810-LM-HK, MSR810-W-LM-HK: Yes

MSR810-LMS-EA: No

MSR810-LMS, MSR810-LUS

No

MSR2600-6-X1

No

MSR2600-10-X1

Yes

MSR 2630

Yes

MSR3600-28, MSR3600-51

Yes

MSR3600-28-SI, MSR3600-51-SI

No

MSR3600-28-X1, MSR3600-28-X1-DP, MSR3600-51-X1, MSR3600-51-X1-DP

No

MSR3610-I-DP, MSR3610-IE-DP

No

MSR3610-X1, MSR3610-X1-DP, MSR3610-X1-DC, MSR3610-X1-DP-DC

Yes

MSR 3610, MSR 3620, MSR 3620-DP, MSR 3640, MSR 3660

Yes

MSR3610-G, MSR3620-G

No

To support SIP trunk, some device models require the Voice Software License. For more information, see license management in Fundamentals Configuration Guide.

allow-connections sip to sip

Use allow-connections sip to sip to enable SIP-to-SIP calling.

Use undo allow-connections sip to sip to disable SIP-to-SIP calling.

Syntax

allow-connections sip to sip

undo allow-connections sip to sip

Default

SIP-to-SIP calling is disabled.

Views

Voice view

Predefined user roles

network-admin

Usage guidelines

After you enable SIP-to-SIP calling, the device works as a SIP trunk device. As a best practice, do not use a SIP trunk device as a SIP UA.

Examples

# Enable SIP-to-SIP calling.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] allow-connections sip to sip

codec transparent

Use codec transparent to enable codec transparent transmission.

Use undo codec transparent to disable codec transparent transmission.

Syntax

codec transparent

undo codec transparent

Default

Codec transparent transmission is disabled. The SIP trunk device is involved in the media negotiation between the calling and called parties.

Views

VoIP entity view

Predefined user roles

network-admin

Usage guidelines

If the SIP trunk device does not support any codecs on the calling and called parties, you can enable codec transparent transmission. The SIP trunk device transparently forwards codec capability sets between the two parties without intervening codec negotiation.

Examples

# Enable codec transparent transmission for VoIP entity 1.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 1 voip

[Sysname-voice-dial-entity1] codec transparent

credentials

Use credentials to create a SIP trunk account.

Use undo credentials to delete a SIP trunk account.

Syntax

credentials number number username username password { cipher | simple } string realm realm

undo credentials { number number | number number username username password { cipher | simple } password realm realm }

Default

No SIP trunk accounts exist.

Views

SIP view

Predefined user roles

network-admin

Parameters

number: Specifies a number for the SIP trunk account, a case-sensitive string of 4 to 31 characters.

username: Specifies a username, a case-sensitive string of 1 to 63 characters.

cipher: Specifies a password in encrypted form.

simple: Specifies a password in plaintext form. For security purposes, the password specified in plaintext form will be stored in encrypted form.

string: Specifies the password. Its plaintext form is a case-sensitive string of 1 to 16 characters. Its encrypted form is a case-sensitive string of 1 to 53 characters.

realm realm: Specifies a realm, a case-sensitive string of 1 to 50 characters.

Usage guidelines

A SIP trunk account contains a phone number, credentials, and realms assigned by the service provider. SIP can send a REGISTER request for the phone number to a maximum of six registrars specified by using the registrar command. SIP uses the realm value in 401/407 responses from the registrars to identify the matching credentials. You can configure a maximum of 12 realms for a phone number, and a maximum of 128 SIP trunk accounts on the device.

Examples

# Configure a SIP trunk account for phone number 1000 that uses the following:

·     Username 1000 and password 1000 for realm server1.

·     Username 2000 and password 2000 for realm server2.

·     Username 3000 and password 3000 for realm server3.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip

[Sysname-voice-sip] credentials number 1000 username 1000 password simple 1000 realm server1

[Sysname-voice-sip] credentials number 1000 username 2000 password simple 2000 realm server2

[Sysname-voice-sip] credentials number 1000 username 3000 password simple 3000 realm server3

Related commands

registrar

media flow-around

Use media flow-around to enable media flow-around.

Use undo media flow-around to disable media flow-around.

Syntax

media flow-around

undo media flow-around

Default

Media flow-around is disabled. Media packets are relayed by the SIP trunk, and the SIP trunk device changes the media address of a media packet to its own address before forwarding the media packet.

Views

VoIP entity view

Predefined user roles

network-admin

Usage guidelines

This feature enables the SIP trunk device to directly forward media packets between SIP endpoints, without changing the media address for the media packets. Use this feature to improve forwarding performance.

Examples

# Enable media flow-around for VoIP entity 1.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 1 voip

[Sysname-voice-dial-entity1] media flow-around

voice-class sip early-offer forced

Use voice-class sip early-offer forced to enable delayed offer to early offer (DO-EO) conversion.

Use undo voice-class sip early-offer forced to disable DO-EO conversion

Syntax

voice-class sip early-offer forced

undo voice-class sip early-offer forced

Default

DO-EO conversion is disabled.

Views

VoIP entity view

Predefined user roles

network-admin

Usage guidelines

An INVITE request with SDP Offer is an early offer, and an INVITE request without SDP Offer is a delayed offer. Some service providers mandate early offer calls for charge security. To meet this requirement, enable DO-EO conversion on the SIP trunk device.

This command does not take effect if codec transparent transmission or media flow-around is enabled.

Examples

# Enable DO-EO conversion on the SIP trunk device.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 1 voip

[Sysname-voice-dial-entity1] voice-class sip early-offer forced

Related commands

codec transparent

media flow-around